| 00:02.22 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 00:03.49 | chendy | post invitation request , i think |
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| 00:35.42 | [TK]D-Fender | [19:05]<KavanS>anyone know why I would see the [May 7 15:09:11] NOTICE[3713]: chan_sip.c:13885 handle_request_invite: Call from â to extension '95035551212' rejected because extension not found. <--- just like it says... there is no match for that extension in the context it was told to search. Look at the SIP debug to see what actual peer it matched and what context it's looking in. |
| 00:35.55 | KavanS | I think it's a nat issue |
| 00:36.13 | KavanS | tracked it down a little further |
| 00:38.19 | [TK]D-Fender | KavanS: No, its |
| 00:38.21 | [TK]D-Fender | not |
| 00:38.33 | [TK]D-Fender | KavanS: EXTENSION NOT FOUND. |
| 00:38.50 | [TK]D-Fender | KavanS: What size of glowing neon sign do we require here? |
| 00:39.21 | [TK]D-Fender | KavanS: Its a dialplan issue unless its failing to math the peer its supposed to at which point its merely a peer issue, not NAT. |
| 00:39.39 | [TK]D-Fender | KavanS: Nat would prevent the call attempt from entering, or audio after it did. |
| 00:39.48 | [TK]D-Fender | KavanS: EXTENSION NOT FOUND. <- exactly what it means. |
| 00:40.06 | [TK]D-Fender | KavanS: So go look where I told you to see what peer its matching and what contesxt its looking in |
| 00:40.11 | [TK]D-Fender | heads out for the night |
| 00:40.14 | [TK]D-Fender | BBL |
| 00:40.21 | KavanS | ok, I will read more |
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| 00:46.42 | Artarian | Hello folks, anybody has experienced problems with conference manager on Elastix? |
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| 00:50.22 | Artarian | sombody alive? |
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| 01:37.26 | hardwire | I seem to have multiple astcanary hanging aboot. |
| 01:37.48 | hardwire | openvz with NET_ADMIN and SYS_ADMIN enabled so that asterisk can set it's caps. |
| 01:38.09 | hardwire | however when I stop asterisk (1.6.1.0) astcanary never goes away |
| 01:39.36 | hardwire | I probably can't run it with -p in an openvz nicely. |
| 01:45.10 | mmlj4 | apostrophic jihad? |
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| 02:14.21 | ks3 | Can I make Asterisk send a SIP 503 service unavailable response? I haven't been able to find a way to do this. |
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| 02:24.39 | pfn | is there an iax incompatibility between 1.4 and 1.6 or something? |
| 02:24.51 | pfn | it always sounds so jittery when I use iax instead of sip... |
| 02:28.34 | drmessano | Which version of 1.4? |
| 02:29.29 | pfn | a recentish version |
| 02:30.12 | jaytee | hmmm, 1.4.recentish |
| 02:30.38 | jaytee | I'm running 1.4.evenmorerecentish |
| 02:30.54 | Deeewayne | 1.4.recentish is totally stable, why would you ever update ? |
| 02:31.18 | jaytee | I like living on the bleeding edge :-) |
| 02:34.42 | jaytee | guy in the next apartment complex has an 8' x 8' fenced in backyard and he's using a weed whacker at 10:30pm to mow the backyard :-) |
| 02:34.54 | drmessano | I've always been a bit confused by Asterisk updates |
| 02:35.53 | drmessano | Like, I always down 1.new.perfectversion, and after a while, I get these email notifications that 1.new.moreperfectversion is out and that 1.new.perfectversion had all sorts of these little buggy thingos |
| 02:36.04 | drmessano | It makes me sad for 1.new.perfectversion |
| 02:36.06 | *** join/#asterisk aenaus (n=hdgfghf@79.107.184.121) |
| 02:36.07 | drmessano | :( |
| 02:37.11 | drmessano | Like, I really liked 1.6.0.5.. But then all the sudden, russellb was all like "ZOMG, fail list: " and I was like "ZOMG, 1.6.0.5 I never knew |
| 02:37.33 | drmessano | and then it's all like "Need to replace you, sorry.." and I feel kinda sad |
| 02:37.36 | pfn | snickers |
| 02:37.44 | pfn | I forget which version because I just svk pull |
| 02:38.08 | drmessano | svk? Is that like the KDE version of SVN or something? |
| 02:38.21 | pfn | svk = distributed svn |
| 02:38.31 | drmessano | Hmmm |
| 02:38.59 | drmessano | See, I am a complete idiot.. I run AsteriskWin32 here.. so that's kinda cool |
| 02:39.10 | drmessano | Lemme see if I have that on my Internet Explorer |
| 02:39.18 | pfn | Mirrored From: http://svn.digium.com/svn/asterisk/branches/1.4, Rev. 187362 |
| 02:40.09 | drmessano | See, I just run like svn update and one of those yum apt-get wget download.exe thingos runs and some linux scrolls by |
| 02:40.19 | drmessano | shrugs |
| 02:40.33 | pfn | is running 1.4-r187362 |
| 02:41.40 | drmessano | is running 1.6.0-r190419 |
| 02:42.35 | pfn | sounds interesting |
| 02:43.01 | drmessano | I need to move to 1.6.1 after I figure out what I will have to live without |
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| 02:59.27 | kc8pxy | drmessano: 1.6.1 drops some things? |
| 03:00.57 | drmessano | No, 3rd party apps like to not work on newer versions |
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| 03:06.59 | VaGoNeTaS | is back from the dead. Gone: 3h 54m 59s |
| 03:07.20 | VaGoNeTaS | i got an issue on one of our customers |
| 03:07.26 | VaGoNeTaS | is a small call center with 12 agents |
| 03:07.31 | VaGoNeTaS | sometimes they can talk but the customer on the line cant hear the agents |
| 03:07.44 | VaGoNeTaS | i was looking on the logs |
| 03:07.50 | VaGoNeTaS | and i found this |
| 03:08.05 | VaGoNeTaS | [May 7 19:31:17] DEBUG[18019] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/1-1 |
| 03:08.06 | VaGoNeTaS | [May 7 19:31:17] DEBUG[18019] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call |
| 03:08.06 | VaGoNeTaS | [May 7 19:31:17] DEBUG[18019] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/1-1 |
| 03:09.00 | VaGoNeTaS | that might be the reason |
| 03:09.08 | VaGoNeTaS | i dont know how to fix it |
| 03:15.26 | *** join/#asterisk borai (n=bora@S0106001c109e98db.no.shawcable.net) |
| 03:15.39 | borai | Hi |
| 03:16.06 | VaGoNeTaS | hi |
| 03:16.27 | borai | I just got a free fax for asterisk license, I am following the instructions I have used the register command now when I execute branchfax over SSH it always gets stuck at a test run |
| 03:17.06 | borai | I have waited for 20 minutes and the output didnt change |
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| 03:22.04 | VaGoNeTaS | i reallly need a hand with this |
| 03:22.27 | stope | are you using g729? |
| 03:23.45 | borai | do you need g729 |
| 03:25.06 | VaGoNeTaS | who? |
| 03:25.28 | VaGoNeTaS | what's g729? |
| 03:26.18 | borai | its a codec |
| 03:26.30 | VaGoNeTaS | u say for my issue? |
| 03:26.47 | borai | I dont know what your issue is |
| 03:26.49 | *** join/#asterisk kerx (n=kerx@adsl-69-104-164-39.dsl.irvnca.pacbell.net) |
| 03:27.03 | VaGoNeTaS | sometimes the calls got cut up |
| 03:27.13 | VaGoNeTaS | and the cx keeps listening the agent |
| 03:27.25 | VaGoNeTaS | but the agent thinks that the call got cut up |
| 03:27.38 | kerx | hi, what is the command for being able to bridge a call ? it doesn't seem bridge() works |
| 03:40.06 | borai | ugh |
| 03:50.08 | Deeewayne | borai, what is branchfax ? |
| 04:03.36 | borai | sorry *benchfax |
| 04:05.06 | Deeewayne | try one of the generic fax modules |
| 04:09.01 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
| 04:09.36 | kerx | anyway to hack Bridge() on 1.4.x ? |
| 04:12.26 | drmessano | What are you trying to "hack"? |
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| 04:16.46 | borai | drmessano: I followed your SPA guide the other day and I got PSTN to VOIP working but when I try VOIP to PSTN I always get a circuits are busy message. |
| 04:16.57 | drmessano | Which guide? |
| 04:18.10 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 04:18.11 | borai | the one you had on the wiki |
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| 04:19.44 | drmessano | Since there's several copies of guides I have done out there, you need to be more specific |
| 04:21.19 | borai | I dont have the url here but it was on a site that was like a wikimedia install. |
| 04:21.26 | drmessano | Ok |
| 04:21.51 | drmessano | Only thing I know to tell you is go back and reread it |
| 04:21.58 | drmessano | Make sure you factory reset FIRST |
| 04:22.43 | borai | I followed the guide completely including the factory reset. |
| 04:23.08 | drmessano | Ok, pastebin configs and debug |
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| 04:23.41 | *** mode/#asterisk [+o angler] by ChanServ |
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| 04:26.03 | borai | why is it that everytime I logon to IRC here one or more of my remote sites are having network issues |
| 04:29.17 | kerx | drmessano, sorry I'm replying late. I have an active call, and I want to setup a featuremap that does a Bridge to a second call (basically a 3-way). |
| 04:29.49 | kerx | Initiated by the featuremap, and I'm using Asterisk 1.4.23.1 |
| 04:30.08 | kerx | A little hesitant to upgrade to 1.6, eventhough 1.6 does include the Bridge() application |
| 04:32.12 | drmessano | 1.6 works fine |
| 04:32.49 | Deeewayne | <-- also likes 1.6 |
| 04:34.29 | kerx | i guess i should just man up then :) i was worried about the call queue's |
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| 04:53.14 | kerx | ~book |
| 04:53.15 | 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 |
| 04:53.38 | kerx | Anyone know if there is a book in the works for Asterisk 1.6 and cover's all the new features? |
| 04:57.21 | drmessano | There is |
| 04:57.25 | drmessano | Its a ways off though |
| 05:08.48 | kerx | Ok, it would be helpful. Big changes I notice in the config's |
| 05:09.03 | kerx | Doesn't look like I can just drop in my 1.4 config's to 1.6 |
| 05:09.10 | kerx | I have pretty basic 1.4 config's |
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| 05:32.00 | cyd777_wrk | hi |
| 05:33.18 | cyd777_wrk | i Have a question. I plan to build a fax server with asterisk, hylafax and iax modem. It will handla 3 or more ISDN line. |
| 05:34.55 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 05:35.34 | cyd777_wrk | I tried to find a BRI card for this, but I can't decide which better for me. I think I have 2 alternatives: 1: Sangoma A500 2-24 Port Scalable S/T BRI, 2: Digium Wildcard B410P |
| 05:37.08 | cyd777_wrk | My question is wich is easer to install in linux, specially debian? What are experiences with these 2 cards? |
| 05:42.44 | KyleK | if you can find howtos for both, might be pretty equal |
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| 05:59.23 | kc8pxy | mebbe I'm not looking in the right places, but i'm still triyng to figure out the lifecycle of sending and recieving faxes via asterisk. and how best to go about configuring it. |
| 05:59.42 | Subdolus | this isn't an asterisk specific issue, but it's annoying and no one can seem to help; I use an PAP2 ATA connecting directly to my service provider. now when I make calls I hear two different, seperate ringing tones |
| 06:00.02 | Subdolus | when the callee picks up, one of them stops but in the background the other one is still ringing |
| 06:00.40 | kc8pxy | Subdolus: huh? i think I'm missing part of that. |
| 06:00.47 | Subdolus | I've scoured through the PAP2 settings and I can't find any distinctive feature... if nothing else I'm hoping someone here can throw a few words at me to describe it better |
| 06:01.17 | Subdolus | kc8pxy: lift receiver, dial number, ring ring, call is answered |
| 06:01.56 | Subdolus | that's how it should go.. but both my provider and ATA are generating the ring rings |
| 06:02.07 | Subdolus | over the top of eachother |
| 06:02.17 | Subdolus | make sense? |
| 06:03.37 | kc8pxy | Subdolus: did you do any config of the ata? |
| 06:04.06 | kc8pxy | Subdolus: what i didn't understand most is why that would happen OOB. |
| 06:04.54 | Subdolus | I may have in the past; we only recently switched to this new provider. never had this problem before |
| 06:05.14 | Subdolus | I have custom dialing rules etc. hoping I don't have to re-flash it |
| 06:07.10 | kc8pxy | Subdolus: is it a aa connected to your asterisk server? |
| 06:07.18 | kc8pxy | ata |
| 06:08.10 | Subdolus | one of the lines goes my asterisk box, yes. but not the line I use soley for this provider |
| 06:08.27 | Subdolus | which is the one that's generating the two tones |
| 06:09.27 | kc8pxy | without knowing more, i'd say your voip provider and yoru dialing config in the ata don't like eachother. |
| 06:10.26 | Subdolus | well... obviously. |
| 06:10.36 | Subdolus | that's what I've spent the last 10 mins trying to explain :) |
| 06:10.51 | kc8pxy | what can we add? |
| 06:11.01 | drmessano | Nothing |
| 06:11.10 | drmessano | Call the provider |
| 06:11.22 | Subdolus | is there a more specific term for this other than 'ringing tone' |
| 06:11.38 | drmessano | "shits broke" comes to mind |
| 06:11.51 | Subdolus | heh |
| 06:11.55 | drmessano | Tell them you continue to hear ringing |
| 06:12.07 | Subdolus | I will |
| 06:12.14 | Subdolus | but they will be about as much help as kc8pxy here |
| 06:12.30 | drmessano | Well, this isnt an ATA problem |
| 06:12.33 | Subdolus | I need to break it down just so they can forward me to the right department |
| 06:12.46 | drmessano | So you need to move on to them |
| 06:13.02 | kc8pxy | Subdolus: not all tech supp are idiots. only the front lines are, with any luck. |
| 06:13.12 | Subdolus | yeah I'll ask them to sort it I suppose |
| 06:13.53 | kc8pxy | is an ex-phone-bank-tech-supp rep. |
| 06:14.02 | Subdolus | doesn't care |
| 06:14.21 | drmessano | Attitude? |
| 06:15.20 | kc8pxy | garrr.. my development asterisk box should be replaced. gcc is barfing seg faults. |
| 06:15.39 | kc8pxy | is compiling spandsp |
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| 06:41.07 | porche | hi all, i have question about g729 |
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| 06:41.46 | porche | for outgoing calls, if license is not enough what happens? |
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| 06:56.53 | porche | ah found, silence calls, |
| 06:57.15 | porche | silent calls, if licence not enough, cool |
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| 07:02.35 | Gopaul | [TK]D-Fender: hi |
| 07:02.48 | Gopaul | [TK]D-Fender: yesterday I tried what you said |
| 07:03.32 | [TK]D-Fender | Gopaul: Which was? |
| 07:04.04 | Gopaul | [TK]D-Fender: the outgoing spool failed error |
| 07:04.18 | Gopaul | [TK]D-Fender: i deleted the voip account from GUI and tried configuring in a custom files, it worked fine without any issues, but today I am trying to execute the program facing some issues |
| 07:05.28 | Gopaul | [TK]D-Fender: the CLI output of the issue what I faced yesterday is here http://pastebin.com/m44a091e8 |
| 07:07.01 | Gopaul | [TK]D-Fender: I just wanted to know one thing, the astdb works as cache memory? |
| 07:07.44 | [TK]D-Fender | Gopaul: no |
| 07:08.54 | Gopaul | [TK]D-Fender: bcoz at first I was trying the call origination with ISDN PRI connection the same agi script worked without any issues for a long time |
| 07:09.28 | Gopaul | [TK]D-Fender: two days back I changed my ISDN PRI to VoIP account from that stage I am facing the issues |
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| 09:03.10 | funkykiwy | hello |
| 09:03.35 | funkykiwy | I got a problem with a TDM410P FXO and dahdi configuration |
| 09:03.45 | funkykiwy | is anybody can help me ? |
| 09:07.47 | sulex | "don't ask to ask, just ask" |
| 09:21.04 | funkykiwy | ? |
| 09:21.07 | funkykiwy | yeah |
| 09:21.11 | funkykiwy | but if nobody reply |
| 09:21.19 | funkykiwy | how can I know |
| 09:21.27 | funkykiwy | if someone want help me :) |
| 09:23.59 | funkykiwy | so whane I make or receive a call |
| 09:24.21 | funkykiwy | dahdi channel is created but a red alarm ended the call immidietly |
| 09:29.10 | kaldemar | show a CLI output of the call with full verbosity. |
| 09:29.21 | funkykiwy | oki |
| 09:30.40 | funkykiwy | 2 sec |
| 09:31.28 | funkykiwy | <PROTECTED> |
| 09:31.28 | funkykiwy | <PROTECTED> |
| 09:31.28 | funkykiwy | <PROTECTED> |
| 09:31.28 | funkykiwy | <PROTECTED> |
| 09:31.28 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.32 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.36 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.48 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.50 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.52 | kaldemar | ~pb |
| 09:31.53 | 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/ |
| 09:31.53 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.54 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.56 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:31.58 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.06 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.10 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.12 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.14 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.19 | kaldemar | don't flood the channel, use a pastebin. http://pastebin.ca for example. |
| 09:32.20 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.24 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.32 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: |
| 09:32.34 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.42 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.48 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.50 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:32.52 | funkykiwy | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/2-1] |
| 09:33.02 | funkykiwy | [May 8 10:58:39] WARNING[17807]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 2: Red Alarm |
| 09:33.04 | funkykiwy | << [ HANGUP (NULL) ] [DAHDI/2-1] |
| 09:33.06 | funkykiwy | <PROTECTED> |
| 09:33.08 | funkykiwy | <PROTECTED> |
| 09:33.10 | funkykiwy | <PROTECTED> |
| 09:33.12 | funkykiwy | [May 8 10:58:40] NOTICE[17021]: chan_dahdi.c:8164 handle_init_event: Alarm cleared on channel 2 |
| 09:33.14 | funkykiwy | here it is |
| 09:33.18 | funkykiwy | oki sory |
| 09:33.20 | funkykiwy | I will not so this anymore |
| 09:34.19 | funkykiwy | http://pastebin.ca/1416205 |
| 09:34.24 | funkykiwy | is it better ? |
| 09:36.04 | kaldemar | what does "dahdi show status" say? |
| 09:36.42 | funkykiwy | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 09:36.42 | funkykiwy | Wildcard TDM410P Board 1 OK 1 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 09:52.15 | funkykiwy | ? |
| 09:52.21 | funkykiwy | no idea ? |
| 09:55.10 | MaliutaLap | ~pb |
| 09:55.11 | 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/ |
| 09:55.20 | MaliutaLap | moron |
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| 09:55.45 | *** part/#asterisk ITguru (n=ITGuru@webfax.impactteachers.com) |
| 09:55.56 | funkykiwy | thks you |
| 09:56.11 | funkykiwy | I like to ask things and being insult |
| 09:58.44 | orn | lol |
| 09:58.47 | orn | this is priceless :) |
| 10:00.05 | funkykiwy | and ? if you want to see all my config file maybe you can ask ? |
| 10:02.04 | orn | kaldemar asked you to show you the output of 'dahdi show status'. do that command in the asterisk CLI and put the output on pastebin. |
| 10:04.23 | funkykiwy | for 2 line i should put it on pastebin ? but if i can read " A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel" |
| 10:04.35 | funkykiwy | and the result is less than tree line |
| 10:04.48 | funkykiwy | Wildcard TDM410P Board 1 OK 1 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 10:04.54 | funkykiwy | isn't it ? |
| 10:06.06 | funkykiwy | http://pastebin.ca/1416232 |
| 10:06.29 | funkykiwy | so for one line I thought that it was not a necessity to put it on pastbin |
| 10:07.01 | orn | you were right in assuming so :) |
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| 10:11.36 | funkykiwy | happy to here :) |
| 10:13.15 | funkykiwy | but you don't have any idea ? |
| 10:13.36 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 10:13.45 | Kernel_Core | hi all |
| 10:15.12 | funkykiwy | hi kernel |
| 10:15.27 | funkykiwy | I'm pretty sure that you've got my solution :D |
| 10:16.02 | funkykiwy | I got a 4fxo port TDM410P |
| 10:16.19 | funkykiwy | and asterisk 1.6.1 and dahdi 2.2.1-rc1 |
| 10:16.30 | funkykiwy | and asterisk 1.6.1 and dahdi 2.2.0-rc1 |
| 10:16.49 | Kernel_Core | then ? |
| 10:16.50 | funkykiwy | and when I make call trouht DAHDI |
| 10:16.57 | funkykiwy | I got and beautiful alarm |
| 10:17.06 | funkykiwy | As you can see here http://pastebin.ca/1416205 |
| 10:17.39 | funkykiwy | http://pastebin.ca/1416232 result of dahdi show status command |
| 10:17.41 | orn | funkykiwy: sorry, I have no idea. I have never used E1s with Asterisk |
| 10:17.47 | funkykiwy | :D |
| 10:17.51 | funkykiwy | to bad |
| 10:18.21 | Kernel_Core | funkykiwy: first of all check your cable |
| 10:18.58 | funkykiwy | all ready check, and they work before I change my configuration ... |
| 10:18.59 | Kernel_Core | funkykiwy: did you set FXSKS properly in /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf ?! |
| 10:19.05 | funkykiwy | yes |
| 10:19.12 | funkykiwy | I paste it |
| 10:19.38 | Kernel_Core | funkykiwy: issue dahdi_tools |
| 10:20.03 | Kernel_Core | and select your TDM410P |
| 10:20.08 | Kernel_Core | it should be 4/4/4 |
| 10:20.43 | funkykiwy | 4/4/2 but I got only 2 cable plug |
| 10:21.36 | Kernel_Core | funkykiwy: change the position of your cables ! |
| 10:21.41 | funkykiwy | ok |
| 10:22.40 | funkykiwy | well done |
| 10:23.03 | Kernel_Core | did it workd ? |
| 10:23.31 | funkykiwy | no |
| 10:23.41 | Kernel_Core | ! |
| 10:23.53 | funkykiwy | here my system.conf |
| 10:23.54 | funkykiwy | http://pastebin.ca/1416250 |
| 10:24.26 | Kernel_Core | paste me /etc/asterisk/chan_dahdi.conf |
| 10:26.47 | funkykiwy | yeah I enlight it... to much comment |
| 10:27.08 | funkykiwy | http://pastebin.ca/1416252 |
| 10:32.33 | Kernel_Core | funkykiwy: this problem once happend to me , and I removed the Rj45 socket and changed it with a new socket and everything became Okey ! |
| 10:32.57 | funkykiwy | you remove what ? how to ? |
| 10:33.46 | Kernel_Core | removed the RJ45 Socket |
| 10:34.08 | funkykiwy | wtf is this :D (sorry i'm a bit noob in network) |
| 10:34.21 | funkykiwy | I 'll searsh for some docs but |
| 10:34.47 | funkykiwy | is it an asterisk or system ? |
| 10:37.55 | funkykiwy | <PROTECTED> |
| 10:38.10 | funkykiwy | sorry |
| 10:38.14 | funkykiwy | ^^ |
| 10:38.24 | funkykiwy | i'm french |
| 10:39.07 | Kernel_Core | funkykiwy: this is a Physical Error and is related to your line |
| 10:39.41 | funkykiwy | ok, so if I unplugged rj45 and plugged it again ? |
| 10:40.06 | Kernel_Core | funkykiwy: maybe it solves the problem but I think you need to change your cable |
| 10:40.22 | funkykiwy | rj45 ? ok I'll think about it |
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| 10:41.38 | funkykiwy | how do you whisper ? :D |
| 10:48.23 | Great_Anta_Baka | if i want to enable faxing on an extension.. do I enable passthrough on the SIP trunk or on the extension? |
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| 10:54.00 | krdian | hi |
| 10:55.47 | bobsaccamano | hi im using a simple GoToIf condition for a call forwarding on busy feature for sip calls...but it aint working..any bright ideas? |
| 10:57.58 | krdian | does anybody using linksys SPA922 ? |
| 11:03.53 | funkykiwy | yeah |
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| 11:09.04 | funkykiwy | byebye |
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| 11:56.48 | joseph__ | guys any one worked before with app_dial.c file |
| 11:57.58 | gordonjcp | buggery |
| 11:58.01 | gordonjcp | I've bricked my 4602 |
| 11:58.10 | gordonjcp | anyone with Avaya wisdom kicking about? |
| 12:01.09 | funkykiwy | hello I'm back because my problem evolved |
| 12:01.15 | funkykiwy | in the bad way in fact |
| 12:01.22 | funkykiwy | now when I boot |
| 12:01.34 | funkykiwy | No fxo card is connected |
| 12:03.23 | RypPn | gordonjcp can you get to its web interface? |
| 12:04.44 | gordonjcp | RypPn: hi |
| 12:04.45 | gordonjcp | nope |
| 12:05.06 | gordonjcp | haven't been able to get a web interface since upgrading to SIP 2.2.2 firmware |
| 12:05.54 | gordonjcp | then I made the terrible mistake off booting it off the wrong TFTP server, which promptly tried to put SIP 1.1 back on it |
| 12:05.59 | RypPn | seen this? http://support.avaya.com/elmodocs2/4602/16_300158_1_1.pdf |
| 12:06.13 | RypPn | 5 thro 7 may be of use |
| 12:06.31 | gordonjcp | it doesn't get that far |
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| 12:08.47 | RypPn | gordonjcp what do you see when it boots up froma cold start? |
| 12:09.56 | gordonjcp | RypPn: it displays "Initializing...", then "Loading... 323tosip_1_1.bin" then it sits on "Downgrading to 1.8.2..." for several minutes, then reboots back to the start |
| 12:10.09 | gordonjcp | at no point does it seem to look for anything off the network |
| 12:10.35 | joseph__ | any one read app_dial.c |
| 12:11.37 | gordonjcp | there's a JTAG interface and a UART interface on the board, so presumably if I had an image or the pinouts for the UART connector I could poke it |
| 12:12.33 | RypPn | Is this on * or accs? |
| 12:12.44 | gordonjcp | * |
| 12:16.19 | RypPn | I've seen that before on cisco's where you wonder why it wont complete, are you watching RRQ's from the tftp to ensure you have all the requested files? |
| 12:17.06 | gordonjcp | it doesn't hit the network at all |
| 12:17.23 | gordonjcp | or at least, wireshark doesn't see anythign |
| 12:17.25 | gordonjcp | *thing |
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| 12:19.49 | RypPn | if you remove it from the network and powercycle it, do you get press * to program? |
| 12:20.55 | gordonjcp | nope |
| 12:21.05 | gordonjcp | just what I said earlier |
| 12:21.35 | RypPn | so it does the init, load, reboot dance? |
| 12:21.42 | gordonjcp | yup |
| 12:21.42 | RypPn | even with no lan cable |
| 12:21.45 | gordonjcp | yup |
| 12:22.10 | gordonjcp | okay, I've found out which pins are ground and TXD on the UART port |
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| 12:23.19 | RypPn | with no lan cable, are you actually disconnecting the power cable? |
| 12:24.03 | gordonjcp | no, I have a cable from the phone to the PoE brick, then the cable from the PoE brick to the switch unplugged |
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| 12:26.26 | RypPn | how is that off-lan on boot then? If you're using the cable for power and connectivity? |
| 12:26.39 | RypPn | tried mute 73738# ? |
| 12:27.41 | gordonjcp | RypPn: tried mute 73738#, no go |
| 12:27.55 | gordonjcp | it's off-lan on boot because the PoE brick isn't connected to the LAN... |
| 12:28.53 | gordonjcp | ie. I am sitting here with a phone, a cable from the network socket on the phone to the phone socket on the PoE brick, a power lead connected to the PoE brick, and no ethernet cable from the PoE brick to the switch |
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| 12:32.36 | RypPn | I've relayed all this to an avaya engineer work next to and he can't suggest anything else unfortunately |
| 12:35.32 | gordonjcp | okay |
| 12:35.49 | gordonjcp | RypPn: does he/she know anything about the serial port on the bottom? |
| 12:36.34 | gordonjcp | mmm, it runs VxWorks |
| 12:36.58 | RypPn | nah, the prob is our kit in gnk is all tdm |
| 12:37.48 | gordonjcp | wonders if Avaya are likely to be helpful |
| 12:37.50 | gordonjcp | (probably not) |
| 12:38.08 | RypPn | they've got the 6402's down in nhbr, but they have no techs down there, lol |
| 12:38.47 | gordonjcp | if nothing else I can see about getting one off eBay and pulling an image off its flash |
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| 12:47.09 | funkykiwy | hello |
| 12:47.18 | funkykiwy | I got an IRQ conflict |
| 12:47.29 | funkykiwy | Is anybody here know how to fix it ? |
| 12:52.52 | *** join/#asterisk ITguru (n=ITGuru@webfax.impactteachers.com) |
| 12:53.02 | RypPn | gordonjcp the last suggestion is to put all the old firmware on the tftp, not sure how that helps when its not talking to the tftp server, but there you have it |
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| 12:57.57 | gordonjcp | RypPn: tried that, left it for a while, I'll come back in a bit and see if it's helped |
| 12:58.17 | RypPn | fingers crossed :) |
| 12:59.03 | *** join/#asterisk tryfan (n=tryfan@i216-58-2-4.cybersurf.com) |
| 12:59.15 | gordonjcp | it's still scrolling the VxWorks screen every few minutes |
| 12:59.44 | gordonjcp | 3.3v, 115200 8N1 on pin 3, pin 4 is ground |
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| 13:01.07 | tryfan | Hey All, I've got what * says is a syntax error in my ael script. error and only place where the error could possibly be is in: http://pastebin.com/d265e8a94 |
| 13:02.17 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 13:02.42 | tryfan | more like, it's the only place in the script where there is an =911 |
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| 13:14.08 | Great_Anta_Baka | hey guys.. I have it set so that there are three different pickupgroups in the office. how do i make it that the reception is only available for pickup if they dial say *8900 (9000 being reception)? |
| 13:14.56 | Great_Anta_Baka | I have normal call pickup working |
| 13:15.09 | Great_Anta_Baka | they just dont want to accidentally pickup receptions calls |
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| 13:23.31 | Great_Anta_Baka | :/ :\ |
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| 13:29.45 | eppigy | hello |
| 13:29.47 | eppigy | i am dave |
| 13:30.40 | UQlev | I am web_dav ;) |
| 13:30.45 | jaytee | \o/ DAVE! |
| 13:31.59 | eppigy | :D |
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| 13:54.41 | funkykiwy | hello everyone |
| 13:55.03 | funkykiwy | I got an IRQ conflict with my TDM410P is someone know a solution ? |
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| 13:56.25 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 13:56.46 | SuPrSluG | try another pci slot is the easiest |
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| 13:58.59 | tleuthauser | Can anyone offer some help on faxing through an ATA? |
| 14:01.24 | SuPrSluG | what's the issue? |
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| 14:01.48 | funkykiwy | ok but i allready try and it's worst :D |
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| 14:02.09 | funkykiwy | I can't make a certain compilatiion of dahdi to change the IRQ ? |
| 14:02.27 | SuPrSluG | does your bios allow you to set irq's for certain items |
| 14:02.44 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 14:03.36 | SuPrSluG | also try shutting off un-needed items such as parallel ports etc not in use |
| 14:04.33 | funkykiwy | It's a hp server, (not mine) and it don't really contain useless thing... |
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| 14:06.24 | funkykiwy | last time I touch a BIOS the computer almost explod :D i'm affraid about that :D |
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| 14:11.28 | xheliox | When a park call times out, it returns to the extension that parked the call, but what happens if they don't answer? |
| 14:11.37 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 14:11.43 | xheliox | Or rather, what extension/context does it call? |
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| 14:12.40 | *** join/#asterisk jnfuller (n=jnfuller@d64-180-203-35.bchsia.telus.net) |
| 14:13.51 | eppigy | .win 32 |
| 14:13.58 | jnfuller | hey does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? |
| 14:15.10 | jnfuller | I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. |
| 14:15.51 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-d94bab22b98d0fd1) |
| 14:16.35 | *** part/#asterisk tleuthauser (i=user@travis.broadbandip.net) |
| 14:17.37 | funkykiwy | no idea sorry |
| 14:17.42 | SuPrSluG | xheliox:http://www.voip-info.org/wiki/view/Asterisk+call+parking |
| 14:18.24 | xheliox | A. That doesn't work. B. That doesn't answer my question. |
| 14:18.53 | xheliox | C. I know how to Google. |
| 14:18.53 | xheliox | :) |
| 14:18.57 | *** join/#asterisk missinglink (n=missingl@CPE-58-161-41-204.nsw.bigpond.net.au) |
| 14:19.33 | jnfuller | what was the question? I came in late. |
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| 14:19.50 | *** join/#asterisk missinglink (n=missingl@CPE-58-161-41-204.nsw.bigpond.net.au) |
| 14:25.59 | funkykiwy | hello, today is a bad day. when I reboot, i need to unplugged and repluged the line on my fxo port to get them work any idea ? |
| 14:27.10 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 14:29.41 | jaytee | I've been advised that I should eliminate the mouth breathing room temperature I.Q. assclown that was "mowing" his 8 ft X 8 ft fenced in "yard" with a weed whacker at 10PM last nite for the sake of the human gene pool. |
| 14:29.59 | funkykiwy | is it a sign of bad installation ? |
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| 14:49.07 | Alborracho | Hi everyone |
| 14:49.08 | KavanS | jaytee, cement his ass....then rent a boat, and let him go |
| 14:50.39 | Alborracho | does anyone know how to change a port destination number in ACK? im currently sending the ACK message via 65530 |
| 14:54.01 | gordonjcp | meh, 10pm isn't that late |
| 14:54.05 | *** join/#asterisk kash (n=ptx0@2001:5c0:1000:a:0:0:0:295) |
| 14:54.25 | gordonjcp | jaytee: I'll swap you for the people two doors along with the noisest Yorkshire Terriers in the world |
| 14:55.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:57.42 | PinkFreud | jaytee: I'll swap you for the nighbors kids who keep blocking my door with their bikes, and The Banshee, some local kid whosse primary method of communication is extremely high-pitched screams |
| 15:02.27 | Brixius | do comments in ael have to start at the beginning of a line or can I do "x=1; //this is a comment" on the same line? |
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| 15:02.57 | *** join/#asterisk bbryant (n=brett@68.59.20.153) |
| 15:04.26 | KavanS | PinkFreud, just perform a post-birth abortion on the screaming one |
| 15:04.38 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:04.51 | Zeeek | Risc for Asterisc |
| 15:04.56 | Zeeek | coming soon |
| 15:07.09 | PinkFreud | KavanS: I've no clue where The Banshee is. It's so high pitched, that I don't know what direction they're coming from. |
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| 15:11.49 | Carlos_PHX | Anyone recommend a consumer/SMB router with QoS or standalone QoS device? Something cheap for small (3-6 phone) locations? |
| 15:12.46 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:12.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:14.28 | Zeeek | hey leifmadsen |
| 15:15.21 | jnfuller | Carlos_PHX, wrt-54g with tomato firmware flash has a good qos engine |
| 15:15.28 | *** join/#asterisk pta200 (n=paolo@69.38.173.162) |
| 15:15.59 | pta200 | Anyone know if and when Asterisk will support IPV6? |
| 15:16.13 | bmoraca | Carlos_PHX: ASA5505 10-user |
| 15:16.39 | curious101 | hi. anyone knows if a payload can be set on G729 codec in Asterisk? Or do I have to recompile? |
| 15:16.44 | Great_Anta_Baka | I have a client who complains that his calls to the US gets droppped after 5 mins. Is there a number that I can call that I can test this out with just to see the CLI when the call drops |
| 15:16.53 | SuPrSluG | PinkFreud: get him a Tin Drum lol |
| 15:17.24 | Qwell | curious101: huh? |
| 15:17.45 | Carlos_PHX | The ASA is a pricey little thing for a small office. |
| 15:18.08 | Brixius | curious101: due to patents, g729 must be licensed and you can get a license from digium |
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| 15:18.10 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:18.16 | bmoraca | Carlos_PHX: you can do it cheap or you can do it right. $350 is a small price to pay for not having to worry about it again. |
| 15:18.40 | Carlos_PHX | Yes, it's great, but if the customer won't buy it, won't help much. |
| 15:18.58 | curious101 | Qwell: "payload" seems to be a parameter that can be set. I need to set it to 20bytes. but I think there's no command to set that in Asterisk |
| 15:19.16 | curious101 | maybe it's hard-coded into the code, but I'm not sure. |
| 15:19.29 | bmoraca | Carlos_PHX: present him with the scenario that if the cheap router breaks or has problems, it will cost him more in new parts and labor to replace it than the difference in the price the first time. |
| 15:19.53 | curious101 | Brixius: yes, we purchased licenses from Digium. |
| 15:20.32 | Carlos_PHX | bmoraca: I understand totally, yet most customers just won't do it. |
| 15:20.39 | SuPrSluG | pta200:it's try http://www.asteriskv6.org or freeswitch for IPv6 |
| 15:21.18 | pta200 | yeah I looked at both |
| 15:21.45 | Carlos_PHX | We were looking for something that's a no-brainer for everyone to use. |
| 15:21.50 | Carlos_PHX | Or even provide it free. |
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| 15:22.57 | Brixius | curious101: ok, didn't understand the question, thought you were asking how you set asteisk to use g729 :) |
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| 15:24.39 | bmoraca | Carlos_PHX: don't give them an option. tell them they need to do it this way or you won't support it. |
| 15:24.44 | curious101 | Brixius: I see. It also seems to be a version compatibility issue- version 1.4.24. The last version we had where that codec worked was 1.4.22 |
| 15:24.58 | bmoraca | Carlos_PHX: SOHO gear in a business environment is a recipe for disaster. even if it is "easy" to use. |
| 15:25.14 | Carlos_PHX | bmoraca: Then they just won't buy, so that's one solution to the problem. |
| 15:25.32 | rob0 | I have an idea!! Wave a magic wand!!! |
| 15:25.39 | bmoraca | Carlos_PHX: is this a hosted PBX environment or is the PBX onsite as well? |
| 15:25.46 | Carlos_PHX | All of our small office customers are on SOHO gear with excellent success other than lack of QoS. |
| 15:25.50 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 15:25.50 | Carlos_PHX | Hosted |
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| 15:25.58 | *** part/#asterisk pta200 (n=paolo@69.38.173.162) |
| 15:27.15 | bmoraca | Carlos_PHX: QoS is worthless. you have no control over the traffic once it leaves the office. the best bet is to get a router that can handle the load properly. an ASA5505 is that. you can hack things together all you want with 3rd party firmwares and what not, but the end result is that you will have problems eventually. |
| 15:27.45 | Carlos_PHX | QoS is not worthless when the bottleneck is the local outbound speed. |
| 15:28.17 | Carlos_PHX | I know it doesn't fix every problem, that's the largest one. |
| 15:29.04 | *** join/#asterisk coppice (n=chatzill@61.12.13.35) |
| 15:29.06 | Carlos_PHX | It's a minor problem where an inexpensive solution is a no-brainer, but expensive ones are probably not worth it. |
| 15:29.22 | bmoraca | Carlos_PHX: that's not technically QoS. it's prioritization, which is something all together different. and still, the best bet is to get a router that doesn't need hacks like that. if you want a cheap router that may or may not work long term, use something like an ASUS WL520-gU flashed with DDWRT. if you want something that will never need to be touched, use an ASA5505 |
| 15:29.29 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:29.44 | bmoraca | Carlos_PHX: there is no "inexpensive solution". or, rather, the ASA IS that solution |
| 15:29.47 | Carlos_PHX | That's why I was here asking if someone had a SOHO recommendation. |
| 15:30.22 | *** join/#asterisk Amazinzay (n=ijensen@67.108.187.186.ptr.us.xo.net) |
| 15:30.49 | Amazinzay | is there a way to check if libpri is installed? |
| 15:31.13 | bmoraca | Carlos_PHX: a SOHO router with a third-party firmware is not a solution to the problem. it's a bandaid that may or may not fall off in an undetermined amount of time. if phone outages is something the customer can deal with, then use that. if not, make them pay for the right solution. you don't screw around with layers 1-3 in a business. downtime is too expensive. |
| 15:32.05 | Carlos_PHX | I'm looking for a bandaid. The problem is too small to be a big deal. If it can be solved with cheap off the shelf hardware good, if not I'll ignore it. |
| 15:32.10 | bmoraca | Carlos_PHX: in addition, when they have downtime due to an improper selection of equipment, they WILL blame it on you and it will reflect badly on your service in their eyes. |
| 15:32.25 | Carlos_PHX | Hahaha, yeah, I'm not exactly new to this. |
| 15:33.00 | bmoraca | Carlos_PHX: i never said you were. how much upstream bandwidth are you talking about? |
| 15:33.12 | Carlos_PHX | We're not talking about the enterprise here, just the small office where nobody cares if there's a little downtime. But we see near-zero downtime with even just a WRT54G |
| 15:33.19 | Carlos_PHX | Generally 896k up. |
| 15:33.22 | riddlebox | I am builiding a test asterisk system, is 1.6.0.x going to be the stable branch of 1.6? or should I start testing 1.6.1.x |
| 15:33.23 | Carlos_PHX | Cheap DSL |
| 15:33.48 | bmoraca | Carlos_PHX: for 6 phones? even if everyone was on the phone with g711, you wouldn't have a problem. that's more than enough. |
| 15:33.50 | Carlos_PHX | All I'm looking to do is WRT + a bit of prioritization. |
| 15:34.32 | Carlos_PHX | And hoping for off the shelf, because I truly don't want to support DD-WRT. |
| 15:35.44 | riddlebox | I use dd-wrt on my wrt-54g |
| 15:35.44 | bmoraca | Carlos_PHX: you can try the WL-500g v2 Premium from ASUS. it supports some prioritization. it's not as cheap as a Linksys router, though. And, again, it may or may not give you the results you're looking for (this will be the case for ANY router marketed toward SOHO). |
| 15:37.41 | Carlos_PHX | I'll take a look at that. It's a start anyway. Pretty much just looking for something that will take a look at that 896k and make sure the voice packets go first. |
| 15:37.48 | funkykiwy | how to unistall dahdi ? |
| 15:37.55 | funkykiwy | and libpri and asterik ? |
| 15:38.31 | bmoraca | Carlos_PHX: the WL-500g v2 premium claims it can with stock firmware. it's about the most powerful SOHO router you can get, as well. |
| 15:38.41 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:39.13 | jameswf | leifmadsen: do you watch southpark |
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| 15:43.03 | Carlos_PHX | bmoraca: Looks pretty good, thanks. |
| 15:44.03 | Carlos_PHX | Pirce is right in range with any small office router. |
| 15:44.20 | *** part/#asterisk Amazinzay (n=ijensen@67.108.187.186.ptr.us.xo.net) |
| 15:51.14 | Zeeek | join #voip-users-conference |
| 15:51.52 | ariel_ | #voip-users-conference |
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| 15:56.22 | leifmadsen | jameswf: a bit |
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| 16:00.18 | jnfuller | I talked to a guy from digium on asterisk-users@lists.digium.com and he provided some good answers to my earlier question. If anyone is interested his response is in Digest, Vol 58k, Issue 19. |
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| 16:31.14 | jnfuller | bye all |
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| 16:54.12 | *** join/#asterisk elmer_ (n=jason@208.176.185.156.ptr.us.xo.net) |
| 16:55.15 | elmer_ | im having problems sending email through asterisk, specifically for voicemail |
| 16:55.35 | elmer_ | i can send email through sendmail via the command line, but cant seem to get them to send via asterisk |
| 16:58.48 | elmer_ | nothing in the maillog to indicate its sending |
| 16:59.07 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:10.34 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 17:12.45 | rob0 | That would probably be a question for your MTA. Perhaps, however, you need to set a mailcmd in voicemail.conf? It will be impossible to debug without logs. When you use sendmail directly, is it logged? |
| 17:15.04 | elmer_ | yeah it is, i can see it write to the log while tailing the file |
| 17:15.28 | elmer_ | i dont think its an MTA issue since i can send email from the command line just fine, right? |
| 17:15.38 | elmer_ | that would tell me its an asterisk config or permission issue |
| 17:17.03 | elmer_ | at least thats what i think |
| 17:18.37 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 17:21.00 | rob0 | ;mailcmd=/usr/sbin/sendmail -t |
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| 17:21.13 | rob0 | I would guess that's the default. |
| 17:21.30 | rob0 | (That's out of the voicemail.conf sample.) |
| 17:22.38 | elmer_ | yeah, thats the default |
| 17:22.47 | elmer_ | ive tried a lot of different things for that command |
| 17:22.57 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:25.42 | Shazaum_away | hellow, I have questions about details of "queues"... (" W:0, C:210, A:34, SL:67.6% within 12s ") |
| 17:25.43 | Shazaum_away | this information, what are these acronyms? |
| 17:26.44 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 17:26.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:27.06 | *** join/#asterisk ks3 (n=ks3@74.203.195.1) |
| 17:27.31 | ks3 | Is there a dialplan function to check if an extension exists in a given context? |
| 17:30.34 | leifmadsen | ks3: in 1.6.x yes |
| 17:30.41 | leifmadsen | ks3: I think it's like... CHECK_EXTEN() or something |
| 17:30.53 | leifmadsen | probably listed in CHANGES file |
| 17:31.07 | leifmadsen | or... ISEXTEN(), can't quite remember |
| 17:31.18 | ks3 | leifmadsen: I'll search around for those... thanks |
| 17:31.40 | leifmadsen | ok, I was sufficiently bothered to look :) |
| 17:31.41 | leifmadsen | 1.6.1 |
| 17:31.42 | leifmadsen | * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for |
| 17:31.42 | leifmadsen | <PROTECTED> |
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| 17:36.09 | [TK]D-Fender | leifmadsen: Well there was a way before.... but this is hopefully better |
| 17:36.57 | elmer_ | so i changed the mailcmd in voicemail.conf to just touch a file in /tmp and it didnt do that when i left a voicemail |
| 17:37.28 | elmer_ | maybe im not doing it right. that should create the file when a voicemail is left right? |
| 17:37.39 | [TK]D-Fender | elmer_: And I don't see your configs, or information as to what user * is running as, etc. |
| 17:37.41 | [TK]D-Fender | ~pb |
| 17:37.42 | 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/ |
| 17:37.44 | [TK]D-Fender | ^^^ |
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| 17:43.41 | *** join/#asterisk desdesdesdes (n=kg@196.211.34.2) |
| 17:43.48 | desdesdesdes | hi all |
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| 17:45.46 | elmer_ | [TK]D-Fender: here is my voicemail.conf |
| 17:45.46 | elmer_ | http://pastebin.com/me1a4b25 |
| 17:45.47 | [TK]D-Fender | elmer_: Not enough to confirm anything. mask ONLY passwords |
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| 17:50.23 | *** join/#asterisk kuku1 (n=ingo@c-98-227-117-244.hsd1.il.comcast.net) |
| 17:51.05 | kuku1 | Why is it that if I ping the asterisk box, I get a 15sm reply, but when I do sip show peers, the status is OK(161 ms). Why is the that value so large ? |
| 17:52.54 | [TK]D-Fender | kuku1: Because that is an application response time which also depends on its pioritization of responing to OPTIONS packets |
| 17:53.11 | kuku1 | Is there a way for me to bring down that value ? |
| 17:53.21 | [TK]D-Fender | kuku1: Qualify is not a reliable method of guaging latency |
| 17:53.25 | kuku1 | Its the same with with 1.4 and 1.6 |
| 17:53.38 | *** join/#asterisk infomomo (n=acer@pf.cmontmorency.qc.ca) |
| 17:53.38 | [TK]D-Fender | kuku1: that number does not necessarly mean ANYTHING |
| 17:53.42 | kuku1 | aha |
| 17:53.57 | kuku1 | Is here a number I should be looking at ? |
| 17:54.23 | infomomo | hey guys, anyone has an idea how to add Option 120 on windows server DHCP |
| 17:54.44 | kuku1 | infomomo: you right click on the range |
| 17:54.52 | kuku1 | on the pool that is |
| 17:55.08 | rob0 | Hmmm, Windows server DHCP, wouldn't that be a question for MS support? |
| 17:55.08 | infomomo | kuku1: option 120 is not there |
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| 17:55.21 | [TK]D-Fender | kuku1: What is your problem? |
| 17:55.31 | infomomo | rob0: actually option 120 is SIP server option |
| 18:00.01 | jaytee | infomomo, are you trying to setup a boot option for phone provisioning? |
| 18:02.14 | infomomo | jaytee: kind of :S, i need to add Option 120 (SIP server) to my dhcp scope |
| 18:02.51 | jaytee | MS's list of options for any given address scope don't include Option 120 |
| 18:03.19 | elmer_ | [TK]D-Fender: here is the full voicemail.conf, minus some extensions. were using extension 2000 to test this, which is included in the pastebin |
| 18:03.19 | infomomo | jaytee: no they dont |
| 18:03.22 | elmer_ | http://pastebin.com/m4ce352b7 |
| 18:03.37 | infomomo | jaytee: do you have idea ? |
| 18:04.52 | jaytee | infomomo, I just set the boot option 66 to point to my server to use for provisioning my Polycom phones. The rest including the "SIP" Server option (my * server's IP address) is set in the Polycom config files. |
| 18:05.25 | infomomo | jaytee: ok , thanks |
| 18:05.49 | jaytee | infomomo, you're welcome |
| 18:06.44 | infomomo | jaytee: i really needed to add the 120 option to my scope for my config :( LOL |
| 18:08.11 | Brixius | I'm rewriting a dialplan that an ex-employee wrote and I believe I can change SET(var2=${MATH(${var1}-1,int)} to var2=${var1} - 1 but just wanted to get a second on that. |
| 18:08.35 | Brixius | opps, that should rewriting a dialplan in ael |
| 18:09.31 | Brixius | opps^2, that should say "rewriting a dialplan an ael" :) |
| 18:11.57 | KyleK | infomomo: cant you add arbitrary options to it? |
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| 18:19.03 | leifmadsen | Brixius: unless he specifically needed it changed to type 'int' to strip off any trailing decimal places |
| 18:19.47 | leifmadsen | Brixius: also, the format would be: Set(var2=$[${var1} - 1]) <-- notice the square braces |
| 18:20.18 | leifmadsen | otherwise you'd literally be assigning '${var1} - 1' to the variable (where ${var1} would be replaced with the value of ${var1}) |
| 18:24.01 | elmer_ | so if i change the mailcmd to "touch /tmp/test" in voicemail.conf it should create that file when a voicemail is left, right? |
| 18:26.48 | rob0 | elmer_, good basic idea, but I would make it a script and maybe dump some environment variables. |
| 18:28.08 | Zeeek | check the recording today: http://tr.im/voip (posted in an hour or so) |
| 18:28.13 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 18:30.09 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 18:30.36 | infomomo | KyleK: yes but there are many options i am not aware of? array, bytes, ip adress, etc... |
| 18:31.35 | nny_1 | hmm i am still poking around on this, but facing an issue. I have a system that uses Vitelity for outbound and DIDWW for did inward dialing. It's basically a pure SIP box for making international calls. Everything works great except one number> If I dial in and hit DISA, the number works as epxected, but if I call it from the Dial app i get wierdness. Any advice? |
| 18:32.48 | [TK]D-Fender | nny_1: Yes. Be a little MORE vague than "weirdness". We might actually suspect something isn't working to your satisfaction |
| 18:32.48 | nny_1 | http://pastebin.com/m5694ec1b is my dial stuff |
| 18:33.22 | [TK]D-Fender | nny_1: And that little snippet really doesn't tell us anything |
| 18:34.16 | nny_1 | [TK]D-Fender: yeah heh good point. the console shows the same thing for both calls as far as the sip progression goes. When i use DISA I can hear the ring tone for the number I am calling in Argentina. When I dial it from the macro I posted I get asterisk |
| 18:34.25 | nny_1 | asterisk's ring, and then silence |
| 18:34.28 | nny_1 | but the channel stays open |
| 18:34.29 | nny_1 | http://pastebin.com/m7193ae17 |
| 18:34.33 | nny_1 | has the console output |
| 18:36.07 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 18:37.04 | generalhan | hey all, i was wondering if there is a way to have a caller authenticate using the credentials in agents.conf ? |
| 18:37.13 | *** join/#asterisk tainted_ (n=Administ@67.43.165.100) |
| 18:37.13 | KyleK | infomomo: you'd probably want to add option 120 as a string since its sip server |
| 18:37.55 | nny_1 | [TK]D-Fender: feel free to ask me for other info. I hope that explanation did the situation justice. |
| 18:38.24 | generalhan | i want to have an extension in my dialplan that asks the password associated with that agent number, and without it will not continue |
| 18:38.40 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:38.56 | KyleK | generalhan: you can arbitrarily ask callers for digits in dialplan logic or agi |
| 18:39.48 | generalhan | KyleK: i know that i can ask a caller to enter some digits, what i dont know is how to check what the caller enters against the password stored in agents.conf |
| 18:40.04 | generalhan | ... if that can be done at all. |
| 18:40.29 | KyleK | whats the password in agents.conf? |
| 18:40.34 | KyleK | is it password or 12345? |
| 18:40.52 | elmer_ | so it doesnt look like mailcmd in voicemail.conf is actually running |
| 18:40.59 | elmer_ | what could cause that? |
| 18:41.14 | [TK]D-Fender | nny_1: Where is the call to compare it to? |
| 18:41.32 | generalhan | KyleK: just a normal set of digits, 1050,1234 |
| 18:42.31 | *** join/#asterisk jmodigb (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 18:42.50 | KyleK | generalhan: you could write a small agi script that asks for the digits and then returns differently based on success/fail |
| 18:43.10 | jmodigb | Has anyone had a problem with the dahdi driver loading on 1.6? |
| 18:43.21 | nny_1 | [TK]D-Fender: posting, but now that you mention it, the succesful one gets past the "Â -- SIP/vitel-outbound-0990a8d0 is making progress passing it to SIP/204.11.194.35-09909090" so I guess that points to part of the issue |
| 18:43.23 | nny_1 | Â -- SIP/vitel-outbound-0990a8d0 is making progress passing it to SIP/204.11.194.35-09909090 |
| 18:43.24 | nny_1 | er |
| 18:43.30 | nny_1 | http://pastebin.com/m7824fc7d |
| 18:43.57 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:44.34 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:45.11 | nny_1 | doing a sip debug on run now |
| 18:45.29 | [TK]D-Fender | nny_1: The effective dials look the same |
| 18:45.34 | [TK]D-Fender | (so far) |
| 18:45.36 | generalhan | KyleK: maybe i am not being completely clear. i have an agent, Agent/1050 ... in agents.conf agent 1050's password, or PIN, is 1234. when Agent/1050 dials extension 7000 i want to prompt this agent to enter his password. if the agent enters 1234 then extension 7000 will move on. my question is how can i check the password in agents.conf to verify the pass they entered is the right one ? |
| 18:46.07 | [TK]D-Fender | jmodigb: Well a.6.0.X has been out for a long while now and we're running fine. Perhaps you could provide actual details about your problem, sin it is just you |
| 18:46.44 | *** part/#asterisk tainted_ (n=Administ@67.43.165.100) |
| 18:47.00 | [TK]D-Fender | generalhan: Firs.. AGENTS don't dial, devices do. First you'd have to identify some sort association of the decive being used to call and an agent and then worrk about authenticating |
| 18:47.18 | *** join/#asterisk juancferrer (n=chatzill@dsl238.dsl.netdoor.com) |
| 18:47.28 | juancferrer | I'm making a call file that places a call to a POTS line and starts playing back an audio message to the receiver.... I receive the call about 10 seconds after placing the call file, but by the time I pick up the phone, the message is already being played. Asterisk begins the audio playback as soon as the call is made, and not after the recipient picks up the phone |
| 18:47.42 | generalhan | [TK]D-Fender: i have already figured out how to distinguish which agent is using the device ... what i dont know, is how to have them authenticate against their password in agents.conf |
| 18:47.44 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 18:48.13 | jmodigb | thx D, This is a fresh install, my first with 1.6.0.9. Have dahdi running, getting no errors on the PRI lines, however i get nothing from dahdi in the asterisk CLI |
| 18:48.33 | jmodigb | seems that the dahdi channel driver isn't there |
| 18:48.51 | [TK]D-Fender | generalhan: "core show function AGENT" |
| 18:49.34 | [TK]D-Fender | jmodigb: Perhaps you didn't configure everything for it. Or the ever popular "Didn't compile * AFTER doing DAHDI" |
| 18:50.19 | jmodigb | I did Libpri, dahdi, asterisk then asterisk addons. Probably did miss something, but what?!?! where?!?! |
| 18:50.22 | *** part/#asterisk juanIMP (n=Juancho@190.26.210.241) |
| 18:50.36 | KyleK | "then extension 7000 will move on"? so I punch in 7000 into the phone, it asks me for password, and then calls it? |
| 18:50.36 | elmer_ | are there any other asterisk user forums than the official one? |
| 18:50.45 | jmodigb | no dahdi commands works in the CLI |
| 18:51.23 | [TK]D-Fender | jmodigb: meaning? |
| 18:51.30 | nny_1 | [TK]D-Fender: got it to go through once, and then the next time.. same issue. The audible part of the issue happens like: Dial in to IVR, hit extension for macro. Playsback "pls hold" and then silence for 10 seconds or so. After that I get a ring, but no output on console and the ring sounds like a North American ring. If the call goes through the ring has a beep to it, a South American ring. When I use DISA it seems to go through every time. |
| 18:51.34 | nny_1 | sip debug here http://pastebin.com/m328a82c5 |
| 18:51.54 | nny_1 | may be some of the succesful debug in there, I am ssh'd in and had to copy/paste the output after it failed |
| 18:52.01 | jmodigb | No such command 'dahdi show channels' |
| 18:52.21 | [TK]D-Fender | nny_1: AH, the ringing issue is because DISA actually answers the source channel, and passes INBAND progress to it with * indications. The direct dial is "passthrough" call progress |
| 18:52.35 | [TK]D-Fender | jmodigb: "module load chan_dahdi.so" |
| 18:53.14 | nny_1 | [TK]D-Fender: i see. the funny thing is the time it was sucessful when i used the macro, the ring was correct |
| 18:54.02 | jmodigb | Eureka! thx, D is see the error |
| 18:54.32 | generalhan | [TK]D-Fender: you're a genious ! |
| 18:54.37 | generalhan | [TK]D-Fender: thanks a million ! |
| 18:55.04 | nny_1 | [TK]D-Fender: er wait hold the phone, DISA just failed now too |
| 18:55.49 | nny_1 | [TK]D-Fender: hmm one sec I have a suspicion |
| 18:56.22 | kc8pxy | [TK]D-Fender: I'm trying to compile spandsp to play with app_fax, but spandsp keeps failing, the assembler keeps saying it's getting bad code. |
| 18:56.57 | kc8pxy | [TK]D-Fender: is that sufficiently off-topic to where i should ask elsewhere? |
| 18:58.56 | bmoraca | kc8pxy: hylafax with iaxmodem is probably simpler to get going... |
| 19:00.47 | nny_1 | [TK]D-Fender: hmm nope. So should I take this up with the provider? Or is there something telling in the sip debug dialog? |
| 19:02.00 | kc8pxy | bmoraca: simpler or better, or both? |
| 19:02.46 | *** join/#asterisk micols (n=mio@81.161.188.225) |
| 19:09.30 | jaytee | someone reported a problem calling out, I tested dialing the same number out our PRI and got a cause 127. which is "intel-working, unspecified". Anyone ever seen that before? |
| 19:10.19 | jaytee | pretty sure it's not our PRI that's the problem. calling the same number from a cell phone I get one ring and then silence but not disconnected. |
| 19:13.25 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:21.31 | juancferrer | Why does playback begin immediately when a call is placed, and not when the recipient picks up the phone? |
| 19:21.57 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 19:24.13 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:25.32 | *** join/#asterisk sumasuma (n=chinnapa@cpe-98-154-108-70.socal.res.rr.com) |
| 19:25.48 | *** join/#asterisk coppice (n=chatzill@61.12.13.35) |
| 19:26.21 | sumasuma | What is the best linux edition to run enterprise linux ? |
| 19:26.27 | sumasuma | What is the best linux edition to run enterprise asterisk ? |
| 19:27.34 | Nugget | Linux is Linux is Linux. Use whichver flavor you hate the least. |
| 19:28.16 | pmhaddad-work | for enterprise solutions i personally run * on hardened Debian or Ubuntu servers |
| 19:28.44 | rob0 | Any OS in an enterprise environment should be managed by qualified professional administrators, too. |
| 19:28.50 | bmoraca | kc8pxy: probably both. |
| 19:30.06 | bmoraca | juancferrer: that's the way Asterisk works with analog channels. it doesn't have any way of knowing when the other side actually picks up the phone. |
| 19:30.59 | juancferrer | bmoraca: is there anything i can do? |
| 19:31.50 | bmoraca | juancferrer: aside from getting a digital connection of some kind (PRI, SIP, etc), you could try the answering machine detection function. |
| 19:33.44 | juancferrer | bmoraca: sipphone is my trunk ,but i'm calling POTS lines... I see that it starts playback when sipphone answers the channel |
| 19:35.30 | nny_1 | [TK]D-Fender: btw it seems the issue was with the provider, thanks for the help either way |
| 19:36.08 | juancferrer | I had it call me and begin recording, and when I played the recording back there were about 8 seconds of silence, and then the ringing |
| 19:36.37 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 19:38.06 | hunmonk | is there a single extension pattern that would match both _NXXNXXXXXX and _1NXXNXXXXXX ? |
| 19:38.57 | bmoraca | hunmonk: probably not. |
| 19:39.05 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 19:39.43 | [TK]D-Fender | hunmonk: no |
| 19:39.52 | hunmonk | bummer |
| 19:41.20 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 19:41.28 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:41.55 | [TK]D-Fender | juancferrer: What playback? Why is something playing back? Where are you showing us your dialplan and CLI output? |
| 19:41.59 | [TK]D-Fender | juancferrer: PASTEBIN <- |
| 19:42.00 | [TK]D-Fender | ~pb |
| 19:42.01 | 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/ |
| 19:42.03 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 19:42.13 | eric_hill | Is there any way to set up a meetme conference dynamically with the manager API? |
| 19:42.54 | bmoraca | eric_hill: isymphony can do it, so i'd suspect the answer to be "yes" |
| 19:43.08 | eric_hill | Will google for that. Thanks. |
| 19:44.56 | *** join/#asterisk ttl- (n=patrick@81.83.174.130) |
| 19:45.12 | *** join/#asterisk ingenius (n=alektro@host63.190-138-44.telecom.net.ar) |
| 19:50.04 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-230-56.dslextreme.com) |
| 19:50.51 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:50.51 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 19:51.00 | juancferrer | http://pastebin.com/m5f930549 |
| 19:51.43 | juancferrer | everything 'works' its just that by the time i pickup, my message playback is already half way finished |
| 19:53.11 | SuPrSluG | juancferrer:use the Wait(3) command in your dialplan |
| 19:54.00 | SuPrSluG | or how long it takes |
| 19:54.39 | jaytee | I figured out what my problem was with an outbound call..... it's because it's naptime. we're not supposed to call them during nap time. |
| 19:54.45 | [TK]D-Fender | juancferrer: use "Playback(silence/2) instead of Wait() |
| 19:55.42 | juancferrer | but i still don't know when the person actually picks the phone up |
| 19:55.48 | *** join/#asterisk Cherebrum (n=Cherebru@209.9.237.93) |
| 19:55.54 | Cherebrum | Can anyone recommend a good SIP DS3 gateway? |
| 19:56.49 | eric_hill | Cherebrum: We're using a Cisco 3745 router with a bunch of 64-channel DSP's and it works very well. |
| 19:58.10 | [TK]D-Fender | juancferrer: Is it still ringing when the playback starts? |
| 19:58.30 | Cherebrum | I don't like Cisco that much. They always break SIP, and RTP. And I always have issues with their DSP firmware doing funny things |
| 19:58.41 | juancferrer | Tried Playback(silence/2), that just waited about 2 seconds, and then did the rest...but by the time i picked up it was already playing back my acknowledge message |
| 19:58.52 | Cherebrum | I prefer Audiocodes, but I'm looking for something cheaper |
| 19:59.07 | Cherebrum | Audiocodes is a~ $100k and only runs on DC power |
| 19:59.30 | eric_hill | We swear by Cisco and haven't had any problems with them once they were operational. YMMV. |
| 19:59.54 | eric_hill | You might try Adtran. They've got quite a bit of SIP gear now. |
| 19:59.57 | juancferrer | from when I drop the call file to when my phone rings it takes ~10 seconds...So i have a Wait(9), my problem is that someone can let it ring a few times, but by that time the rest of my dialplan is already running |
| 20:00.10 | Cherebrum | eric_hill: They don't have anything that can handle a DS3 |
| 20:00.33 | [TK]D-Fender | juancferrer: How fast do you see the ANSWERED message appear as you watched it dial your cell? |
| 20:01.04 | juancferrer | 2 seconds |
| 20:01.32 | [TK]D-Fender | juancferrer: then your provider is answering the call and THEN continuing to terminate your call. You are screwed. |
| 20:01.38 | Cherebrum | hmmm... maybe this guy: http://www.dialogic.com/products/gateways/img_gateways/IMG1010.htm |
| 20:01.45 | eric_hill | Cherebrum: What about getting direct IP from the carrier? A 100MB or Gig-E fiber from your carrier may be cheaper than a DS3... |
| 20:01.48 | [TK]D-Fender | juancferrer: They answer you, and then place their own outbound call |
| 20:01.59 | juancferrer | its sipphone |
| 20:02.10 | juancferrer | Would a different provider work? |
| 20:02.29 | [TK]D-Fender | juancferrer: many other providers pass on progress OOB, this one doesn't |
| 20:03.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:03.28 | juancferrer | any recommendations? I'm in the US |
| 20:04.01 | Cherebrum | eric_hill: It's more expensive to get a DS3 from the carrier when I have to do MPLS and they are telling me they REQUIRE a Acme Packet SBC that costs $80k |
| 20:06.12 | eric_hill | Cherebrum: Time to get a new carrier. That's crap. |
| 20:07.00 | juancferrer | [TK]D-Fender: so sipphone does In band DTMF? |
| 20:08.16 | [TK]D-Fender | juancferrer: this has NOTHING to do with DTMF |
| 20:08.51 | [TK]D-Fender | juancferrer: Sipphone answers ASTERISK and then they place the outbound call. They do not pass on their outbound call's progress to * before answering. |
| 20:09.30 | juancferrer | [TK]D-Fender: so time for a new sip provider? or is there some setting somehwere? |
| 20:09.54 | [TK]D-Fender | juancferrer: What part of SCREWED was not unmistakenly clear? |
| 20:10.12 | [TK]D-Fender | juancferrer: There is no setting. THEY answer you then place the outbound call on their side. |
| 20:10.16 | [TK]D-Fender | juancferrer: You have no chance. |
| 20:11.16 | *** join/#asterisk brunner (n=chris@66.35.172.123) |
| 20:11.25 | brunner | Has anyone here ever worked with a Cisco AS5400? |
| 20:11.32 | *** join/#asterisk davidf88 (n=davidf88@92.16.26.57) |
| 20:12.08 | eric_hill | brunner: Once. What's up? |
| 20:12.37 | brunner | eric_hill: I'm trying to figure out what types of trunk signaling are possible with a T3 |
| 20:13.00 | juancferrer | k, new provider it is |
| 20:13.11 | eric_hill | brunner: Sorry, I only did a bunch of T1's on the 5400, never worked with a T3 there. |
| 20:13.33 | brunner | gotcha, thanks anyway |
| 20:17.05 | juancferrer | [TK]D-Fender: so which providers will work? |
| 20:19.54 | [TK]D-Fender | juancferrer: Can't tell you offhand, I'd ask around. |
| 20:20.37 | juancferrer | [TK]D-Fender: ok, thanks for the help |
| 20:22.03 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 20:22.03 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 20:23.45 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 20:23.54 | seb- | [TK]D-Fender: are you around for a quick question? |
| 20:24.16 | [TK]D-Fender | seb-: shoot |
| 20:24.51 | seb- | [TK]D-Fender: you've helped a lot and thanks to you i think my friggin * server is all secure and debugged now |
| 20:25.20 | seb- | [TK]D-Fender: i think i even got appconference (Xen friendly MeetMe replacement) working |
| 20:25.33 | seb- | [TK]D-Fender: i drop into the conf room w/o any error message AFAIK |
| 20:25.59 | *** join/#asterisk matt_keys (n=matt_key@208.71.232.42) |
| 20:26.03 | matt_keys | I've got NXXNXXXXXX AND 1NXXNXXXXXX as my dial rules. If I dial 7063752165 (date and time) it tells me i need to dial a 1... if I dial 17063752165, it says the call cannot be completed as dialed. |
| 20:26.21 | seb- | [TK]D-Fender: i just need us both to join the "conference" and for you to say something so that i can verify i really did get appconf to work |
| 20:26.38 | seb- | [TK]D-Fender: cuz obviously by myself i don't know...see the problem? |
| 20:27.11 | [TK]D-Fender | matt_keys: "it"? What the hell is "it"? :) |
| 20:27.22 | seb- | the thing |
| 20:27.47 | [TK]D-Fender | seb-: You need friends? :) |
| 20:27.48 | matt_keys | that bitch in the handset.. |
| 20:28.07 | matt_keys | she can never make up her mind :) |
| 20:28.09 | seb- | [TK]D-Fender: err well * wise friends yes |
| 20:28.09 | [TK]D-Fender | matt_keys: Soory, * doesn't come with a bitch in the handset. In fact... * doesn't even come with a handset |
| 20:28.39 | [TK]D-Fender | seb-: PM me the # to call and I'll dial in |
| 20:28.57 | matt_keys | the "you must dial a one" is * ... the call cannot be completed is 'bell |
| 20:28.57 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 20:29.19 | matt_keys | wait, no that's backwards sorry |
| 20:29.21 | [TK]D-Fender | matt_keys: Sounds like the telco doesn't like what you are dialing, so go dial something else |
| 20:29.41 | matt_keys | call cannot be completed is *, must dial a 1 is telco |
| 20:29.46 | jaytee | and hopefully it won't be "naptime" |
| 20:29.48 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 20:30.00 | [TK]D-Fender | matt_keys: Since when does * "talk" to you? |
| 20:30.31 | [TK]D-Fender | matt_keys: pastebin is your friend.... |
| 20:30.33 | [TK]D-Fender | ~pb |
| 20:30.33 | 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/ |
| 20:30.52 | matt_keys | -- Executing [17063752165@from-internal:4] Playback("SIP/0152-b7305b98", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack |
| 20:31.18 | KavanS | so [TK]D-Fender have you met Allison yet? |
| 20:31.20 | [TK]D-Fender | matt_keys: this is your dialplan, why did YOU put that there and wht was it supposed to mean? |
| 20:32.07 | [TK]D-Fender | KavanS: Nope. |
| 20:32.27 | KavanS | roger that |
| 20:32.42 | [TK]D-Fender | KavanS: I've never been to an Astricon, etc, only some smaller local conferences |
| 20:32.51 | matt_keys | N matches any number 2-9... so NXX is area code, NXXXXXX is the 7 digit afterwards |
| 20:33.00 | KavanS | ahh true |
| 20:33.04 | KavanS | I didn't know the turnout to those things |
| 20:33.08 | KavanS | is Astricon huge? |
| 20:33.13 | [TK]D-Fender | matt_keys: Why did you put that line in your dialplan? |
| 20:33.29 | [TK]D-Fender | matt_keys: PASTEBIN <- |
| 20:34.19 | matt_keys | i'm not sure what you're asking |
| 20:34.45 | [TK]D-Fender | matt_keys: This is your dialplan. What is that line supposed to represent in there? |
| 20:35.00 | matt_keys | what line? the NXXNXXXXXX? |
| 20:35.19 | matt_keys | I'm using FreePBX, where it says "Dial Rules". |
| 20:35.19 | [TK]D-Fender | matt_keys>-- Executing [17063752165@from-internal:4] Playback("SIP/0152-b7305b98", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack <-- this lijne of &^#$ing dialplan being EXECUTED |
| 20:35.30 | [TK]D-Fender | matt_keys: GUI's are NOT supported here |
| 20:35.40 | [TK]D-Fender | matt_keys: Go ask in their support channel |
| 20:35.58 | *** part/#asterisk matt_keys (n=matt_key@208.71.232.42) |
| 20:37.50 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:41.50 | ariel_ | is there a support channel for ABE ?? |
| 20:42.22 | [TK]D-Fender | ariel_: No, thats what you pay digium for. |
| 20:43.54 | ariel_ | [TK]D-Fender: yes but I like this type of support system better then having to call them on the phone... |
| 20:44.21 | [TK]D-Fender | ariel_: well they don't have a dedicated channel. General * stuff can still be asked here. |
| 20:44.24 | ariel_ | rpath sucks, (thinking of redoing system) |
| 20:44.42 | [TK]D-Fender | ariel_: And I like getting free cars, just that nobody seems to want to give me any. I guess we're both screwed |
| 20:45.04 | ariel_ | lo |
| 20:45.52 | Carlos_PHX | Hmm |
| 20:46.09 | *** part/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 20:49.43 | ariel_ | OK it's time to reformat my lab server and start over with this ABE. (Thinking of giving up on ABE all together). |
| 20:55.20 | *** join/#asterisk troy- (n=troy@worldnet.tauri.ca) |
| 20:55.40 | bmoraca | ariel_: talking to Digium last year at voicecon in san francisco, they said that AsteriskNOW was a superior product to ABE |
| 20:55.59 | bmoraca | ariel_: if ABE is still on rpath, i'd tend to agree. (asterisknow isn't using rpath anymore) |
| 20:56.44 | jameswf | my nat details now autoupdate... I dont know why I didn't do that before.. |
| 20:58.56 | KavanS | ariel_, rpath is the ghey |
| 21:01.12 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 21:06.52 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:07.57 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 21:14.50 | eric_hill | The asterisk manager API doesn't list the Originate action as an available command (Permission denied). What do I need in the read/write area of manager.conf to get this command available? |
| 21:19.01 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-96-232-153-31.nycmny.east.verizon.net) |
| 21:19.59 | VJFROMGT | I am having issues with zap channel |
| 21:20.18 | VJFROMGT | 2 identical connection on span 1 and span 2 but span 2 keeps saying down |
| 21:20.19 | VJFROMGT | http://pastebin.ca/1416847 |
| 21:21.19 | eric_hill | nm, privilege "originate" |
| 21:22.06 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 21:22.39 | Brixius | I'm trying to convert from pbx_config to pbx_ael I'm having a problem with a macro, I've pasted the conf and ael versions in http://pastebin.com/d1ac31142 can it looks like the first line isn't working in the ael version but does in the conf |
| 21:27.29 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:27.42 | Brixius | I updated the pastebin with the error that is generated: [May 8 16:23:56] WARNING[4553]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '<', expecting $end; Input: |
| 21:27.43 | Brixius | <sip:9522304100@137.192.21.50;isup-oli=0;user=phone>;tag=158932c0+1+554a0001+8d53eb54 |
| 21:27.43 | Brixius | ^ |
| 21:27.43 | Brixius | [May 8 16:23:56] WARNING[4553]: ast_expr2.fl:411 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source. |
| 21:27.43 | Brixius | <PROTECTED> |
| 21:27.55 | Brixius | shoot, sorry didn't mean to paste that there. |
| 21:27.59 | generalhan | anyone know if you can specify an agents password in the agent logon line from the dialplan so that the agent isnt prompted for it ? |
| 21:28.12 | Brixius | http://pastebin.com/d1a5607d9 |
| 21:28.35 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 21:28.37 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 21:31.58 | *** part/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 21:34.22 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 21:35.14 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 21:38.06 | ariel_ | Yipee got 10 new Polycom Spectralink 8030 for our lab and techs.... Push to talk baby..... now to get them setup and configured... |
| 21:43.50 | KavanS | push to talk? |
| 21:43.54 | KavanS | link/summary of features |
| 21:45.56 | hardwire | do you big guys ever deal in daily billing for wholesale traffic? |
| 21:46.16 | bmoraca | yuck! |
| 21:46.19 | hardwire | I'm being told the industry standard for billing is daily when dealing with wholesale traffic, |
| 21:46.34 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:46.44 | hardwire | and I'd just like to say.. I've never seen an organization organized enough to deal with that.. but I have seen the means for people to pull daily reports and evaluate their own usage. |
| 21:47.35 | bmoraca | our wholesale PRI provider bills us monthly for usage. |
| 21:52.21 | [TK]D-Fender | KavanS: www.polycom.com |
| 21:52.57 | KavanS | [TK]D-Fender, nice, that's not the exact link though |
| 21:52.58 | *** join/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com) |
| 21:53.26 | bmoraca | is it so difficult to go through the product page? i found it in like 30 seconds |
| 21:56.54 | [TK]D-Fender | KavanS: This is the "skill testing question" where if you can't navigate their site to find the damn product page yourself you're not WORTHY of our help :) |
| 21:57.11 | KavanS | ok. |
| 22:01.19 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 22:02.58 | *** join/#asterisk Rickardo (n=chatzill@200.204.124.211) |
| 22:03.54 | *** join/#asterisk docid (n=frankie@whthyt253-26.northwestel.net) |
| 22:04.13 | docid | aybody have experence with openvox t1 cards? any reason they should be avoided? |
| 22:04.28 | beek | [TK]D-Fender: You're in rare form today. Feeling better? |
| 22:06.01 | bmoraca | docid: they don't seem to be any cheaper than Sangoma...so...given the choice, I'd take Sangoma |
| 22:06.32 | docid | sangoma 4xt1 is around $1600, openvox 4xt1 is $695 |
| 22:06.46 | docid | i use sangoma in my surrent instalations, and like them very much |
| 22:06.51 | docid | but with that huge gap... |
| 22:06.54 | Rickardo | Hello everybody... i'm using elastix and I have problems to manage conference calls using conference manager... The settings on conference manager doesn't make any effect in the conference. I Can't set up time, PIN numbers.. etc.. |
| 22:07.22 | docid | Rickardo, that would be a question for #elastix, but they are pretty dead right now |
| 22:07.36 | *** part/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 22:07.57 | Rickardo | I Tried #elastix... |
| 22:08.20 | Rickardo | without success |
| 22:08.20 | [TK]D-Fender | Rickardo: GUI's are not supported here |
| 22:08.23 | docid | yes, i know...i saw...ive never been able t get the confrence module to work either |
| 22:08.31 | Rickardo | ahhh.. ok.. |
| 22:08.54 | docid | but i pretty much gave up on most of the gui features, do a;lmost all my work in the confs now |
| 22:08.54 | [TK]D-Fender | beek: Largely, tiny bit lingering and thats it |
| 22:08.55 | bmoraca | docid: there's no substitute for quality. any install that needs 4 PRIs likely won't be bothered by a one time cost of $900. I'd take the more expensive solution that I KNOW works. |
| 22:09.07 | docid | ya do got a point |
| 22:09.39 | [TK]D-Fender | [18:06]<docid>sangoma 4xt1 is around $1600, openvox 4xt1 is $695 <--- Oh... which Sangoma card EXACTLY? |
| 22:09.48 | jaytee | [TK]D-Fender, lingering? did ya have a touch of the swine flu? |
| 22:09.51 | docid | and theres no beating sangomas customer support, |
| 22:09.52 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 22:10.06 | bmoraca | docid: A108D = $4100...yikes! |
| 22:10.08 | docid | ahh, lemme look again |
| 22:10.42 | [TK]D-Fender | jaytee: Finally a disease men can fear. I mean mad cow? No threat... because we're all PIGS : Now the tables have turned... |
| 22:10.55 | [TK]D-Fender | docid: So... which card? |
| 22:10.59 | jaytee | [TK]D-Fender, lol...you and Denny Crane! |
| 22:11.04 | docid | A104 |
| 22:11.23 | [TK]D-Fender | docid: Let me add that only a complete retard would by a card WITHOUT HWEC. |
| 22:11.29 | [TK]D-Fender | :) |
| 22:11.39 | jaytee | mouth breathers abound! |
| 22:11.42 | [TK]D-Fender | docid: Time to increase that amount again : A104d <- d |
| 22:11.47 | beek | evening jaytee |
| 22:11.52 | bmoraca | docid: www.voiplink.com has the A104 for $1400 and the A104D for $2240 |
| 22:11.53 | jaytee | evening beek |
| 22:12.13 | docid | point taken, but we dont seem to get echo on our current install that will be very simular to our coming install |
| 22:12.25 | jaytee | I'd rather spend the extra money on a Sangoma than watch my PRI's ping pong up and down all day on an openvox card. |
| 22:12.47 | docid | aye, cant afford any extra instability over what we have already |
| 22:12.59 | bmoraca | docid: susceptibility to echo depends on your distance from the switch and the line quality between you and the switch. if they're not in exactly the same place, then what you currently have means nothing. |
| 22:13.03 | docid | so there are known issues with the openvox gear? |
| 22:13.04 | [TK]D-Fender | docid: Very similar = prepare for the bullet to enter your skull. |
| 22:13.28 | [TK]D-Fender | docid: Cheap stuff built on old Digiums designs they dropped for a reason |
| 22:13.35 | docid | wow, you are in rare form TK, but thanks for the kik in the pants.... |
| 22:13.48 | jaytee | docid, think of Grandstream for phones and then apply the same level of "quality" to analog and digital cards |
| 22:13.50 | [TK]D-Fender | docid: No, I'm going for the headshot. |
| 22:14.26 | docid | ahh, refer to my revious comment for aiming help...obviously thats where the headshot will be pointed |
| 22:14.45 | jaytee | I feel good now that I know that ISDN cause code 127 means "don't call during nap time!" |
| 22:14.59 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 22:15.01 | beek | jaytee: WTF? |
| 22:16.20 | jaytee | problem this afternoon. woman at my office was convinced there was "something wrong with the phone system". I called the three numbers she gave me. First two were fast busys from the telco. Got the same response on my cell phone too. Third call rings then sounds like it's picked up but no one there. They were answering and then hanging up the call after a few seconds. |
| 22:16.45 | beek | Ahhh... and it returned code 127? |
| 22:16.46 | jaytee | called back twice and finally got someone to speak. They told me, "don't call here now, it's nap time!" |
| 22:16.56 | beek | Now that's funny... |
| 22:16.57 | jaytee | yep |
| 22:17.01 | jaytee | cause 127 |
| 22:17.08 | docid | lolz |
| 22:17.20 | jaytee | I'm used to seeing 16 on normal hangups so this had me going for a bit |
| 22:17.25 | *** part/#asterisk Rickardo (n=chatzill@200.204.124.211) |
| 22:18.07 | jaytee | I think they were putting the phone at their end on hold and then hanging up the reciever |
| 22:18.33 | bmoraca | lol...marketing tripe: "Connecting via USB 2.0 it eliminates the need for intermediating PCI used in other channel banks." as if USB controllers aren't attached to the PCI bus anyway. lolz |
| 22:18.34 | beek | Sounds like an invitation to write an evil script to generate call files... |
| 22:18.47 | jaytee | lol |
| 22:20.22 | bmoraca | dude, i totally want my PSTN connectivity through a USB device. quad T1 PRI in USB2.0 form! |
| 22:24.30 | [TK]D-Fender | Plug&Play? I need this thing to WORK!!!! |
| 22:34.45 | brunner | what kind of equipment would I need to strip off one or two channels from my PRI to use as a BRI? |
| 22:35.41 | bmoraca | i'd imagine asterisk could do that if you had a BRI card. |
| 22:36.59 | KyleK | brunner: what are you doing with the rest of the channels on the PRI? |
| 22:39.16 | brunner | KyleK: it's hooked up to a Nortel PBX thingy right now |
| 22:39.49 | brunner | bmoraca: well that's a good idea... not sure why I didn't think of that |
| 22:40.35 | brunner | hmm... but I need to use the BRI for data, not voice |
| 22:40.53 | brunner | it's for a hardware codec for broadcast |
| 22:43.17 | [TK]D-Fender | brunner: * will not support this... you'll have to find some other kind of device I'm bettting |
| 22:43.32 | bmoraca | brunner: yeah, if you need it for data, you'll need something else |
| 22:44.00 | bmoraca | brunner: not only that, but you can't carve out data channels on a T1 that's configured as a PRI |
| 22:44.49 | bmoraca | brunner: your telco should be able to provision the T1 to provide both. there are devices that can split a PRI off of a data T1. |
| 22:44.56 | bmoraca | brunner: what, exactly, are you trying to accomplish? |
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| 23:09.00 | brunner | bmoraca: I want to continue using most of the PRI channels for the PBX, but use two of the B channels for a broadcast codec so that I can make a 64k data call to a studio that has a BRI and broadcast codec on their end. |
| 23:11.25 | bmoraca | brunner: I don't think you'll be able to do that without the help of the telco. an alternative could potentially be a BRI-to-IP gateway on each end, but i'm not sure of the total setup, so that may not work. |
| 23:12.11 | bmoraca | HAH! first swing back in the big leagues and A-Rod hits a homer! |
| 23:12.40 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 23:16.05 | carrar | ~book |
| 23:16.06 | 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 |
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| 23:17.02 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 23:17.39 | elitecoder | I need to map phone numbers to their registered time zones. Anyone know where I can find the data to do this? |
| 23:17.54 | elitecoder | It's for an auto dialer, to keep from calling too early or late. |
| 23:18.04 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 23:21.04 | bmoraca | well, if you google an area code, it'll tell you what state it's in |
| 23:21.24 | kc8pxy | which shoudl be fairly close. |
| 23:22.08 | elitecoder | yeah well I wont be googling tens of thousands of numbers |
| 23:22.25 | kc8pxy | bmoraca: hyliafax/iaxmodem to funnel my faxes through my asterisk server, requires a fax modem, right? |
| 23:22.30 | bmoraca | who would you google tens of thowsands of numbers? |
| 23:22.45 | bmoraca | kc8pxy: not necessarily. are you trying to send or receive faxes? |
| 23:22.54 | kc8pxy | bmoraca: yes. |
| 23:22.59 | bmoraca | s/who/why/ |
| 23:24.02 | kc8pxy | elitecoder: why not just file them my timezone, and only manually check the border codes? |
| 23:24.16 | kc8pxy | meaning, the onles close to a zone line. |
| 23:24.24 | elitecoder | cause I'd need to know that to begin with :D |
| 23:24.26 | bmoraca | kc8pxy: hylafax requires some kind of modem. iaxmodem is a utility that gives you a tty that hylafax can use to talk to asterisk. so, the faxes will go out whatever your asterisk dialplan is configured to send calls out |
| 23:25.02 | bmoraca | elitecoder: you only need to find the timezones once. store them in a database and then compare the numbers to that database before you dial. |
| 23:25.20 | bmoraca | if you really were an elite coder, you'd have thought of that :P |
| 23:25.33 | kc8pxy | bmoraca: in contrast, with spandsp, and app_fax, i would plug my fax into an fxs port? |
| 23:26.16 | elitecoder | Ok let me explain what a real solution will look like: elitecoder, this data is stored by website/company X and is available here. If you import it into a database you can do a search against these parts of the number and come up with a fairly acurrate guess on the timezone. |
| 23:26.37 | bmoraca | kc8pxy: if you have a hardware fax machine, it doesn't matter. you could even use an ATA to do that. hylafax and app_fax are for sending and receiving faxes without the use of a hardware fax machine for applications such as fax-to-email. |
| 23:26.43 | elitecoder | or maybe a tutorial on getting the timezone from the areacode and exchange |
| 23:27.27 | bmoraca | elitecoder: the only data you'll be able to find is area code to locality. it'll be up to you to figure out what timezone that locality is in. |
| 23:28.46 | kc8pxy | bmoraca: how do i go about connecting my hardware fax to my asterisk server then? mebbe I've been barking up the wrong tree. |
| 23:28.51 | bmoraca | elitecoder: if it makes you feel any better, there really aren't that many area codes. it'd only take you a couple hours. |
| 23:29.05 | bmoraca | kc8pxy: use an ATA such as a linksys PAP2T |
| 23:29.08 | elitecoder | oki |
| 23:29.10 | bmoraca | kc8pxy: or an FXS port |
| 23:29.21 | elitecoder | i'll do it if I gotta. I'm sure there's a list already made somewhere |
| 23:29.37 | bmoraca | kc8pxy: neither requires spandsp or any other fax application on your asterisk server |
| 23:30.34 | bmoraca | kc8pxy: if you're using an FXS port, though, you might have timing issues if your trunk is on a different card, such as a PRI. faxing over the internet is generally not recommended either, though if you set your max speed to 9600 it'll almost always work |
| 23:32.19 | bmoraca | elitecoder: do a google search for "time zones by area code". just a thought, you know? |
| 23:32.42 | elitecoder | I did. I was ooking for something more advanced |
| 23:32.47 | elitecoder | but I'd settle for that |
| 23:33.20 | bmoraca | elitecoder: you were going to have to enter the data into your database anyway. |
| 23:34.23 | elitecoder | ok i'll be back later guys |
| 23:34.24 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 23:35.11 | bmoraca | kc8pxy: i generally tell my customers to keep a POTS line active for faxing/alarm/DSL. or, i set them up with fax-to-email and just use an ATA for outbound faxing from a hardware fax machine |
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| 23:43.31 | seb- | So when someone types the extension# in a dialplan context, the dialplan will automagically jump to those commands? |
| 23:48.56 | seb- | actually,,looks like initial commands have "extension" of "s" and then you add "s" commands to jump to other extension#'s right? |
| 23:52.54 | bmoraca | seb-: ... |
| 23:52.57 | bmoraca | ~book |
| 23:52.58 | 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 |
| 23:53.03 | bmoraca | read it |
| 23:55.03 | kc8pxy | bmoraca: why have a POTS line if they will being using voip for everything else? |
| 23:55.20 | bmoraca | kc8pxy: because VOIP isn't appropriate for everything else |
| 23:55.51 | *** join/#asterisk Apure (n=Guest355@c-68-45-150-58.hsd1.nj.comcast.net) |
| 23:55.58 | *** part/#asterisk Apure (n=Guest355@c-68-45-150-58.hsd1.nj.comcast.net) |
| 23:56.28 | bmoraca | kc8pxy: you can't get a DSL circuit without a POTS line. alarm and fire companies won't support you many times without a dedicated analog line. some credit card terminal companies require it as well. |
| 23:57.09 | bmoraca | kc8pxy: there's an important distinction between VOICE over Internet Protocol and EVERYTHING over Internet Protocol. some stuff just doesn't work over VoIP. |
| 23:57.13 | kc8pxy | ok.. makes sense now. |
| 23:58.07 | generalhan | anyone know if there is a way to change the file name used to save agent's calls? i know i can change the directory, but i want to change the actual file name forluma |
| 23:58.22 | generalhan | formula even ! |
| 23:59.36 | bmoraca | anyway, it's going home time |