| 00:00.18 | elitecoder | I'm just really confused as to which channels are used for what |
| 00:00.19 | DanyWalker | because when i run my script say: undefined variable amp_conf in .... |
| 00:00.55 | grEvenX | elitecoder: what other channels do you refer to? asterisk-dev is used for development talk, bugs/features etc. |
| 00:01.08 | elitecoder | sheeit |
| 00:01.25 | elitecoder | yeah that's the channel I'm referring to |
| 00:01.46 | elitecoder | I'll just ask my dam question and see what happens lol. |
| 00:01.58 | elitecoder | I'm writing an autodialer using the AGI and PHP. |
| 00:02.01 | grEvenX | elitecoder: much better :P |
| 00:02.12 | elitecoder | I'd like to place a call. Which action would be best to place a call, wait for silence, play a message and wait for input then add them to a queue? |
| 00:02.40 | grEvenX | DanyWalker: check reload.php on line 10... |
| 00:03.12 | DanyWalker | yes |
| 00:03.27 | DanyWalker | i cheked but i think that hot errors there |
| 00:03.43 | grEvenX | DanyWalker: paste the line... |
| 00:04.43 | elitecoder | DanyWalker: What're you making? |
| 00:04.43 | rob0 | Fried green tomatoes. |
| 00:04.52 | grEvenX | elitecoder: some good tips for referene: http://www.voip-info.org/wiki-Asterisk+manager+API |
| 00:04.59 | DanyWalker | i'm using a script to reload |
| 00:05.25 | elitecoder | I'm using the AGI. Different I think |
| 00:05.44 | elitecoder | thanks tho :D |
| 00:05.47 | grEvenX | elitecoder: that link is for the AGI.... |
| 00:05.59 | grEvenX | ah |
| 00:06.00 | grEvenX | sorry |
| 00:06.03 | grEvenX | too tired here |
| 00:06.07 | grEvenX | AGI, of course :P |
| 00:06.16 | elitecoder | haha thanks it does say stuff about actions |
| 00:06.19 | elitecoder | maybe that will help me |
| 00:06.24 | elitecoder | that's sorta what im looking for |
| 00:06.27 | elitecoder | like a funciton list |
| 00:06.35 | _brent_ | russellb: your ais module may be broken in rc4 |
| 00:06.47 | DanyWalker | elitecoder, i making this: http://pastebin.com/m18fb6f7d |
| 00:06.51 | _brent_ | russellb: or something else may be broken and affecting your module |
| 00:07.15 | _brent_ | russellb: the module works great in rc2, but not rc4 |
| 00:07.25 | grEvenX | DanyWalker: where the heck do you get your config variables from ? $amp_conf["AMPMGRUSER" |
| 00:07.26 | _brent_ | i haven't tried it in rc3 yet |
| 00:07.40 | grEvenX | DanyWalker: where's $amp_conf array defined? |
| 00:08.04 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 00:08.14 | DanyWalker | $amp_conf i cant found :( |
| 00:08.38 | grEvenX | DanyWalker: well, it's kinda hard connecting to the manager with a username/password of you don't supply any! |
| 00:09.10 | Poincare | is there a way to clear the conntrack table of to send a 'reset all connections' to a router? |
| 00:09.16 | Poincare | of = or |
| 00:09.16 | grEvenX | elitecoder: http://www.voip-info.org/wiki-Asterisk+AGI (Dialing Out) |
| 00:09.49 | DanyWalker | yes, but i set user an passworg in phpagi.conf |
| 00:11.32 | grEvenX | you're doing something strange... |
| 00:12.02 | grEvenX | you can't just throw in a bunch of code and just expect it to work :P |
| 00:12.46 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 00:12.47 | grEvenX | by using this line "$astman->connect("localhost", $amp_conf["AMPMGRUSER"] , $amp_conf["AMPMGRPASS"])" you have required that you allready have defined $amp_conf |
| 00:16.02 | DanyWalker | :< |
| 00:20.33 | *** join/#asterisk Grnd-Wire (i=user@75.147.178.170) |
| 00:20.52 | Grnd-Wire | Good afternoon all.. Can someone tell me what good the "Wait for Leader" function performs on a MeetMe conference? My users can still talk to each other! Is this normal? |
| 00:24.11 | elitecoder | thanks |
| 00:25.20 | *** join/#asterisk imm (n=ian@nv-76-3-138-164.dhcp.embarqhsd.net) |
| 00:26.07 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-247.rochester.res.rr.com) |
| 00:30.01 | elitecoder | On this page: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate where it says "Action: Originate "... near the top |
| 00:30.09 | elitecoder | is that an actual string of the command to send or? |
| 00:31.01 | Grnd-Wire | It looks just like a call file.. I've never used the mgr interface - but when you build a call file, that is one of the actual lines. |
| 00:31.32 | elitecoder | hmmm ok |
| 00:32.31 | Grnd-Wire | Can someone tell me what good the "Wait for Leader" function performs on a MeetMe conference? My users can still talk to each other! Is this normal? |
| 00:34.33 | octagon | should i be able to use SIP to call a ring group directly? like 6400@sip.myhost.com? i have a ring group defined on exten 6400, but the logs say: [Apr 17 20:33:07] NOTICE[14838] chan_sip.c: Call from '' to extension '6400' rejected because extension not found. |
| 00:37.20 | elitecoder | Anyone know how I'd go about making a call without connecting to a local phone first? |
| 00:37.37 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:37.47 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 00:44.15 | mmlj4 | elitecoder: console dial |
| 00:55.26 | *** join/#asterisk theone1976 (n=theone@200.30.187.67) |
| 00:57.22 | Grnd-Wire | Can someone tell me how the "Wait for Leader" option is supposed to work on a MeetMe conference? My users can still talk to each other, which doesn't seem normal. |
| 00:57.35 | theone1976 | Does anyone if it |
| 00:58.06 | theone1976 | Does anyone if it's possible to change the filename under /var/spool/asterisk/monitor? |
| 00:59.53 | theone1976 | I need to add the dialled number within the file's name. Currently it shows extension number, date, time, uniqueID. |
| 01:06.01 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 01:06.17 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 01:10.54 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 01:12.51 | ruben23 | <PROTECTED> |
| 01:13.00 | ruben23 | are codec used by local SIP client different for codec used by asterisk server to VIOP... |
| 01:13.12 | ruben23 | sample ulaw/alaw from softphones to asterisk then g729 tform asterisk server to voip.. |
| 01:21.21 | pfn | it can be the same or different |
| 01:21.29 | pfn | if they're different, asterisk will need to transcode |
| 01:24.26 | ruben23 | pfn: coz i want to setup a linux desktop for my SIP client..problem si no softphones for linux with g729 codec...but my asterisk server is using g729 codec.. |
| 01:25.05 | ruben23 | how do i setup it...to change the codec for my SIP softphones to my asterisk server... |
| 01:26.26 | ruben23 | coz when i used different codec aside form g729 now like alaw/ulaw the softphone will display call fialed not acceptable here". |
| 01:27.41 | ruben23 | the CLI output for my asterisk is no compatible codecs |
| 01:28.13 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 01:29.33 | pfn | canreinvite=no |
| 01:29.44 | ruben23 | http://pastebin.com/m4c6d3049 |
| 01:29.47 | pfn | if asterisk can't talk g729, then it can't transcode |
| 01:29.52 | ruben23 | heres my translation |
| 01:30.23 | pfn | you can't reinvite if you need asterisk to transcode |
| 01:30.41 | ruben23 | pfn:where do i set that..? |
| 01:30.46 | ruben23 | on the sip.conf |
| 01:30.54 | pfn | on the peer/user entry in sip.conf |
| 01:31.23 | ruben23 | so canreinvite=no....? |
| 01:31.29 | ruben23 | thats the config..? |
| 01:31.38 | pfn | that's about right |
| 01:32.38 | *** join/#asterisk Nasra (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 01:33.25 | ruben23 | pfn: thats the only setting and it will transcode..? |
| 01:33.42 | ruben23 | thus transcode affects performance of my call...? |
| 01:33.53 | pfn | transcoding will always affect performance |
| 01:33.59 | pfn | but if you have enough cpu power, it won't affect your call |
| 01:35.37 | ruben23 | pfn: so if my sip client is g729 to g729 to server...there is no transcoding...? |
| 01:36.40 | ruben23 | is it canreinvite=no or canreinvite=yes ...? |
| 01:45.35 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:45.55 | ruben23 | <PROTECTED> |
| 01:49.27 | styelz | does that help maybe your disallow/allow need tweaking http://www.voipuser.org/forum_topic_358.html |
| 01:53.44 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 01:54.15 | f0ner00t | Hello ... I set up nat but I got Sip / 2.0 404 Not found. |
| 01:55.45 | ruben23 | styelz:if im behind nat...do i have to input nat=yes peer/user on sip.conf...? |
| 01:56.09 | ruben23 | pfn: my codec transcoding now is working..thanks |
| 02:04.09 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 02:04.58 | *** join/#asterisk kib (i=kib@static-98-108-58-92.chi01.dsl-w.verizon.net) |
| 02:07.02 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:08.04 | kib | I have been having trouble getting a call oto work on a didforsale test did |
| 02:08.54 | kib | I keep getting the following error anytime I make a call [Apr 18 01:58:55] NOTICE[15778]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"Jason" <sip:301@xxx.xxx.xxx.xxx>;tag=as1a871d2a' |
| 02:09.23 | kib | I have tried all sorts of invite= and reinvite= combinations of yes and no in all sorts of places |
| 02:09.27 | kib | but nothing seems to work |
| 02:10.20 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 02:11.15 | kib | http://98.108.58.92/sip.conf this is the block I am using to connect. . ther eis more to the sip.conf file but I only put the parts related to the trunk I am trying to use |
| 02:11.38 | kib | what am I doing wrong here? |
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| 02:45.26 | Slade- | hey I'm looking at upgrading an old phone system that has 3 incoming lines to asterisk.. whats a good yet cheap card i can use with asterisk |
| 02:46.06 | *** join/#asterisk martyn-dev (n=andres_m@190.26.11.214) |
| 02:46.18 | martyn-dev | Hi |
| 02:46.44 | juanIMP | hi martyn-dev |
| 02:46.53 | martyn-dev | hi juanIMP how are you ¿ |
| 02:47.16 | juanIMP | pretty nice martyn-dev and you? |
| 02:47.23 | martyn-dev | good, thank u :) |
| 02:47.47 | *** join/#asterisk LeddyHM (n=NONE@75.63.105.141) |
| 02:49.07 | martyn-dev | I try register a softphone to asterisk extern, i am behind a proxy, but i dont know why not register with my asterisk extern :( |
| 02:50.47 | martyn-dev | I have configured in my proxy address my x.x.x.x addres to my proxy .. have i to set the port 3128 in some place?? i dont know where. |
| 02:54.18 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 02:54.52 | ruben23 | what is better codec..gsm or the ulaw/alaw codec....? |
| 02:55.42 | martyn-dev | ruben23: i prefer gsm |
| 02:55.44 | martyn-dev | :D |
| 02:56.13 | ruben23 | martyn-dev:why..? |
| 02:56.44 | ruben23 | martyn-dev: your using gsm now.. for your SIP client..? |
| 02:57.34 | ruben23 | martyn-dev: can i use ulaw only and alaw only...not both of the two codec..? |
| 02:57.57 | Slade- | theres another question. whats a good softphone? :) |
| 02:58.06 | martyn-dev | not both, really i dont know ruben23 |
| 02:58.15 | martyn-dev | Slade-: What OS ? |
| 02:58.25 | martyn-dev | ruben23: do you know this page http://www.asteriskguru.com/tools/bandwidth_calculator.php ? |
| 02:58.28 | Slade- | martyn-dev: windows. and nix actually |
| 02:58.53 | martyn-dev | gsm use 13 K |
| 02:58.56 | Slade- | doesnt need to be the same client :) |
| 02:59.17 | martyn-dev | Slade-: I think that Xlite and Zoiper is good :) |
| 02:59.29 | martyn-dev | Zoiper with IAX and gsm codec is great |
| 02:59.49 | *** join/#asterisk kumarphilly (n=kumar@66.160.197.30) |
| 03:00.10 | kumarphilly | hello |
| 03:00.20 | kumarphilly | efnet asterisk channel is fail |
| 03:00.24 | kumarphilly | anyone wanna help here lol |
| 03:11.05 | Juggie | ask, dont ask to ask |
| 03:11.49 | kumarphilly | ite |
| 03:12.42 | kumarphilly | well... i am planning to set up asterisk for a small business.. its prety simple 1 call in number and then you dial the extension and then 6 people dialing out... i want it so that everyone call call out at the same time if needed... |
| 03:12.54 | kumarphilly | the only issue is i dont know what service provider to go with |
| 03:13.17 | kumarphilly | even though this is for a business i would say currently less then 5k minutes forsure |
| 03:19.37 | Nasra | any1 know of good service provider that offer DID'S to the caribbean? |
| 03:20.43 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:21.10 | *** join/#asterisk astassistant (n=support@h-72-244-204-143.sttnwaho.dynamic.covad.net) |
| 03:21.59 | Slade- | wow you can use asterisk with just a regular modem? |
| 03:24.01 | kc8pxy | Slade-: i'm still what you'de call a noob to asterisk, but I'm not sure i'd trust even a PCI hardware modem as a fxs card. |
| 03:30.47 | Slade- | kc8pxy: looking at the x100p things they seem cheap and useful :) |
| 03:32.57 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:34.07 | rob0 | s/and useful// |
| 03:34.31 | kc8pxy | rob0: hehe.. |
| 03:35.20 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 03:35.37 | kc8pxy | Slade-: they are only useful for testing. because of quality issues that are known to be present, they should not be used in a "this is my REAL asterisk server!:)" situation. |
| 03:36.04 | coppice | there is nothing wrong with using an x100p card, if you use it with OSLEC |
| 03:36.40 | Slade- | ack. it creates lots of line echo? |
| 03:36.49 | kc8pxy | coppice: ok, mebbe I'm showing my ignorance. what's OSLEC ? |
| 03:37.02 | coppice | a decent free echo canceller |
| 03:37.43 | Slade- | so overall sound quality is bad with x100p cards? |
| 03:39.03 | Slade- | or is needing something like oslec common? |
| 03:40.08 | coppice | using a decent echo canceller is a normal requirement |
| 03:40.15 | Slade- | ok |
| 03:40.33 | Slade- | even when you move up to the t1 level. etc? or just on the small cards |
| 03:41.05 | coppice | you pretty much always need an echo canceller on the PSTN, to remove echo from the phone at the far end. |
| 03:41.07 | kc8pxy | Slade-: every text and tutorial i've seen says they they are good cards for testing, to get yoru fewt wet with dahdi/zap, but they are specifically NOT recommended for normal use. i'm assuming that's because thee are other choices that don't need that noise canceling as much. |
| 04:01.37 | *** join/#asterisk ukine (n=ukine@6532200hfc143.tampabay.res.rr.com) |
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| 04:18.40 | kc8pxy | dahdi +kvm/xen is a bad mix??? or is it fine? |
| 04:19.53 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 04:20.40 | theone1976 | I'd like to have the dialed phone number included in the filename recordings (/var/spool/asterisk/monitor). Is this possible? |
| 04:21.19 | [TK]D-Fender | theone1976: You set the name, so go do it |
| 04:21.51 | *** part/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 04:22.17 | [TK]D-Fender | kc8pxy: Yes virtualizing a sensitive timer is a bad idea |
| 04:22.45 | theone1976 | D-Fender: ? |
| 04:24.36 | kc8pxy | [TK]D-Fender: kk.. but if i have a small farm of asterisk servers, only the ones with dahdi cards NEED to be non-virtual? |
| 04:25.48 | [TK]D-Fender | kc8pxy: No, the ones needing DAHDI at all. |
| 04:25.58 | theone1976 | Current files show up in the following manner > OUTxxx-20090417-210512-2142223832.2862.gsm |
| 04:26.29 | theone1976 | extension+date+time+uniqueid |
| 04:27.20 | kc8pxy | [TK]D-Fender: so any server that routes calls that include fxo/fxs/pri, all of those servers need to be non-virtual.. yes? |
| 04:27.31 | theone1976 | Is there a variable that I should use in the extensions.conf file so that the dialed number is included? |
| 04:28.18 | [TK]D-Fender | kc8pxy: You know this is pretty darn simple. any system needing DAHDI should be non-VM |
| 04:28.59 | kc8pxy | [TK]D-Fender: ok. so i should keep it as far from my sterisk farm as i can. |
| 04:48.40 | keith4_ | how do I Goto an extension that's defined like this? exten => _1NXXNXXXXXX,1,NoOp |
| 04:48.55 | keith4_ | do I have to give it a label? |
| 04:59.33 | *** join/#asterisk Pan3D (n=Pan3D@63.208.160.186) |
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| 05:21.43 | Grnd-Wire | Can someone tell me how the "Wait for Leader" option is supposed to work on a MeetMe conference? My users can still talk to each other, which doesn't seem normal. |
| 05:29.06 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
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| 05:37.51 | *** join/#asterisk jbjapan (n=jbjapan@221.242.85.34) |
| 05:37.58 | jbjapan | hello! |
| 05:39.46 | octagon | hi! |
| 05:40.38 | kc8pxy | jbjapan: heya! |
| 05:43.34 | jbjapan | kc8pxy : Are you familiar with call forwarding on asterisk? |
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| 05:56.23 | kc8pxy | jbjapan: I'm an asterisk noob still.. no. |
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| 07:09.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 07:32.05 | cool^tom | Hi |
| 07:33.17 | cool^tom | Is there a major problem configuring a PRI Card? |
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| 08:04.45 | jerlique | ping |
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| 10:43.00 | kib | I keep getting the following error anytime I make a call [Apr 18 01:58:55] NOTICE[15778]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"Jason" <sip:301@xxx.xxx.xxx.xxx>;tag=as1a871d2a' |
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| 12:50.35 | eppigy | MONRING |
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| 12:53.37 | MaliutaLap | eppigy: almost ... just 1 hr 7 minutes to go |
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| 13:53.11 | gr0mit | any snom experts here? |
| 13:53.54 | h-idrisi | gr0mit, what is the problem may be i can help |
| 13:53.58 | mmlj4 | expert enough to know... never mind |
| 13:54.52 | gr0mit | am using a 360, with f/w 7.3.14 |
| 13:55.35 | gr0mit | came back after about 4 days away to see http://wendolene.txrx.org.uk/secretstuff/snombug.jpg |
| 13:55.58 | gr0mit | (excuse the dust on the phone!) |
| 13:57.16 | gr0mit | anyone else seemn this display shift? |
| 13:58.34 | gr0mit | phone works fine apart form the display being garbled |
| 14:00.27 | gr0mit | a reboot cures the probl |
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| 14:23.16 | kib | anyone here every worked with didforsale? |
| 14:23.26 | kib | been getting a problem anytime I Try to route a call that way |
| 14:23.49 | kib | I put int he configs exactly as they have it on their site but I keep getting invite errors |
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| 14:47.08 | knarfly | can some tell me what this means, I'm trying to test some new extensions.. |
| 14:47.08 | knarfly | exten => s,n(lbl_dial-in_0),Set(TIMEOUT(response)=10) |
| 14:47.46 | knarfly | particularly what's with the (lbl_dial-in_0)...this is in a context called dial-in |
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| 14:54.10 | jgoo | fucking +i in this room. anyway. I have a PAP2T and it won't make calls between the two extensions, each can make calls, and receive calls, and via the FOP I can connect the two phones and talk, but dialing the one extension from the other doesn't work, has anyone else seen this behaviour? |
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| 15:33.26 | kib | jgoo: why +i? |
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| 15:52.27 | kumarphilly | hola |
| 15:52.33 | kumarphilly | so whats a good service provder for a small business |
| 15:52.36 | kumarphilly | multiple call out |
| 15:52.40 | kumarphilly | and one DID |
| 15:52.47 | *** join/#asterisk ingenius (n=alektro@host43.190-138-51.telecom.net.ar) |
| 15:54.41 | kumarphilly | wahts counts as business with callcentric |
| 15:58.10 | gr0mit | kumarphilly, which country? |
| 15:58.33 | kumarphilly | US |
| 15:59.08 | gr0mit | ahok, not able to advise, sorry |
| 15:59.33 | kumarphilly | oh darn=/ |
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| 16:05.21 | Grnd-Wire | good morning everyone.. Can someone speak me to me about the quality of the MeetMe conference function?? I am noticing that the "wait for leader" option does not work. Is it supposed to? |
| 16:07.48 | Steve_J-obs | what do you mean the quality of the meetme function? |
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| 16:09.53 | Grnd-Wire | Steve_J-obs: As I said, I am noticing the "wait for leader" (w) option does not work. Is it supposed to? |
| 16:16.26 | kumarphilly | Steve_J-obs, you u know of SIP service providers in the US |
| 16:18.55 | Grnd-Wire | kumarphilly: I use voipstreet - and like them a lot. |
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| 16:25.10 | kumarphilly | 1.9 cents flat per out |
| 16:25.19 | kumarphilly | minute* |
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| 16:36.54 | kumarphilly | Monthly quota of 500 simultaneous ring talk minutes. Additional minutes are $.04/minute |
| 16:36.56 | kumarphilly | what does that mean |
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| 17:14.10 | Grnd-Wire | kumarphilly: What are you reading? |
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| 17:41.21 | kib | does anyone one know what causes this error with didforsale services: [Apr 18 01:58:55] NOTICE[15778]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"Jason" <sip:301@xxx[Apr 18 01:58:55] NOTICE[15778]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"Jason" <sip:301@xxx.xxx.xxx.xxx>;tag=as1a871d2a.xxx.xxx.xxx>;tag=as1a871d2a |
| 17:41.45 | kib | not having any luck finding out what I did wrong . I put int he configs exactly as they listed them on the didforsale web site |
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| 18:23.24 | Slade- | hey whats a good/cheap brand of ip phone? |
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| 18:26.04 | rhombus | Pattern match question: _[*#0-9]! <--- does that match * or # then a digit, or does it match *, then #, then a digit? |
| 18:27.51 | kib | I thinkt he better way to find 0-9 is X |
| 18:28.24 | kib | if I recall N = 0-9, N = 2-9, and I forget the 3rd one which means 1-9 |
| 18:28.30 | rob0 | What's the point of that? "Match any button that can be pushed on a phone?" |
| 18:28.43 | kib | but puttingin a bracket like that sould make it match anything within the bracket |
| 18:28.50 | rhombus | kib: Sure, that's fine... I took it from an example. I'm more interested in the question of how * and # interact in that example. |
| 18:28.50 | rob0 | X=0-9 right? |
| 18:28.56 | kib | but I assumed before now that you sould separate th elist with , and ranges with - |
| 18:29.15 | rhombus | so, my hunch so far is this: |
| 18:29.22 | kib | if youw ant to match those in order I would think _*#[0-9] |
| 18:29.24 | kib | but I am no expert |
| 18:29.55 | kib | you might need to add a . at the end if you wantt o include strings longer then that |
| 18:29.55 | rhombus | _[*#0-9]! will match *7, #7, *712, #712 |
| 18:30.22 | kib | I a mpretty sure that will only match * or # or any single digit |
| 18:30.30 | kib | I owould have to try it to see for sure though |
| 18:30.33 | rhombus | i'll have to try it :) |
| 18:31.04 | kib | to match something beginnin with * or # maybe this will work |
| 18:31.18 | kib | _[#,*]. |
| 18:31.30 | kib | that means it will begin with either # or * and have 1 or more digits following |
| 18:31.45 | kib | the following digits might be another * or # in that case |
| 18:32.36 | kib | or maybe |
| 18:33.10 | kib | _[*,#]X. will match either #7 or *7 or anything with more digits |
| 18:33.32 | kib | but a t least it requires a number follow the * or # |
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| 18:34.46 | degrade | Has anyone using asterisk on solaris 10? Or tried it? |
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| 18:35.57 | elitecoder | When placing a call with the gateway interface, how do I go about setting the caller id? I'm really new to it |
| 18:36.21 | elitecoder | I don't even fully understand how to place a call seeing a sample line would really help :x |
| 18:36.24 | elitecoder | lol |
| 18:36.34 | rhombus | kib: okay, the answer is that it matches either or both :) |
| 18:38.04 | rhombus | _[*#X]! matches *7, #7, *# -- as a pattern it's kind of useless, but at least it establishes the behavior :) |
| 18:40.28 | kib | by that deinfation it will also match 77 |
| 18:42.47 | kib | elitecoder: there is a callerid field in about any context |
| 18:43.14 | kib | callerid=Test <nxxnxxxxxx> |
| 18:43.33 | kib | I believe the part between the bracket is the phone number and what comes before it is the text name |
| 18:43.46 | elitecoder | ok |
| 18:43.55 | kib | although depending on the direction of the call your sip provider may or may not replace the callerid info |
| 18:43.58 | elitecoder | so that's how you set it? |
| 18:44.10 | elitecoder | you just send over a key/value in key=value format sorta? |
| 18:44.21 | kib | but if your sending it down to another oe of your devices or servers then it will likely be passed through unharmed |
| 18:44.29 | elitecoder | alright |
| 18:44.36 | elitecoder | we're gunna use bandwidth.com |
| 18:44.38 | kib | I set it in the sip.conf for each phone |
| 18:44.48 | kib | then any calls that come from that sip entry should get the callerid set |
| 18:45.01 | kib | if there is no setting it will sue what the caller provides if any or set it to unknown |
| 18:45.06 | elitecoder | oh is there a way to do that for everything? then I don't have to do it in my script |
| 18:45.11 | elitecoder | ok |
| 18:45.37 | kib | I thought about bandwidth.com but I was not sure if they were worth the cost and minimums they required. we are starting small so I began with did3sale and gafachi |
| 18:45.55 | kib | you want the same callerid for every phone? |
| 18:46.02 | kib | you could set it int he block you defined for your upstream |
| 18:46.13 | elitecoder | in sip.conf? |
| 18:46.15 | kib | then any calls that passes over that trunk will get it set |
| 18:46.20 | kib | althoug I never tried that before |
| 18:46.27 | elitecoder | ok well it's not a huge issue :] |
| 18:46.32 | elitecoder | yet |
| 18:46.34 | kib | either sip.conf or iax.conf or whatever method you handle your calls |
| 18:46.43 | kib | you could also use variables |
| 18:46.46 | elitecoder | yeah we're using their sip trunks |
| 18:46.56 | kib | althoug I have never tried this they appear like ${VARIABLE} |
| 18:47.16 | elitecoder | I'll see what I can do, once i get to that. |
| 18:47.28 | kib | there I a nice pdf document on t he main web site and in the sip chapter itr speaks of using the callerid field for the phone devices |
| 18:47.35 | elitecoder | ok |
| 18:47.40 | elitecoder | brb |
| 18:47.42 | kib | also you could include settings that are common |
| 18:48.03 | kib | create a [common] block and then include it in each other block you like so you can save the repetitive typing |
| 18:48.32 | kib | several ways to do the same thing I guess |
| 18:48.46 | kib | but the callerid field is the answer you need. but its up to bandwidth.com if they accept your callerid or replace it |
| 18:49.31 | kib | I wish there was more helpers around today. been trying to get an error messag elooked by someone today. I guess I should go seeking a paid consultant |
| 18:51.59 | kib | [Apr 18 18:51:28] NOTICE[15778]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"Jason" <sip:301@xxx.xxx.xxx.xxx>;tag=as28085d87' |
| 18:55.57 | Slade- | hey whats a good/cheap brand of ip phone? |
| 18:58.19 | kib | Slade-: softphones are very cheap |
| 18:58.38 | kib | personally I use a normal phone with a voip box from say linksys |
| 18:58.42 | kib | or something like this |
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| 19:06.36 | elitecoder | or you can get a headset :D |
| 19:06.41 | elitecoder | and something like xlite |
| 19:07.33 | Steve_J-obs | hi guys , can anyone here help find out I have one way audio in my new h323 setup please? |
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| 19:11.25 | kib | Steve_J-obs: one major reason for one way audio I find is NAT...if your system has a 192.186.xx.xx address then you have nat and should configure your device accordingly |
| 19:28.26 | Subdolus | I would've thought Steve Jobs would know that |
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| 19:31.45 | Slade- | kib: yea was looking for a hardphone.. |
| 19:32.28 | Subdolus | ATA |
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| 19:36.59 | _sasa_ | hello |
| 19:37.18 | _sasa_ | anybody knows where can I get the libnewt sources for download ? |
| 19:45.57 | kib | where I can I hire someone to fix my problem I am seeing right now |
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| 19:46.26 | kib | Slade-: I prefer just to get a voip netbox which accepts one or more normal phones |
| 19:46.38 | kib | the oness I buy are like small dsl modems |
| 19:47.09 | kib | they run about $20 or so in most places and you can use any normal phone you like. I find this a bit cheaper then buying a voiip capable phone direclty |
| 19:47.37 | _brent_ | kib: for a business or household? |
| 19:49.26 | _brent_ | kib: if it's for a business, you could look for used phones. you can find used linksys spa 942's for like 40 bucks |
| 19:49.59 | _brent_ | ATA (and their equivalents) are great for home users, but no business deserves that |
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| 19:58.26 | kib | _brent_: small office |
| 19:58.42 | Slade- | interesting :) |
| 19:58.47 | kib | and _brent_ I was not asking about ophones I was asking abut an error message |
| 19:59.09 | kib | Slade- was asking about phones |
| 19:59.16 | _brent_ | ok, gotcha |
| 19:59.20 | kib | I jusmentioned what I use here |
| 19:59.36 | _brent_ | i PM'ed you the name of a consultant that can help troubleshoot |
| 19:59.39 | kib | a small linksys deice I get for like $20 or so and any normal phone is most cost effecitve for our needs |
| 20:01.16 | rob0 | $20? The cheapest I've seen is $30, and those were locked to Vonage. |
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| 20:01.52 | rob0 | But yeah, I got my SPA 2000 ata off of ebay for $35ish. |
| 20:05.19 | elitecoder | YAWN |
| 20:13.06 | _sasa_ | heh, is digium site down ? |
| 20:13.22 | _sasa_ | or is it just me ? |
| 20:13.53 | _brent_ | http://downforeveryoneorjustme.com/ |
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| 20:17.13 | _sasa_ | http://downforeveryoneorjustme.com/download.digium.com |
| 20:17.24 | _sasa_ | thx _brent_ |
| 20:21.33 | kib | rob0: we get them for about that from our old provider |
| 20:21.53 | kib | although they went out ofb usiness so I am not 100% sure where to get them for now but if they managed to get them for that price I will probably find them sooner or later |
| 20:22.03 | kib | I just looked online and the same model we have at voipsupply is $50 |
| 20:22.17 | kib | we got the oens we have cheap since we bought around 15 at once I think |
| 20:22.25 | kib | still it is cheaper then buying a whole phone |
| 20:22.28 | kib | even at the $50 price |
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| 20:57.45 | *** join/#asterisk freddyk (n=freddyk@host33-39-dynamic.50-79-r.retail.telecomitalia.it) |
| 21:03.55 | kumarphilly | grantm, are you grandwire |
| 21:04.04 | kumarphilly | grnd wire |
| 21:04.04 | hardwire | is hardwire |
| 21:05.13 | hardwire | anybody in albania register a domain for me? |
| 21:13.26 | *** join/#asterisk bl4 (n=bl4qkuba@dsl1-ore-80.fiber.net) |
| 21:18.23 | *** join/#asterisk shinao1 (n=shinao1@41.219.223.11) |
| 21:18.46 | _sasa_ | is zapatel.conf supposed to be in /etc/asterisk ? |
| 21:20.12 | justdave | there's a zaptel.conf and a zapata.conf |
| 21:20.38 | justdave | zaptel.conf is usually in /etc, zapata.conf is usually in /etc/asterisk |
| 21:21.07 | justdave | and if you're running a semi-recent version of Asterisk 1.4 or newer you should probably have neither (in favor of dahdi stuff) |
| 21:22.56 | _sasa_ | oh ok, so i must compile dahdi if i want to get 1.6 running |
| 21:23.04 | _sasa_ | thx justdave |
| 21:23.44 | _sasa_ | but for my tdm410 i also need zapatel correct ? |
| 21:27.25 | _sasa_ | and should I remove zapatel if I install dahdi ? |
| 21:32.32 | knarfly | can someone tell me why this extension is suddenly giving me trouble in Asterisk-1.6.0.9 |
| 21:32.32 | knarfly | http://pastebin.com/m51374c36 |
| 21:33.50 | knarfly | _sasa_: 1.6 only works with dahdi, yes you need to install dahdi...recommend linux-dadhi-complete |
| 21:37.15 | war9407 | without asteirsk-gui I can get outbound caling workng fine |
| 21:37.25 | war9407 | with asterisk-gui though I create my trunks etc and specify a dialing plan |
| 21:37.28 | war9407 | but not taking |
| 21:37.35 | war9407 | [Apr 18 17:36:57] NOTICE[13090]: chan_sip.c:17295 handle_request_invite: Call from 'line1' to extension 'XXXXXXx' rejected because extension not found. |
| 21:37.36 | war9407 | arg pita. |
| 21:40.53 | _sasa_ | knarfly: thx, so I can deinstall zaptel correct ? |
| 21:44.14 | rob0 | ~dahdi |
| 21:44.15 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 21:45.56 | *** part/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 21:46.07 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 21:46.24 | [TK]D-Fender | knarfly: Maybe you could actually SHOW us the problem |
| 21:46.36 | _sasa_ | why on 64bit dahdi installs libs in /usr/lib instead of /usr/lib64 |
| 21:46.38 | _sasa_ | ? |
| 21:48.07 | rob0 | That's because /usr/lib is probably reserved for 32-bit libraries, for compatibility. |
| 21:50.43 | _sasa_ | so |
| 21:50.59 | _sasa_ | why then compilled on 64bit they go into /usr/lib ? |
| 21:54.36 | _sasa_ | it installs the libraries in /usr/lib and perl modules in /usr/lib64/perl/ which for perl is correct |
| 21:55.11 | _sasa_ | but i think that the libraries should also go into /usr/lib64 and not /usr/lib |
| 22:06.12 | knarfly | [TK]D-Fender: sorry dude, got disconnected there |
| 22:07.02 | knarfly | my dialplan has several contexts which work fine but when I load asterisk the log file says that it was unable to include context music in context deluxe |
| 22:07.14 | knarfly | stil the extensions work |
| 22:13.42 | *** join/#asterisk shinao1 (n=shinao1@41.219.247.166) |
| 22:14.33 | *** join/#asterisk Steve_J-obs (i=Steve_J-@pool-71-190-74-157.nycmny.east.verizon.net) |
| 22:14.46 | Steve_J-obs | hello everybody! |
| 22:15.55 | [TK]D-Fender | I still don't see the error |
| 22:21.00 | war9407 | ?? voicemail |
| 22:22.18 | Fabian- | touch /var/spool/asterisk/voicemail/context/user/INBOX/msg0000.txt |
| 22:24.50 | war9407 | when someone calls me, how do I make it PROMPT, for user A, type 6000 for user B, type 6001 |
| 22:24.54 | war9407 | currently it only goes to voicemail for 6000 |
| 22:25.04 | war9407 | exten => 123,n,Voicemail(6000) |
| 22:25.25 | [TK]D-Fender | war9407: Because you are telling it explicitly what mailbox to go to there. |
| 22:26.07 | war9407 | I have been googling for awhile and there is/are seems to be no real simple examples which is a pITA, I gave up using the asteirk-gui |
| 22:26.17 | war9407 | that thing is a PITA to use, /me prefers vim + edit conf files |
| 22:26.31 | war9407 | [TK]D-Fender: ok |
| 22:26.48 | [TK]D-Fender | war9407: "core show application voicemail" |
| 22:27.00 | war9407 | http://www.latenightpc.com/blog/archives/2006/11/19/taking-the-call-my-extensionsconf-for-asterisk-12-and-how-it-works |
| 22:27.03 | war9407 | k, will also try that |
| 22:30.36 | Steve_J-obs | can anybody please give me a hand with a codec problem using h323? |
| 22:34.03 | *** join/#asterisk macros73_ (n=cs@c-71-61-74-104.hsd1.pa.comcast.net) |
| 22:34.40 | *** part/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 22:34.47 | *** join/#asterisk chiwawa_42 (n=jerome@can59-3-82-233-175-214.fbx.proxad.net) |
| 22:35.13 | _sasa_ | how can I know which module I need to load for my tdm410 ? |
| 22:35.16 | war9407 | [TK]D-Fender: for regular voicemail |
| 22:35.27 | war9407 | hm nvm, |
| 22:35.31 | war9407 | I will just keep gogling and testing |
| 22:36.21 | justdave | _sasa_: sorry, got distracted earlier... if you have zaptel on there already and you're putting in dahdi, then you need to actually remove all the zaptel stuff before you compile asterisk or it'll get confused |
| 22:36.26 | [TK]D-Fender | _sasa_: tdm24xxp |
| 22:36.53 | chiwawa_42 | Hi ! I'm preparing my first VoIP setup, and i'm currently reading the O'Reilly book on Asterisk, but still have some pending questions. I've been offered some cheap Openvox cards and couldn't find comprehensive feedback about them. Their site claims full compatibility but I'm guessing there' a catch somewhere. Could anyone explain me what could it be ? |
| 22:38.10 | [TK]D-Fender | chiwawa_42: Shitty support from the maker and based on lod Digium designs |
| 22:38.13 | [TK]D-Fender | old* |
| 22:38.21 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 22:39.19 | chiwawa_42 | [TK]D-Fender: so they basically works, but should they fail, i'd probably get stuck with no support ? |
| 22:39.25 | war9407 | [TK]D-Fender: link to good example script to dial multiple extensions and leave a voice message for a particular one? |
| 22:39.30 | war9407 | [TK]D-Fender: for 1.6.x? |
| 22:40.27 | [TK]D-Fender | chiwawa_42: Might work OK, might cause trouble in systems that other makers wouldn't, and you may find yourself without much help |
| 22:41.05 | carrar | war, how about 1.4? |
| 22:41.21 | chiwawa_42 | so it's all about luck or is there any known issues, or particular case known not to work well with these cards ? |
| 22:41.25 | [TK]D-Fender | war9407: Basic IVR. "core show function timeout", read up on the "t" and "i" standard extensions, "core show application waitexten", etc |
| 22:41.45 | [TK]D-Fender | chiwawa_42: Nothing particular. Randomness is part of the problem |
| 22:42.05 | war9407 | Command 'core show function timeout' failed. |
| 22:42.10 | war9407 | No function by that name registered. |
| 22:42.19 | war9407 | the other one worked though |
| 22:42.38 | carrar | core show function TIMEOUT |
| 22:42.59 | war9407 | thx. |
| 22:43.05 | chiwawa_42 | ok... any known statistics ? About my planed setup, I'm using a VIA C5@1GHz (epia-pd) with 512Mb RAM to serve 4-8 FXS and 4 DECT/SIP phones. I'd use 2FXO lines and 2 outgoing SIP accounts (with LCR). The board would be an OpenVox A1200P |
| 22:43.20 | [TK]D-Fender | war9407: "core show function TIMEOUT" <- sorry, capitals count |
| 22:43.51 | carrar | I got yr back! :) |
| 22:43.54 | carrar | phear |
| 22:44.09 | [TK]D-Fender | chiwawa_42: Epia uses a crappy PCI bus, you may or may not run into issues |
| 22:44.22 | [TK]D-Fender | carrar: And get away from my back! |
| 22:44.28 | carrar | haha |
| 22:44.33 | [TK]D-Fender | posts an "exit only" sign over his ass |
| 22:44.36 | carrar | *wink* *wink* |
| 22:45.38 | chiwawa_42 | [TK]D-Fender: cheap analog card and crappy bus, CPU probably too slow for soft echo, I guess that's a _really_ bad idea, is it ? |
| 22:46.56 | [TK]D-Fender | chiwawa_42: "Your failure is complete" - Darth Sidious |
| 22:48.25 | chiwawa_42 | alright... Considering I have no extra budget, and the mainboard is already avaible to me, what should I do to avoid weird issues ? Would the use of some ATA be better than the analog card ? |
| 22:49.24 | carrar | AudioCodes makes nice 8 port FXS ATA's |
| 22:50.44 | carrar | MP118/FXS - 8 FXS Ports |
| 22:50.48 | carrar | http://www.audiocodes.com/products/mediapack-11x-uc |
| 22:51.01 | chiwawa_42 | got it, 400EUR ex. VAT |
| 22:51.20 | carrar | I've used the AudioCodes stuff before |
| 22:53.51 | _sasa_ | thx justdave |
| 22:54.06 | _sasa_ | thx TK-Fender |
| 22:56.24 | chiwawa_42 | Grandstream GXW-4008-FXS looks similar for hal the price of the audiocodes, may it work as well ? |
| 22:56.24 | tzafrir_laptop | [TK]D-Fender, crappy PCI bus? |
| 22:56.25 | _sasa_ | do i need libpri if I dont have ISDN ? |
| 22:56.33 | tzafrir_laptop | _sasa_, no |
| 22:56.45 | _sasa_ | thx |
| 22:56.55 | chiwawa_42 | tzafrir_laptop: Older VIA chipsets ar known for some lazy IRQ support |
| 22:56.57 | tzafrir_laptop | [TK]D-Fender, this is based on? |
| 22:57.23 | [TK]D-Fender | tzafrir_laptop: I've heard a few accounts in here over the years of those whove found them flakey for TDM use |
| 22:57.29 | tzafrir_laptop | this sounds like a nice rumour. I heard the same stories about "intel chipsets" before |
| 22:59.10 | tzafrir_laptop | speaking of OpenVox: theyir driver finally supports red alarms on FXO |
| 23:00.20 | *** join/#asterisk dzup (i=dzup@bnc1.shellium.org) |
| 23:00.53 | chiwawa_42 | tzafrir_laptop: i815 did that too, but this one was for P3, it's rather old... |
| 23:05.45 | chiwawa_42 | Grandstream GXE-5028 appears to do exactly what I want, not needing a server to work... Maybe I took it the wrong way :s |
| 23:07.49 | carrar | Grandstream is crap |
| 23:08.22 | carrar | </IMHO> |
| 23:10.25 | chiwawa_42 | carrar: what makes you say that ? bad sound quality ? hardware issues ? |
| 23:12.34 | [TK]D-Fender | ~gs |
| 23:12.35 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:12.41 | [TK]D-Fender | ~grandstream |
| 23:12.42 | infobot | grandstream is probably the Yugo of VoIP hardware. Run. Run away now. |
| 23:14.13 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:17.31 | chiwawa_42 | [TK]D-Fender: that's nice, but what is it based on ? |
| 23:17.59 | rob0 | experience, I would think? |
| 23:18.19 | [TK]D-Fender | chiwawa_42: tons of users in here with reports of poor audio & build quality |
| 23:18.25 | [TK]D-Fender | chiwawa_42: Cheap shit |
| 23:18.36 | chiwawa_42 | I mean, everyone seems to aknoweldge gs hardware is crap, but it's still widely used by cheap VoIP provider and seems to work. It is supporter for auto-provisioning in trixbox, the developpers wouldn't have bothered if they were such craps, would they ? |
| 23:18.38 | rob0 | Go ahead and buy them ... save money, have fun. |
| 23:18.52 | [TK]D-Fender | chiwawa_42: Then again you are aiming bargain basement and desparately hoping to get away with a little as you can |
| 23:19.09 | rob0 | Oh yes, invoke trixbox as an example of quality. :) |
| 23:19.17 | chiwawa_42 | i'm lloking for a reasonably cheap system for 6 FXS and 2FXO, preferably to run on an existing VIA Epia PD, that's all I got for now |
| 23:20.01 | rob0 | ATA's are cheaper than even Grandstream. |
| 23:20.04 | chiwawa_42 | [TK]D-Fender: I'm not looking for an Hi-fi sound wuality or tons or fancy features, even vocemail is optional here |
| 23:20.36 | rob0 | Voicemail is included with the purchase price of *. |
| 23:20.40 | [TK]D-Fender | chiwawa_42: You like echo on your calls? ok/fine/sure |
| 23:21.10 | [TK]D-Fender | chiwawa_42: You know what... just go find out for yourself and stop asking. |
| 23:21.42 | [TK]D-Fender | chiwawa_42: You don't want to hear bad news and thats what pretty much all the choices you are looking at now will get you around here. So just go do it |
| 23:21.46 | chiwawa_42 | [TK]D-Fender: well, depends on how much echo are we looking at, if there's a way to avoid it with less than $50/port, i'll go for it, if not, i'd look for the best compromise |
| 23:22.24 | chiwawa_42 | [TK]D-Fender: what would you recommend for a decent quality at a decent price ? |
| 23:22.42 | [TK]D-Fender | chiwawa_42: 2 FXO is the hard part to get at good quality |
| 23:23.01 | [TK]D-Fender | chiwawa_42: without coming at a premium. |
| 23:24.40 | chiwawa_42 | ok, got it |
| 23:25.04 | [TK]D-Fender | chiwawa_42: At a minimum I suppose I'd start with a TDM410P w/ 2 FXO modules, and the FXS would all be ATA's |
| 23:25.43 | chiwawa_42 | yeah, that's what I was thinking of, a board with hardware EC for FXO |
| 23:26.35 | [TK]D-Fender | chiwawa_42: the EC will add a fair bit to your cost. Do you predict your needs to expand in the near future |
| 23:26.37 | [TK]D-Fender | ? |
| 23:26.42 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:27.08 | chiwawa_42 | no, the only possible expansion is to add some SIP DECT phones |
| 23:27.15 | chiwawa_42 | no extra analog lines |
| 23:27.25 | kib | has anyon here eevery got didforsale to work on asterisk? |
| 23:27.46 | kib | if so what changes do I need to make to the configs they have on their web site to make it work? |
| 23:27.52 | [TK]D-Fender | chiwawa_42: You could start with the base TDM410P and see if SWEC cuts it, and add teh HWEC module if needed |
| 23:30.16 | hardwire | http://hardwire.pastey.net/112567 |
| 23:30.33 | *** join/#asterisk ingenius (n=alektro@host154.190-224-88.telecom.net.ar) |
| 23:30.43 | hardwire | any reason mixmonitor would cause asterisk crashyness every 30 minutes or so while dealing with 20+ calls from zap to zap |
| 23:30.53 | elitecoder | Does anyone have experience with dialing multiple numbers simultaneously? |
| 23:31.06 | hardwire | its recording to memory which isn't running dry at all.. and the script it runs basically just returns |
| 23:31.34 | hardwire | I'm not sure if the script is causing issues or not.. I can't easily test mixmonitor without a script to make sure memory isn't all used up |
| 23:32.25 | [TK]D-Fender | elitecoder: "core show application dial" <- its all there |
| 23:32.41 | chiwawa_42 | [TK]D-Fender: If I undrstood you right, for the FXS ports, I'd use any ATA BUT grandstream, that's it ? |
| 23:32.58 | war9407 | exten=>*60,1,VoiceMailMain(6000) vs exten=>*60,1,VoiceMailMain(6001) |
| 23:33.01 | [TK]D-Fender | chiwawa_42: Linksys PAP2T / 2102, etc |
| 23:33.08 | war9407 | is this possible to have 2 different voice mail boxes if I am not using sip phones |
| 23:33.13 | chiwawa_42 | ok :) |
| 23:33.20 | war9407 | they both seem to use 6000, or 6001 has problems, 6000 works though |
| 23:33.26 | elitecoder | thanks [TK]D-Fender |
| 23:33.34 | [TK]D-Fender | war9407: Voicemail boxes have no relation to SIP phones |
| 23:33.38 | chiwawa_42 | yeah, thanks a lot [TK]D-Fender |
| 23:33.41 | _sasa_ | is there somebody who has a package of the dahdi for slamd64 linux ? |
| 23:34.02 | [TK]D-Fender | war9407: And you just showed me the SAME EXTENSION twice. the second one would overwrite the first |
| 23:34.09 | _sasa_ | i'm getting kernel panicks each time i boot |
| 23:34.10 | war9407 | typo ;) |
| 23:34.47 | [TK]D-Fender | war9407: pastebin actual relevent dialplan and CLI output |
| 23:35.52 | war9407 | http://pastebin.com/m2b6b68ae |
| 23:37.02 | war9407 | well I got one thing working today |
| 23:37.08 | war9407 | can dial into voice mail from pots phone |
| 23:37.14 | war9407 | and check messages, quite niec |
| 23:37.25 | war9407 | tomorrow/next week will try to figure out how to make it a menu and access two different extensions |
| 23:39.56 | war9407 | <PROTECTED> |
| 23:39.56 | war9407 | Segmentation fault (core dumped) |
| 23:39.56 | war9407 | l1:~# |
| 23:39.57 | war9407 | ouch |
| 23:40.53 | carrar | OH LUCY, you've got some splaining to do! |
| 23:41.40 | [TK]D-Fender | To the moon Alice! TO THE MOON! |
| 23:42.18 | war9407 | http://pastebin.com/m25a45c56 |
| 23:42.34 | war9407 | there is the backtrace |
| 23:45.03 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:47.22 | war9407 | damn it did again |
| 23:47.28 | war9407 | compltely reproducible ;( |
| 23:49.53 | war9407 | Leave a message, when done press the pound key. |
| 23:49.55 | war9407 | <segfault> |
| 23:51.00 | rob0 | segpound |
| 23:51.07 | chiwawa_42 | [TK]D-Fender: just wondering, what about using a Linksys SPA3102 as FXO ? Could it work as well as the TDM410P ? |
| 23:52.35 | [TK]D-Fender | chiwawa_42: iffy... harder to handle call spill-over, lack of flash control which should be avoided anyway, harder to tweak if the basics prove spotty |
| 23:52.45 | [TK]D-Fender | chiwawa_42: I wouldn't use it for primary business use |
| 23:53.03 | [TK]D-Fender | chiwawa_42: very low volume usage only |
| 23:53.06 | chiwawa_42 | it's not business actually, it's for a student house ;) |
| 23:53.20 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 23:54.13 | chiwawa_42 | despite that, i'd take your advice for the tdm410p I think, unless I could afford better, like maybe a xorcom astribank... |
| 23:58.08 | [TK]D-Fender | chiwawa_42: tzafrir_laptop here works for them, and while I have largely philosophical & economical issues with them I don't much by way of "problems". Do they have a specific model that fits your deployment needs & budget where you are? |
| 23:58.47 | chiwawa_42 | XR0030 : 6FXS+2FXO, 530⬠|
