00:01.06 | *** part/#asterisk nny (n=scott@64.203.237.47) |
00:01.36 | *** join/#asterisk lordmortis (n=lordmort@203-59-207-20.dyn.iinet.net.au) |
00:03.27 | phix | soho gear? what that? |
00:03.35 | phix | small office home office? |
00:04.15 | Chainsaw | phix: Read: Silly consumer-grade stuff |
00:04.42 | Mango | bmoraca: What soho gear specifically? |
00:06.35 | phix | ah ok |
00:06.51 | Mango | Eh, some soho gear isn't bad. |
00:06.56 | Mango | PAP2T :) |
00:07.02 | phix | hmmmm, do I want a Linksys SPA 942 or not :\ |
00:07.08 | bmoraca | Mango: all SOHO gear is terrible. |
00:07.15 | bmoraca | phix: no |
00:07.17 | Mango | phix: You can do better, likely. |
00:07.21 | voipmonk | spa 942's arent horrible |
00:07.26 | voipmonk | but the polycoms sound better |
00:07.27 | voipmonk | :) |
00:07.32 | Mango | Yeah. |
00:08.15 | Mango | The sound quality is pretty good. Far better than a POTS phone of course. But, there are other phones (Polycom) in the same price range with more features and yet others (Aastra) in a slightly lower price range, also with more features. |
00:08.30 | Mango | bmoraca: Hehe. What in particular were you working with? |
00:09.21 | phix | voipmonk: which polycom though |
00:09.29 | voipmonk | how many lines do you need? |
00:09.48 | phix | voipmonk: I can get some polycom 300 somethings, but apparntly they are worste than a SPA 942 |
00:10.02 | phix | voipmonk: 2 - 4 inclusive |
00:10.10 | voipmonk | "apparently" ? |
00:10.26 | phix | voipmonk: yes, according to [TK]D-Fender :) |
00:12.09 | voipmonk | its up to you and your budget :) |
00:12.45 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
00:14.15 | *** join/#asterisk grabes222 (n=grabes@72.20.207.237) |
00:14.19 | phix | voipmonk: hmmm well what additional features would a same price ranged polycom have over a 942? |
00:14.54 | voipmonk | i find the sound is different |
00:15.02 | voipmonk | but I personally own a spa 942 on my desk at home |
00:15.11 | voipmonk | i dont use it anymore tho :) |
00:15.14 | voipmonk | i use an iaxy |
00:15.17 | grabes222 | Anyone having any issues with cdr_adaptive_odbc not logging? Its registered, the ODBC connection is valid no errors are happening.. *1.6.1.6 |
00:15.20 | voipmonk | and a cordless |
00:15.21 | bmoraca | Mango: it was a crappy 8 port VPN "router" from NetGear. but that's besides the point. it's all garbage |
00:15.40 | phix | voipmonk: iaxy? |
00:15.51 | voipmonk | yes... :) |
00:15.53 | voipmonk | an iaxy |
00:15.59 | phix | which is/'? |
00:16.07 | phix | and what type of cordless? |
00:16.33 | bmoraca | does Pickup() work fairly well now? my last experiences with it weren't all that great... |
00:19.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
00:21.18 | *** join/#asterisk psilikon (n=psilikon@140-1.35-65.tampabay.res.rr.com) |
00:24.58 | riddlebox | bmoraca, I've never had issues with it |
00:26.44 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
00:28.50 | *** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk) |
00:36.46 | phix | hmmmm, Power over Ethernet |
00:37.07 | phix | any issues linking a POE switch to another POE switch? |
00:40.39 | Nugget | don't cross the streams |
00:41.38 | coppice | that rule is only for pissing contests |
00:41.40 | riddlebox | lol |
00:41.50 | riddlebox | and the proton packs |
00:42.34 | phix | haha |
00:42.58 | phix | That didn't actually answer my question correctly :) |
00:43.07 | phix | are you saying I should or shouldn't do that? |
00:48.31 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
00:48.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
00:50.04 | *** join/#asterisk propellerhead (n=yogurt2u@host251.200-82-124.telecom.net.ar) |
00:50.19 | drmessano | Why worry? Each of us is carrying an unlicensed nuclear accelerator on his back... |
00:50.22 | drmessano | Switch me on |
00:55.19 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
00:55.19 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
00:55.56 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
01:02.07 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
01:03.15 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
01:21.00 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
01:24.50 | *** join/#asterisk m477au (n=m477au@60-242-4-45.static.tpgi.com.au) |
01:25.17 | m477au | Anyone around able to help me with invoking a agi script? |
01:26.44 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) |
01:46.01 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
01:55.15 | *** join/#asterisk b1u3m3th (n=b1u3m3th@home.bryantfamily.ws) |
01:56.10 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
02:00.31 | *** join/#asterisk EnrGE (n=NrjChnd@202.170.42.67) |
02:04.35 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
02:05.06 | EnrGE | hi folks |
02:05.19 | Mango | Hi. |
02:05.21 | b1u3m3th | hello |
02:05.28 | EnrGE | :) |
02:07.21 | EnrGE | intro: enrge location: fj since m comparatively new to irc |
02:15.52 | drmessano | Oh a newbie |
02:16.26 | drmessano | sharpens his knives and polishes his ~ key |
02:20.45 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
02:25.30 | EnrGE | lol drmessano |
02:31.42 | carrar | w00t |
02:32.32 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
02:39.59 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
02:41.22 | dandate2 | comcast cable business services or phillipine E1 line |
02:42.30 | drmessano | They can't extradite you if you move to ecuador |
02:43.08 | dandate2 | lol |
02:43.23 | dandate2 | no i am in a small village of the phillipines ain't no extradition here |
02:43.38 | dandate2 | noone even checks visas heh |
02:44.17 | carrar | I'm in a small village also |
02:44.28 | carrar | village of a few million! |
02:44.36 | carrar | well thousands |
02:45.17 | dandate2 | my agents still complain of line issues at times, and random disconnects in the middle of the call. i'd like to just say cuz they hung up on your ass but wondering if a $1000/mo phillipine leased wire will do better than american digital |
02:45.56 | carrar | Isn't SIP service from America to the phillipines Illegal? |
02:46.17 | carrar | (in phillipine) |
02:46.33 | dandate2 | yes it is |
02:46.39 | dandate2 | but i am a foreign national |
02:46.42 | dandate2 | so noone here cares |
02:46.43 | carrar | haha |
02:46.51 | carrar | oh |
02:46.54 | carrar | then it's ok |
02:46.57 | dandate2 | yes |
02:46.59 | carrar | haha |
02:47.58 | carrar | Not sure we can support your illegal activities here |
02:48.06 | carrar | Mr. Criminal Man |
02:48.17 | dandate2 | haha |
02:48.19 | dandate2 | whatever |
02:48.23 | dandate2 | screw smart |
02:48.24 | dandate2 | and globe |
02:48.28 | dandate2 | those bastards |
02:48.43 | carrar | Father Internet would be sad |
02:48.46 | dandate2 | lol |
02:48.55 | carrar | Father Internet: http://pics.osburn.com/photo/42882/original |
02:49.09 | dandate2 | the telco companies herein the phillipines offer ridiculously bad service but have made the government outlaw all competition to them |
02:49.10 | *** join/#asterisk xpot-mobile (n=james@173.8.94.1) |
02:49.15 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
02:49.33 | dandate2 | the phillipine gov and telco companies would like to see your people living in tiki huts eating bananas and coconuts but taking pharmaceutical drugs for all time |
02:49.47 | carrar | thats not so bad |
02:49.52 | carrar | relaxed life |
02:49.52 | dandate2 | haha i know |
02:49.56 | dandate2 | thats why i like it here |
02:50.07 | dandate2 | even if i was broke i could live for free off coconuts and bananas and find free housing |
02:50.31 | dandate2 | all the barong barong here have big screen TVs, and internet service lol |
02:50.38 | carrar | yeah, if bear can live off piss, you can live off of coconuts |
02:50.43 | carrar | heh |
02:50.59 | dandate2 | but man this place is poor |
02:51.05 | carrar | parden my fowl language |
02:51.10 | dandate2 | i can buy 12 packs of cigarettes for $4 usd |
02:51.19 | carrar | bbl, dinner |
03:05.35 | EnrGE | so ppl goin to astricon next month? |
03:06.01 | russellb | i am! |
03:06.23 | EnrGE | that makes two of us. :) anyone else? |
03:08.23 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
03:14.06 | *** join/#asterisk [8none1] (n=[8none1]@c-76-22-141-39.hsd1.tn.comcast.net) |
03:16.48 | [TK]D-Fender | phix: IP-30X is functionally inferior to SPA-94X |
03:17.16 | [TK]D-Fender | phix: IP32X/33X bring the bar up there that for most use I'd rather have the Polycom. |
03:19.10 | drmessano | Astricon is for those people still using Asterisk.. I have moved on to Facebook-Over-IP |
03:19.49 | KavanS | I myspace'd over IP once |
03:20.21 | [TK]D-Fender | KavanS: Not in here, this is a FAMILY CHANNEL! |
03:20.28 | KavanS | lol |
03:20.38 | p3nguin | Pfft. IP is so 20th century. |
03:20.50 | KavanS | so when is ipv6 coming to the interwebz? |
03:21.03 | KavanS | or is ipv6 going to be vaporware like duke forever |
03:22.38 | KavanS | dandate2, that's awesome....12 packs of cigs for $4 usd |
03:22.44 | KavanS | dandate2, wtf do you do there to stay alive? |
03:22.53 | [8none1] | http://penrose.uk6x.com/ |
03:23.25 | KavanS | ahhh |
03:24.41 | denon | KavanS: IPv6 is here .. you're just not in the loop :) |
03:24.48 | denon | or should I say Linked In, with the current conversation |
03:24.49 | denon | ;) |
03:25.13 | KavanS | yeah I know it's around...but it's not like your residential isp is issuing ipv6 to customers |
03:25.19 | KavanS | unless I was unaware of this? |
03:25.28 | p3nguin | There are some. |
03:26.16 | denon | there are a few yeah |
03:27.20 | denon | comcast claims they're going to do general deployment next year |
03:27.23 | denon | and is already doing trials |
03:29.08 | KavanS | wow, that's impressive |
03:31.31 | denon | spose so |
03:31.44 | denon | endlusers wont know any different, except that their firewall won't be as well tested under 6 |
03:32.00 | denon | so we'll see a new wave of exploits while vendors trying to figure out what a : is |
03:35.07 | carrar | russellb |
03:35.14 | carrar | watch out for swineflu! |
03:35.21 | carrar | I'm suppose to go too |
03:35.33 | carrar | not sure I want too after hearing about that other geek convention |
03:35.58 | russellb | it would be totally worth it |
03:36.13 | carrar | I went 2 years ago, same place |
03:36.18 | carrar | was worth it then |
03:36.33 | russellb | ah, 2 years ago was Phoenix, but a different venue |
03:36.37 | carrar | swine flu is worth it is what you're saying? :) |
03:36.42 | russellb | yes! |
03:36.44 | carrar | haha |
03:37.04 | carrar | You'll come to my funeral? |
03:37.28 | carrar | "He was a great attendee" |
03:37.42 | carrar | kept quite |
03:50.15 | *** join/#asterisk shinao1 (n=shinao1@41.219.207.196) |
04:14.07 | *** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net) |
04:16.28 | geneticx | hello everyone..I would like to get some suggestions. I was given the task of implementing a voip solution in suriname, what would be the best: buying a digium card like a TDM400P and using regular local pots, or trying to find a VoIP provider that has a close POP to reduce latency...hummm what you guys think? |
04:17.28 | Mango | Is this for local calls or international? |
04:19.21 | geneticx | well, they are going to be using the digium card for local calling only..as far as international calling which they need, I'm thinking to peer with the asterisk box that we have here in the U.S. and route their international calls..what do you think? |
04:20.07 | Mango | You could peer with the box in the US for North American calling. Or, direct to the carrier may be an even better option. |
04:20.35 | Mango | For local calling, the less expensive option would likely be to find a VoIP provider with a close POP. |
04:20.38 | geneticx | direct to their local carrier? |
04:21.20 | Mango | However, I don't know of any quality VoIP providers in that part of the world. |
04:21.20 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
04:21.43 | Mango | But if there are one/a few that is the route I would prefer. |
04:22.05 | geneticx | yeah that would be ideal, but damn I'm with you on not knowing the quality of VoIP providers there.. |
04:24.11 | Mango | It would take some experimenting I imagine. |
04:25.01 | geneticx | since I don't really know any VoIP providers around there, I would probably prefer for them to get a regular analog line to the office for local calling, and implementing international calling somehow either with peering with the asterisk in the states, or just like you said straight to the carrier |
04:25.43 | geneticx | yeah, but don't know if I would have time to experiment =D |
04:25.55 | Mango | Lol. In that case, ya, a FXO card would likely be your best bet. |
04:26.02 | Mango | I've heard good things about Sangoma. |
04:26.15 | Mango | You may want to investigate them as well. |
04:26.39 | geneticx | Ok, sounds good enough |
04:26.59 | geneticx | I'm out, got to do lots of planning tomorrow.. =D |
04:27.14 | geneticx | thank you for your advice. |
04:28.28 | Mango | good luck! |
04:29.16 | geneticx | thanks! you will probably see me here again as the project advances..=D |
04:37.15 | Gokee2 | Hello everyone, Is there any way to see the results of a register command? Should I be showing some stuff in "sip show registry"? I found a few posts on the mailing list about the sipgate register command not working and that causing incoming calls to not work. Also I noticed if I command out the register command nothing changes, outgoing still works incoming doe not. |
04:39.00 | [TK]D-Fender | Gokee2: If you don't give them an IP and you don't register then you aren't going to get calls. |
04:39.06 | [TK]D-Fender | ~sipregister |
04:39.06 | infobot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
04:40.20 | Gokee2 | [TK]D-Fender, I have a register command I just don't think its working... |
04:40.57 | [TK]D-Fender | Gokee2: And what does "sip show registry tell you"? |
04:41.17 | [TK]D-Fender | Gokee2: And what does "sip show registry" tell you? |
04:41.34 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
04:41.53 | *** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net) |
04:42.22 | Gokee2 | [TK]D-Fender, Wow, after a few days of fighting with it I just realized something... My register command was not in the general section! I changed it and now sip show registry shows stuff |
04:43.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
04:43.35 | Gokee2 | And now incoming calls work! |
04:43.53 | [TK]D-Fender | \o/ |
04:45.07 | superbeef | If i wanted a sandbox enviornment, would it be possible to have 2 PBXs and have a T1 card in each connected direclty to each other? |
04:45.34 | Gokee2 | My, I feel really stupid wasting days over that little problem |
04:45.37 | denon | superbeef: of course |
04:46.08 | superbeef | cool |
04:46.22 | superbeef | i need to do that.. i'm getting burned switching from zaptel to dahdi with a sangoma card |
04:52.12 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
04:57.12 | *** join/#asterisk mintos (n=mvaliyav@209.132.186.254) |
05:04.47 | m477au | trying to initiate a call from the asterisk manager |
05:22.28 | tzafrir_laptop | Action: Originate |
05:29.39 | m477au | yeah |
05:29.44 | m477au | sorry, forgot I started typing something |
05:29.49 | m477au | keep getting this in the logs |
05:29.52 | m477au | == Starting SIP/6000-0da30a60 at from-internal,SIP/iinetout/0402686828,1 failed so falling back to exten 's' |
05:30.00 | m477au | that's for internal to external |
05:30.14 | m477au | however if I call something on the iinetout trunk to an internal extension |
05:30.15 | m477au | it works |
05:30.56 | m477au | fputs($socket, "Action: Originate\r\n" ); |
05:30.57 | m477au | <PROTECTED> |
05:30.57 | m477au | <PROTECTED> |
05:30.57 | m477au | <PROTECTED> |
05:30.57 | m477au | <PROTECTED> |
05:30.57 | m477au | <PROTECTED> |
05:34.14 | kaldemar | m477au: SIP/iinetout/0402686828 is not an extension |
05:35.10 | m477au | even with just 04 it does the same thing |
05:35.50 | m477au | however, if I change it to *97 |
05:35.54 | m477au | it connects me to voicemail fine |
05:37.44 | kaldemar | m477au: the exten needs to be a matching extension in from-internal |
05:42.34 | m477au | ok, so I need to set up a context for this to work as desired |
05:44.02 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
05:44.41 | Gokee2 | Whats the difference between TDM402B and the TDM410P? I also see "Wildcard" from time to time, whats that mean? Anyone know of a list of all the digium cards? |
05:45.17 | kaldemar | m477au: no, you need to call the right extension |
05:47.35 | *** join/#asterisk pfn (n=pfnguyen@66.245.252.239) |
05:48.42 | kaldemar | Gokee2: give links that refer to those, since digium doesn't have products by those names |
05:49.49 | m477au | kaldemar, I'm not calling an extension, I'm calling an external number |
05:49.52 | *** join/#asterisk monstertruck (i=korolev@c-75-74-122-15.hsd1.fl.comcast.net) |
05:50.32 | *** join/#asterisk |Cybex| (n=John@212.178.82.26) |
05:50.36 | kaldemar | m477au: with originate, you call an extension. extension is a number in your dialplan. |
05:54.50 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
05:54.58 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
05:54.59 | Gokee2 | kaldemar, For the TDM402B http://www.cetusvoip.com/product_info.php?products_id=1801 and http://www.soho-voip-phone.com/Asterisk_Hardware/digium_Digium_TDM402B.html wildcard I have seen http://www.ipphone-warehouse.com/Digium-Wildcard-TDM02B2-p/tdm02b.htm I forget where else I have seen wildcard but I know I have seen it somewhere else as well |
05:56.32 | kaldemar | Gokee2: that TDM402B has 2 FXO modules. |
05:57.06 | m477au | kaldemar: found the problem, lack of callerid |
05:57.18 | m477au | <PROTECTED> |
05:57.19 | m477au | <PROTECTED> |
05:57.19 | m477au | <PROTECTED> |
05:57.49 | m477au | soon as I added a callerid, it worked. |
05:57.58 | kaldemar | Gokee2: that TDM02B2, based on the picture, is a discontinued analog card model. |
05:59.39 | Gokee2 | kaldemar, Ah, I guess I should find a non-discontinued card to buy then |
05:59.55 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
06:00.08 | kaldemar | Gokee2: wildcard is just a name part in some models. |
06:00.27 | kaldemar | Gokee2: look at the ones listed here: http://www.digium.com/en/products/ |
06:02.16 | Gokee2 | Why do I see some TDM410p's? All thats listed there (for analog 4 ports) seems to be the TDM410? |
06:03.06 | kaldemar | what do you mean? |
06:03.46 | Gokee2 | Like say here http://cgi.ebay.com/Digium-TDM410P-with-2-FXO-ports_W0QQitemZ300346351007QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item45ee099d9f&_trksid=p3286.c0.m14 lits it as a TDM410P |
06:04.43 | kaldemar | Gokee2: so what? it is a TDM410. |
06:05.00 | Gokee2 | kaldemar, Then whats the "P" for? |
06:05.05 | *** join/#asterisk tm1985 (n=tm@082-146-101-077.stat.adsl.xs4all.be) |
06:05.27 | kaldemar | TDM410 is the card itself. you can equip it with modules: http://www.digium.com/en/products/analog/s400m.php |
06:05.44 | kaldemar | resellers call their combinations whatever they wish |
06:05.50 | Gokee2 | telephonydepot.com also lists it as a TDM410P? Can I get a non-P version? |
06:06.10 | Gokee2 | (telephonydepot.com has no modules) |
06:06.22 | kaldemar | Gokee2: there's no P in TDM410. completely different digital cards have P in their model. |
06:06.55 | kaldemar | it's just a letter they chose to put in an item. it means nothing. |
06:07.03 | Gokee2 | kaldemar, Ok, I won't try and make since of all the different stuff at the end of the card name then |
06:07.59 | kaldemar | don't look at the model names, try to figure out what they're actually selling you instead. |
06:08.24 | Gokee2 | ok |
06:14.29 | *** join/#asterisk Woody2143 (n=Woody214@209.244.4.189) |
06:15.06 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
06:18.25 | *** join/#asterisk fiddur (n=fiddur@dhcp08.textalk.com) |
06:29.31 | *** join/#asterisk xrmx__ (n=rm@host197-226-dynamic.1-79-r.retail.telecomitalia.it) |
06:39.00 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
06:39.20 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
06:39.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
06:40.30 | *** join/#asterisk gardo (n=gardo@121.97.109.223) |
06:43.44 | tm1985 | can anybody help with a dahdi issue? |
06:50.37 | tm1985 | when I got a incoming call it gives for example 13121110 on my phone display but I need to have a 0 in front of them so that I get 013121110 |
06:52.26 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
06:52.39 | kaldemar | tm1985: modify callerid in your dialplan. Set(CALLERID(num)=0${CALLERID(num)}) |
06:58.59 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
06:59.30 | *** join/#asterisk kc2tnk (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
06:59.52 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
07:00.00 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
07:03.05 | superbeef | Is this invalid in asterisk 1.4? exten => _.,2,Goto(from-internal|BYEXTENSION|1) |
07:04.28 | *** join/#asterisk af_ (n=getsmart@88-149-241-21.dynamic.ngi.it) |
07:04.31 | *** join/#asterisk AL-Hadi (n=User@unaffiliated/al-hadi) |
07:04.36 | tm1985 | kaldemar, when I do that I want to call a queue then I get moh and no ringtone |
07:04.39 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
07:04.50 | kaldemar | superbeef: BYEXTENSION hasn't been valid for years |
07:05.08 | superbeef | kaldemar: lol... context rules from really old asterisk installs |
07:05.17 | superbeef | i just rebuilt one of them,, now it's not playing nice |
07:05.19 | kaldemar | superbeef: replace it with ${EXTEN}. and replace _. with something else, e.g. _X. |
07:06.01 | superbeef | it has _X my paste was just bogus |
07:06.36 | superbeef | so..... _X.,2,Goto(from-internal|${EXTEN}|1) |
07:06.57 | kaldemar | tm1985: so you wan't ring tone? "core show application Queue" in CLI will give you options for app Queue. you'll find the answer there. |
07:07.28 | tm1985 | no I need a ring tone but I don't get it? |
07:08.00 | kaldemar | s/wan't/want/ |
07:08.16 | kaldemar | tm1985: read that again |
07:08.24 | tm1985 | oke |
07:10.19 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
07:10.19 | superbeef | kaldemar: hey thanks, that fixed it.. now I can sleep! |
07:10.38 | tm1985 | kaldemar thx found it |
07:12.45 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
07:14.32 | tm1985 | kaldemar if you dial with DAHDI you need for example Dial(DAHDI/1/${EXTEN}). Can you change the 1 from channel with a asterisk variable like the ${EXTEN} |
07:15.12 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
07:15.41 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
07:15.58 | kaldemar | tm1985: sure |
07:16.06 | tm1985 | what is it called |
07:17.00 | kaldemar | you're making no sense now. you need to define a channel in the dial. there is no channel until you do so. |
07:17.31 | *** join/#asterisk Grof (n=dule@89.201.165.226) |
07:17.32 | kaldemar | if you want the current channel, it is stored in CHANNEL, but you can't dial the same channel you're already using. |
07:19.01 | kaldemar | what you might be looking for is groups in chan_dahdi.conf. you can assign multiple channels to a group and then use for example Dial(DAHDI/g0/${EXTEN}) to dial using some available channel in the group. |
07:20.06 | tm1985 | ah I will try this |
07:28.28 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
07:31.53 | *** join/#asterisk datacompboy (n=datacomp@213.187.251.250) |
07:32.09 | datacompboy | Hi all! :) Is there method to prefix dialed extension from particular peer ? |
07:32.49 | datacompboy | i.e. all calls from [peer1] come to [general] section as-is; but all calls from [peer2] come to [general] as p-${EXTEN} |
07:33.12 | datacompboy | main problem is extensions not numbers, but alpha-numeric |
07:33.26 | tm1985 | kaldemar thx that is what I needed |
07:35.14 | kaldemar | datacompboy: make an own context for peer2 where you add the prefix and the go to [general] |
07:35.28 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
07:35.31 | datacompboy | kaldemar: well, how to match any alpha-numeric extensions, |
07:35.48 | datacompboy | ? if i do exten => _., -- it match also s, h internal extensions |
07:37.12 | kaldemar | datacompboy: _[a-g,jr] <-- like that |
07:37.28 | kaldemar | so don't do _. |
07:38.07 | datacompboy | but what if incoming call will be to s@gate ? how to separate internal "s" and incoming "s" ? that is main problem |
07:40.30 | kaldemar | with different contexts |
07:41.03 | kaldemar | or just give yourself a break and don't allow arbitrary alpha-numeric extensions |
07:41.37 | datacompboy | kaldemar: that are SIP<->Skype gate. so ther will be arbitraty alpha-numeric (and dots, and etc) extensions... |
07:43.37 | *** join/#asterisk Tim_Toady (n=moi@adsl194-8.kln.forthnet.gr) |
07:44.17 | *** join/#asterisk Micc (n=Micc@c-71-231-123-28.hsd1.wa.comcast.net) |
07:45.05 | Micc | Is there a way to park a call in a specific parking spot instead of using the park exten which reads it back to you? |
07:45.36 | tm1985 | kaldemar do you know something of redirect calls to mobile numbers with dahdi? |
07:47.05 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
07:47.16 | kaldemar | tm1985: please elaborate |
07:49.37 | tm1985 | We have a isdn provider an we are possible to redirect calls to for example a mobile phone by dialing *21*TELNR# |
07:50.18 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:e0b2:150:ec3e:2a25) |
07:50.19 | *** join/#asterisk user4545 (n=sipip@p57B1F38A.dip.t-dialin.net) |
07:50.44 | tm1985 | Now I want to do exten => 600,1,Dial(DAHDI/g0/*21*telnum#) but that doesn't work do you know how I have to do that |
07:51.45 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
07:52.33 | *** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
07:52.57 | user4545 | Hi, i have a problem... if I call to my number by Sipgate, then I become message in asterisk: "Call from '1234567' to extension '1234567' rejected because extension not found." |
07:53.02 | user4545 | why? |
07:53.10 | user4545 | can any me help? |
07:53.27 | tm1985 | Do you know something about that? |
07:54.12 | kaldemar | tm1985: well, not unless you tell how it doesn't work. |
07:55.56 | kaldemar | user4545: you need to add the extension in your dialplan. |
07:56.19 | user4545 | I have it extension |
07:56.29 | tm1985 | http://pastebin.com/d50aed328 here are my errors |
07:56.58 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
07:57.04 | tm1985 | http://pastebin.com/d5a159049 |
07:57.11 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
07:57.12 | tm1985 | that are my exten for the 600 |
07:57.47 | user4545 | kaldemar: I have it extension in my dialplan |
07:58.06 | kaldemar | user4545: not in the right context, show your dialplan and the relevant sip.conf context |
07:58.15 | tm1985 | http://pastebin.com/m5ebc2c38 => exten |
07:59.07 | kaldemar | your cli output doesn't match the dialplan |
08:01.43 | tm1985 | The wierd thing is when i call a sip are playback a soundfile It that exten works but when want to dial that *21*TELNR# I get the error that extension isn't know |
08:02.01 | user4545 | kaldemar: http://pastebin.ca/1568049 my dialplan please see it |
08:10.41 | tm1985 | Hello need some help with DAHDI. I have provider that provides ISDN and not VOIP. Our provider give us also to chance to redirect to for example a mobile phone with *21*TELNR# |
08:11.58 | tm1985 | I'm trieing to dial this with exten 600 but always get this error : http://pastebin.com/d50aed328 |
08:12.05 | tm1985 | can someone help me??? |
08:13.09 | tm1985 | http://pastebin.com/d50aed328 this Is that exten 600 |
08:13.56 | tm1985 | sorry this is the exten 600: http://pastebin.com/m50b7a4d4 |
08:16.59 | *** join/#asterisk wathek (n=wathek@41.224.132.54) |
08:18.17 | tm1985 | is nobody here who can help me? |
08:25.02 | tm1985 | Hello need some help with DAHDI. I have provider that provides ISDN and not VOIP. Our provider give us also to chance to redirect to for example a mobile phone with *21*TELNR# |
08:25.13 | tm1985 | I'm trieing to dial this with exten 600 but always get this error : http://pastebin.com/d50aed328 |
08:25.25 | tm1985 | can someone help me with this??? |
08:27.22 | *** join/#asterisk xrmx__ (n=rm@host197-226-dynamic.1-79-r.retail.telecomitalia.it) |
08:34.25 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
08:34.27 | Polysics | hello |
08:34.38 | Polysics | i finally managed to fix the sip users setup |
08:35.05 | Polysics | since i am using mysql for the sip users, do you recommend using db for the extensions too? |
08:35.08 | tzafrir_laptop | tm1985, what is '*21*' ? A valid number? |
08:35.22 | Polysics | it is going to become a 200ish users distributed service |
08:36.01 | tm1985 | What do you understand under valid Number? |
08:36.38 | Polysics | so far everything is working with static extensions |
08:37.33 | Polysics | btw, if i put single user extensions in db, can i still use static extensions for some things? |
08:40.16 | tm1985 | http://pastebin.com/m6b378094 |
08:40.36 | tzafrir_laptop | tm1985, you try calling that number . Is it a number you can actually dial to? |
08:41.22 | tzafrir_laptop | *21*TELNR# is not a valid phone number |
08:42.08 | tm1985 | no there we have set our number that we want to dial but I change that so that not everybody has my personel mobile number |
08:42.15 | tzafrir_laptop | You probably meant *21*83567 |
08:42.21 | tzafrir_laptop | err... |
08:42.27 | tzafrir_laptop | You probably meant *21*83567# |
08:42.39 | tm1985 | yes |
08:42.40 | user4545 | Hi, i have a problem... if I call to my number by Sipgate, then I become message in asterisk: "Call from '1234567' to extension '1234567' rejected because extension not found." |
08:42.59 | tzafrir_laptop | I also suspect that '#' is not part of the number |
08:43.00 | user4545 | but I have this extenstion |
08:43.36 | tzafrir_laptop | Does Sipgate have this extension? |
08:43.37 | tm1985 | yes for redirecting to that number you have to dial *21*83567# for example |
08:43.40 | wathek | any one would help me to test my Asterisk configuration please ? |
08:45.07 | tm1985 | the # is needed tzafrir!!! |
08:45.36 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
08:45.41 | user4545 | <PROTECTED> |
08:46.03 | user4545 | see it |
08:46.05 | user4545 | http://pastebin.ca/1568049 |
08:50.23 | *** join/#asterisk mort_gib (n=mjensen@195.166.201.234) |
08:50.27 | user4545 | can anybody help me? |
08:50.27 | tm1985 | tzafrir the # is needed in the phone number |
08:52.43 | mort_gib | Crackling noises on incoming call but not outgoing?? Any ideas?? |
08:53.32 | Polysics | doesn anyone used realtime extensions with mysql? |
08:53.53 | Polysics | if i move them there, will the extensions in leave in extensions.conf still work? |
08:56.30 | tm1985 | Does anyone knows how I can dial *21*TELNR# with dahdi? |
09:02.30 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
09:08.38 | renzoe | guys, can you enlighten me on what PRI line is? the digium reseller is recommending us to have PRI for 60-70 office users |
09:08.49 | tzafrir_laptop | tm1985, please enable pri debug an show a trace of the attempted call |
09:09.22 | wonderworld | renzoe: PRI is a multiplexed ISDN. you get about 30 ISDN channels from one line, depending on where you live |
09:09.40 | wonderworld | i think in the US it's 28 channels |
09:09.46 | wonderworld | in europe 32 or so |
09:10.05 | renzoe | is this PRI can have a unique telephone number per ip phone? |
09:10.11 | renzoe | here in the UAE they said 30 |
09:10.13 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
09:10.14 | tm1985 | how do you enable that debug tzafrir? |
09:10.16 | wonderworld | so you wouldn't need to book 30 ISDN-lines for your 60 office users |
09:10.52 | renzoe | 30 channels meaning 30 simultaneous inbound/outbound? |
09:10.53 | wonderworld | you normaly get a 100-number number-block like 1111111-0 to 1111111-99 |
09:11.07 | wonderworld | renzoe: yes, 30 "real" ISDN lines |
09:11.33 | wonderworld | you have to check the details with your telco. no idea about how things are in the UAE |
09:12.24 | renzoe | becasue we are thingking of uilding our own asterisk based system the first one we checked is setting up 8 ports digium AEX800 cards with echo canceller. which do you think is the mest setup? |
09:12.47 | renzoe | *best |
09:13.15 | *** part/#asterisk NicoB (n=nibou@front2.nbi.fr) |
09:13.26 | wonderworld | nope, the AEX800 card is a card for analogue phones |
09:13.37 | wonderworld | you need a card that supports E1/T1 |
09:13.39 | renzoe | wonderworld. yes he told me that also that i will get 100 numbers |
09:14.20 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
09:14.32 | wonderworld | http://www.voipsupply.com/dgm-te205p |
09:14.42 | wonderworld | thats a digium card for T1 lines |
09:14.48 | renzoe | so a cisco ip phone will not be compatible with AEX800? really sorry just new to this pbx thing |
09:15.07 | Chainsaw | renzoe: Cisco IP phones work on ethernet. |
09:15.26 | wonderworld | yes, you first need to connect the asterisk-pc to your telco somehow |
09:15.29 | Chainsaw | renzoe: The only telephony adapter you need in your server is one to connect to your ISDN line(s). |
09:15.41 | wonderworld | how you connect your phones in your company with asterisk is a different thing |
09:16.21 | renzoe | i see. now getting some light |
09:16.39 | wonderworld | like [telco providing T1 line ] --> [Asterisk ] -> [LAN] -> [Your phones] |
09:16.54 | renzoe | :) so i need to accomplish forst on how to connect my asterisk to my telco |
09:17.10 | wonderworld | probably yes :) |
09:17.52 | wonderworld | check if your cisco phones can do SIP.... that would be one standard wy to connect them to asterisk |
09:18.18 | renzoe | but will the AEX800 can do the thing? we also in a tight budget. if either can do the same thing then the next think i will look is the cost |
09:18.32 | tm1985 | http://pastebin.com/m8fd7417 |
09:19.26 | wonderworld | nope. the AEX800 is just for connecting analog telephone lines or analog phones to asterisk. your T1 lines isn't analog and your Cisco phones aren't either. |
09:19.29 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
09:19.59 | tm1985 | Does anyone knows how I can dial *21*TELNR# with dahdi? |
09:21.14 | wonderworld | tm1985: never did that, what happens when you just dial it with dahdi? |
09:21.17 | renzoe | i see got it. by the way how much is the PRI in your area? i already sent an inquiry but it really takes time to reply since its ramadan here |
09:21.37 | wonderworld | i live in germany and i can get a PRI for €99 a month |
09:21.48 | wonderworld | (just for the line, no calls included) |
09:22.18 | tm1985 | It doesn't work don't know how I have do that in asterisk and dahdi |
09:22.35 | renzoe | is that euro? |
09:22.38 | tm1985 | And can't find any information also |
09:23.03 | wonderworld | yes, EURO |
09:23.37 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
09:23.57 | wonderworld | tm1985: what do you want to do with that code? |
09:24.22 | renzoe | thanks a low wonderworld. now i just waiting for the supplier to reply |
09:24.28 | *** join/#asterisk Supermule1980 (n=test@smtp.techbiz.dk) |
09:24.40 | wonderworld | no problem.... |
09:24.51 | tm1985 | to redirect calls to that number |
09:25.28 | wonderworld | where is the call coming from? |
09:25.43 | tm1985 | outside |
09:26.05 | wonderworld | and it sould go to another outside number? |
09:26.27 | tm1985 | indeed |
09:26.32 | wonderworld | ok |
09:26.49 | wonderworld | i don't really get why you would need the *21*123# |
09:26.59 | wonderworld | why not let dahdi just dial out to that number? |
09:27.20 | wonderworld | * would switch the calls together |
09:27.22 | renzoe | by the way wonderworld, once the PRI card is installed its already capable of call listen/intrude and recording? or asterisk is the one handling it? |
09:27.46 | wonderworld | asterisk is doing that |
09:27.52 | wonderworld | look into chan_spy and mixrecording |
09:28.17 | wonderworld | be sure to buy a card with a hardware echo canceler too |
09:29.30 | renzoe | is that an add-on to the PRI card? or a separate PCI card? |
09:29.35 | tm1985 | to redirect automaticlly in the weekends |
09:30.18 | wonderworld | tm1985: you can do that with dialplan logic if your approach fails |
09:30.25 | renzoe | i saw the echo cancellation in digium and its an add-on to the aex800 |
09:31.20 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
09:31.35 | mbrevda | is seeking uk businsess grade did's |
09:32.45 | wonderworld | tm1985: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours |
09:34.06 | wonderworld | renzoe: thats an add-on to the pci-card. some merchants sell the card with it some sell it without |
09:34.17 | wonderworld | be sure to get one with EC, because you will need it |
09:34.35 | wonderworld | asterisk can do software EC but it doesn't work as well as a hardware solution |
09:34.44 | wonderworld | you will have echos if you don't get a hardware EC |
09:37.16 | wonderworld | to save you a lot of time and hassle ask your telco as well about their signalling settings (you'll have to put them into the config-file for your pci-card) |
09:46.50 | tm1985 | Does anyone knows how I can dial *21*TELNR# with dahdi? |
09:47.14 | tm1985 | this for redirection to a mobile phone |
09:48.08 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
09:50.12 | tzafrir_laptop | tm1985, for starters, 'T', 'E' etc. are not digits you can dial |
09:50.57 | tm1985 | for TELNR we use a number like 049856862 for example |
09:51.46 | tm1985 | we don't use *21*TELNR# we use for example *21*045632869# for example |
09:52.33 | wonderworld | tm1985: as i told you, for the thing you want to do http://www.voip-info.org/wiki/view/Asterisk+tips+openhours would probably just work fine |
09:53.51 | tm1985 | But we want to set it one when the last one leaves and back off when the first person come back we can't predict when does hours are exactly |
09:54.40 | wonderworld | ok, then you can create an extension like 9999 or something that just writes a variable to the asterisk DB like "0" for normal office hours and "1" for night service |
09:55.05 | wonderworld | you can read that var in your extension and send calls to the right place afterwards |
09:55.27 | wonderworld | like people can call "9999" on their phone when they leave |
09:55.37 | wonderworld | and calls will be rerouted after that |
09:58.06 | tm1985 | That is not what we really looking for |
10:00.18 | wonderworld | k then i didn't get you, sorry |
10:05.37 | *** join/#asterisk AlHafoudh (n=AlHafoud@chello089173071159.chello.sk) |
10:10.06 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
10:15.38 | *** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl) |
10:17.16 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
10:17.37 | cjk | hi, what is the best way to disable musiconhold for zap channels? |
10:20.08 | *** join/#asterisk dakota (n=chatzill@myw-stp-66-18-80-214.sentechsa.net) |
10:21.41 | *** join/#asterisk gardo (n=gardo@121.97.136.60) |
10:21.56 | *** join/#asterisk [netman] (n=netman@216.Red-88-17-240.dynamicIP.rima-tde.net) |
10:22.32 | [netman] | hi all, I got an "Unable to support trunking .... without zaptel timing" error , but my ztdummy module is loaded. Any suggestions, please? |
10:25.12 | angryuser | [netman], type "zaptel show status" in CLI look if it really On |
10:25.24 | angryuser | it is* |
10:26.38 | [netman] | thx angryuser , give a minute, plz |
10:26.39 | garymc | Hi I altered some default passwords, couldnt get into my gui. Sorted that now. But my 2 phones show as connected, but they cant call each other anymore. Here is an asterisk output. http://pastebin.ca/1568116 Can anyone tell me whats going on here? |
10:27.14 | wonderworld | cjk: do you use the m option in your dial command? |
10:27.42 | kaldemar | garymc: go to #freepbx |
10:27.59 | [netman] | angryuser: that is. Thank u very much |
10:28.01 | cjk | wonderworld, no |
10:28.06 | garymc | Thought it coulda been an asterisk thing Kaldemar |
10:28.17 | garymc | ive put it in there also :S |
10:30.46 | garymc | Its ok i fixed it |
10:30.49 | garymc | sorry |
10:31.26 | wonderworld | cjk: for whom do you want to disable it? people calling you from outside to a sip-phone inside? |
10:34.25 | *** join/#asterisk khussein78 (n=khussein@dogbert.palnet.com) |
10:34.33 | *** part/#asterisk dakota (n=chatzill@myw-stp-66-18-80-214.sentechsa.net) |
10:50.44 | *** join/#asterisk cjk (n=cjk@85.93.204.22) |
10:56.35 | garymc | anyone got a minute to test my sip extension? |
10:56.45 | garymc | using zoiper or some other softphone? |
10:57.02 | garymc | just opened ports you see :S |
11:01.57 | tm1985 | Does anyone knows how I can dial *21*TELNR# with dahdi? |
11:03.04 | *** join/#asterisk Intel`` (n=clc@213.132.40.2) |
11:03.22 | Intel`` | hi guys, which do you prefer. setting up fxo or setting up isdn? |
11:06.51 | cjk | wonderworld, people from the outside calling activate my moh which i hear instead of theirs |
11:08.03 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
11:08.11 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
11:09.05 | garymc | im gonna be setting up isdn so ill choose isdn :P |
11:09.08 | tm1985 | Does anyone knows how I can dial *21*TELNR# with asterisk and dahdi? |
11:16.37 | Intel`` | garymc what are the avantages? i dont know if isdn will be too much for our requirements |
11:17.23 | garymc | ISDN30 is cheaper for me than a good broadband connection. I can get 30 channels down one cable |
11:17.46 | garymc | But at the minute im only gonna have the smallest option and thats 8 channels |
11:18.21 | garymc | so its like £300 per quarter. But if i want a 2mb up and down broadband they want £1200 per quarter |
11:18.39 | Intel`` | by the way im new to pbx and i wanted to know the meaning of "channel" and "lines" because usually that's the first one sales are asking |
11:19.10 | Intel`` | they were asking me how many |
11:19.33 | garymc | channels are calls down one line |
11:20.30 | garymc | eg. I got ISDN30 with 8 channels. ( ican have 30 channels if want but need to pay more per quarter) this means i can have 8 simultaneous calls at the same time down the one line |
11:20.50 | garymc | so i can have 8 workers on the phone to differnt customers at once |
11:21.19 | fiddur | tm1985: You keep asking the same question. If people doesn't answer, try putting the question in a different way... Maybe you just want exten => _*21*X!#, 1, Dial(DAHDI/g0/${EXTEN}) ? dialling * and # really shouldn't be any different from any number. |
11:21.50 | Intel`` | channels are i see i see so if i will get 10 lines with lets say 8 channels each, that's 10*8 users in a call simultaneously? |
11:23.04 | garymc | yeah thats 80 ncalls at the same time |
11:23.06 | garymc | I think |
11:23.12 | garymc | *calls |
11:23.16 | Intel`` | yup |
11:23.24 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
11:24.09 | Intel`` | but can i get 1 line with lets say 20 channels? will that get clooged up or something? |
11:25.21 | Intel`` | that's why you distribute it on multiple lines? |
11:25.44 | tm1985 | Has anybody ever just to dial *21* in asterisk? |
11:26.29 | fiddur | tm1985: Yes, that's not a problem. |
11:27.40 | tm1985 | with that exten that you entered above? |
11:29.05 | tm1985 | Because that don't work with me!!! |
11:29.15 | fiddur | tm1985: JUST *21* is just Dial(DAHDI/g0/*21*) ...but the provider then checks if that is a valid number or not. Most probably, it will answer that it's not a valid numer, e.g. hangupcause 28 |
11:29.58 | fiddur | tm1985: If the provider wants *21*xxxxxxxxxx#, then you have to include the whole number, not just *21* |
11:30.00 | *** join/#asterisk doolittlework (n=user@196-209-90-86-rrba-esr-3.dynamic.isadsl.co.za) |
11:30.04 | doolittlework | hi htere |
11:30.55 | *** join/#asterisk gsiener (n=gsiener@d-63-245-116-186.batelnet.bs) |
11:31.06 | fiddur | tm1985: lines 6-7 in your paste http://pastebin.com/d50aed328 shows that the provider doesn't like the number *21* in itself. |
11:31.07 | tm1985 | I have set exten => 600,1,DIAL(DAHDI/g0/*21*0489562356#) but that doesn't work |
11:31.54 | fiddur | tm1985: In the paste that's not the exten used... maybe you missed a 'dialplan reload'? |
11:32.30 | doolittlework | i want to record my sip channel if i use the mixmonitor works fine, but once i end the call it ends the monitor and end the call, how can i get monitor called played back 2 me once i end the call? |
11:32.46 | tm1985 | If i do anything else but the *21* dial it see the exten 600 and use it |
11:34.10 | wonderworld | tm1985: maybe you have to pause for a short amount of time after the *21* ? |
11:34.36 | wonderworld | for your telco to recognize it? |
11:34.42 | fiddur | tm1985: In http://pastebin.com/m8fd7417 it states clearly that "Ext: 1  Cause: Invalid number format (28)," |
11:34.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
11:35.18 | fiddur | tm1985: I think this is really a question for your provider, not a question of how to get asterisk to send the number. |
11:35.39 | tm1985 | aahhh oke |
11:36.46 | tm1985 | I will try with a pause |
11:37.06 | wonderworld | i wouldn't know how to do it though |
11:37.47 | Intel`` | what's the difference between free and commercial asterisk softwares? |
11:38.05 | wonderworld | i think on cell-phones you can hold the "0" key for a while and it will put a "p" into the number. that does pause a second or so. not sure how to do that with asterisk. |
11:38.32 | doolittlework | i want to record my sip channel if i use the mixmonitor works fine, but once i end the call it ends the monitor and end the call, how can i get monitor called played back 2 me once i end the call? |
11:38.51 | fiddur | tm1985: From Dial: "If you need a .5 second pause while dialing a number you can insert a w in the appropriate place. " |
11:39.55 | fiddur | ...but since they said invalid number for *21* I doubt that will work |
11:39.59 | Intel`` | guys. is call monitor,barge,whisper not available on asterisk based pbx? just digium appliance? |
11:40.12 | tm1985 | I doesn't work |
11:40.31 | Intel`` | i am reading a comparison and the supplier tells me that its not. but just want to confirm |
11:40.55 | kaldemar | Intel``: yes they are |
11:41.08 | wonderworld | i am having a problem with transfering calls. i assigned *1 and *2 in features.conf to enable people to transfer calls. it works, but not in a reliable way. every 5th try or so fails. they hear "transfer" they enter the extension but the call is never transfered. any idea how to debug that? |
11:41.39 | wonderworld | i played around with the timeouts and set them real high for testing but it didn't fix it |
11:41.58 | tm1985 | fiddur My provider offers the *21*NUMBER# for redirecting calls |
11:42.11 | wonderworld | phones are SNOM hardware sip-phones |
11:42.14 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
11:42.41 | Naikrovek | yawns |
11:43.10 | fiddur | tm1985: Well, from your logs they reject that particular number in your particular case. You'd better ask them. |
11:43.40 | tm1985 | when I use a phone outside asterisk and dial *21*NUMBER# it works |
11:44.56 | wonderworld | tm1985: try to use it with the redial-function of that "outside" phone and check if it still works.... to verify that it's not about pausing after the special combination |
11:46.45 | kaldemar | tm1985: what phone are you using outside asterisk? |
11:46.57 | kaldemar | tm1985: are you using the same ISDN line? |
11:47.13 | tm1985 | yes |
11:48.03 | tm1985 | we are using the samen isdn line |
11:48.33 | kaldemar | with an ISDN phone? |
11:49.13 | tm1985 | In asterisk i'm using IP phone SPA962 |
11:49.18 | fiddur | tm1985: And normal calls from asterisk to other numbers work on DAHDI/g0? |
11:49.29 | tm1985 | yes |
11:49.55 | wonderworld | maybe it's about "pridialplan" in chan_dahdi.conf |
11:50.27 | fiddur | yes, it could be dahdi that says invalid number format, without even sending it to the provider perhaps... |
11:50.41 | fiddur | I'm not enough versed in dahdi i'm afraid... |
11:50.54 | wonderworld | me neither, just guessing |
11:50.58 | wonderworld | check if thats set to unknown |
11:51.02 | tm1985 | what do you have to do with pridialplan |
11:51.24 | tm1985 | yes but it's set into comment |
11:52.08 | tm1985 | does it need to be unknown or not? |
11:53.25 | wonderworld | i way really just guessing. if it's commented out, it probably already is unknown as that seems to be the default |
11:54.09 | wonderworld | you might want to try to play around with other values for pridialplan and prolocaldialplan |
11:54.10 | fiddur | tm1985: pridialplan default is national according to http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf |
11:54.25 | wonderworld | not sure if it would do anything good but worth the try |
11:54.56 | wonderworld | you have to do a "dahdi restart" in cli after doing changes to chan_dahdi.conf |
11:55.09 | wonderworld | watch the cli, i think for some changes to dahdi you even have to restart asterisk |
11:59.06 | tm1985 | I tried national and unknown but result stays the same |
11:59.46 | wonderworld | k, now i am really out of ideas.. |
11:59.56 | wonderworld | maybe try posting your problem on some forums. |
12:00.05 | wonderworld | or ask again later |
12:00.22 | tm1985 | I put it on several forums |
12:01.05 | tm1985 | I searching for several days now |
12:01.08 | wonderworld | what happens when you dial the number from a phone atatched to asterisk instead of dialing it automatically from the dialplan? |
12:02.18 | wonderworld | i know, should be exactly the same thing, but who knows.... |
12:02.40 | tm1985 | I type *21 and then I get that there is no extension *21 |
12:03.54 | wonderworld | create one for it and try again |
12:04.02 | tm1985 | Call from '40' to extension '*21' rejected because extension not found. |
12:04.20 | *** join/#asterisk bboness (n=bones@acdc.internet.ao) |
12:05.02 | wonderworld | lile exten => _*21X.,1, |
12:05.22 | bboness | Is there any way to define the source ip address when talking to a sip peer? |
12:05.30 | *** join/#asterisk mwalling (i=mwalling@97.107.128.165) |
12:05.38 | fiddur | tm1985: If you call *21* from a normal phone, is it answered first before you give the rest of the number + # ? |
12:06.00 | tm1985 | fiddur yes |
12:07.20 | fiddur | tm1985: You could try Dial(DAHDI/g0/*21*,,D(0489562356#)) perhaps... then it waits until answer, rather than waiting exactly .5 seconds... |
12:08.13 | wonderworld | fiddur: THAT sounds good |
12:08.18 | fiddur | but still.. since *21* gave hangupcause 28, I don't see that it could work... |
12:10.02 | tm1985 | I get the same problem!!!! |
12:11.48 | tm1985 | exten => 600,1,Dial(DAHDI/g0/*21*,,D(0498506822#)) |
12:11.58 | tm1985 | In extensions.conf |
12:12.06 | wonderworld | cause 28? |
12:12.09 | *** join/#asterisk AllstateComputer (n=brian@c-76-108-186-218.hsd1.fl.comcast.net) |
12:12.41 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
12:12.59 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
12:13.50 | tm1985 | http://pastebin.com/m225855ca yes again cause 28 |
12:15.42 | wonderworld | did you try a PRI debug already? |
12:16.28 | *** part/#asterisk mwalling (i=mwalling@97.107.128.165) |
12:16.29 | wonderworld | to see what is going on exactly? |
12:17.04 | wonderworld | is it a digium card on ptp-ISDN ? |
12:17.06 | tm1985 | yes set that in the pastebin |
12:17.26 | tm1985 | <PROTECTED> |
12:17.35 | tm1985 | and use zaphfc |
12:17.44 | wonderworld | k. you might want to try misdn instead of dahdi |
12:18.27 | wonderworld | dahdi sucks with european telcos and isdn. many features arent't properly included for the european market |
12:18.39 | wonderworld | or play around with chan_dahdi.conf |
12:18.48 | wonderworld | i think thats really all i can suggest |
12:19.50 | tm1985 | I have tried isdn with mISDNv2 same problem |
12:19.55 | wonderworld | doh |
12:20.23 | tm1985 | Somebody suggest to try it with dahdi |
12:21.27 | wonderworld | or try to get to some technician at your telco. the standard hotline won't probably know what to do |
12:21.55 | fiddur | Since normal calling works, it is obviously working... just weird for this special number... |
12:22.17 | tm1985 | I know it sounds wierd |
12:23.29 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
12:23.42 | wonderworld | or could it be that your "normal" working phones send something else than *21* when you dial *21* ? |
12:23.44 | *** part/#asterisk Grof (n=dule@89.201.165.226) |
12:24.04 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
12:24.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
12:24.04 | wonderworld | well probably not if thats the code you got from the telco... |
12:24.27 | tm1985 | I think not |
12:24.32 | fiddur | tm1985: do you have support on the isdn-card? ...you could try asking them... |
12:25.21 | tm1985 | I goin to restart my the program |
12:25.41 | ariel_ | Morning everyone |
12:26.09 | leifmadsen | morn |
12:26.42 | ariel_ | does anyone know of a way via asterisk 1.6 to resend to the phones the message that they have voicemail? |
12:26.45 | *** join/#asterisk tm1985 (n=tm@082-146-101-077.stat.adsl.xs4all.be) |
12:26.54 | wonderworld | ariel: yes |
12:26.56 | leifmadsen | ariel_: MinivmMWI |
12:27.12 | tm1985 | wonderworld I had problem with my chat program |
12:27.21 | leifmadsen | there are possibly other ways of doing it |
12:27.24 | ariel_ | minivmMWI wow, don't know this one will look it up t/y |
12:27.46 | ariel_ | I have some polycom's 8020/8030 that are passive phones |
12:28.17 | ariel_ | they get the vm icon if there on but if there turn off and back on they don't get it. until a new vm comes in. |
12:28.18 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
12:28.22 | leifmadsen | ariel_: I used it to control MWI for hot-desking agents |
12:28.36 | leifmadsen | ariel_: then they need to SUBSCRIBE to a mailbox |
12:28.38 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
12:28.42 | ariel_ | leifmadsen: t/y |
12:29.00 | leifmadsen | (SUBSCRIBE is caps because that is the SIP message, and not because I'm yelling at you like [TK]D-Fender) |
12:29.00 | tm1985 | If you have some ideas for the *21* let me know |
12:29.23 | ariel_ | I know this command |
12:29.33 | ariel_ | I fully understand been here long enough |
12:30.10 | tm1985 | Does anyone use *21* to redirect phones???? |
12:30.12 | [TK]D-Fender | leifmadsen: focus, not "yelling" |
12:30.21 | tm1985 | with asterisk and dahdi??? |
12:30.22 | leifmadsen | I don't agree |
12:30.23 | [TK]D-Fender | leifmadsen: Yelling is all-caps :) |
12:31.17 | *** join/#asterisk gsiener (n=gsiener@d-63-245-116-186.batelnet.bs) |
12:31.50 | [TK]D-Fender | leifmadsen: Half of communication is interpretation and you're certainly entitled to yours. I have however cleared the reasoning behind the intended interpretation. :) |
12:32.18 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-15-216.acanac.net) |
12:32.50 | leifmadsen | we'll agree to disagree then |
12:32.50 | ariel_ | wow there is only 4 lines of info on voip-info for mini-vm |
12:33.00 | leifmadsen | ariel_: there is lots more in doxygen |
12:33.11 | leifmadsen | http://www.asterisk.org/developers |
12:33.24 | wonderworld | who is running voip-info anyways? |
12:34.13 | wonderworld | is it associated with asterisk / digium ? |
12:34.35 | [TK]D-Fender | leifmadsen: I'll see how many disagree along with you after clarification before I openly accept "we" as a substantial percentage. |
12:34.37 | [TK]D-Fender | ;) |
12:35.05 | leifmadsen | [TK]D-Fender: I meant you and me will disagree. I wasn't speaking for anyone else. |
12:35.23 | leifmadsen | wonderworld: it is not -- it is run by a third party company. The name is at the bottom of the page. |
12:35.38 | [TK]D-Fender | leifmadsen: Yup.. I clearly haven't had enough coffee yet... |
12:35.56 | [TK]D-Fender | ~wikis |
12:35.57 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
12:35.57 | Naikrovek | I wonder if you'd ever get enough coffee in you |
12:36.19 | Naikrovek | haven't seen you cross the line from cranky to hyper yet |
12:36.19 | [TK]D-Fender | Naikrovek: Nope, still a little blood lingering in my caffeine steam :) |
12:36.25 | Naikrovek | hehe |
12:36.30 | wonderworld | that's strange. if they go down, 90% of documentation will be gone :) |
12:36.50 | Naikrovek | wonderworld: a lot of it is rubbish |
12:36.55 | leifmadsen | and 70% of out of date odcs |
12:36.56 | leifmadsen | docs* |
12:37.01 | wonderworld | yes it is....but it helped me many times |
12:37.04 | [TK]D-Fender | wonderworld: Yes, but 99% of wrong documentation ;) |
12:37.11 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
12:37.41 | fiddur | leifmadsen: but easier to find that asterisks real documentation... |
12:38.14 | fiddur | leifmadsen: search the net for asterisk application dial, and where do you end up? |
12:38.14 | leifmadsen | the doc/ directory is hard to find, along with the generated PDF in every release? |
12:38.28 | fiddur | leifmadsen: Yeps. Everyone looks at the net first :) |
12:38.33 | leifmadsen | fiddur: I just always run 'core show application <foo>' from my console |
12:38.56 | wonderworld | http://www.the-asterisk-book.com/ is ok as well |
12:39.09 | Naikrovek | who is writing that asterisk cookbook |
12:39.16 | fiddur | leifmadsen: Yes... And I know it's there too, but newbies don't... and somehow, I prefer documentation in a browser rather than in a cli... |
12:39.20 | Naikrovek | they have a bunch of recipes on a wiki that can be edited |
12:39.33 | leifmadsen | Naikrovek: no one is writing that book. It is dead. |
12:39.40 | Naikrovek | d'oh |
12:39.43 | Naikrovek | bummer |
12:40.08 | leifmadsen | I was one of the ones who was supposed to write it, but then I got busy doing consulting and making actual money :) |
12:40.24 | leifmadsen | $500 every quarter doesn't quite cut it |
12:40.41 | Naikrovek | fair enough |
12:40.45 | wonderworld | yes |
12:40.50 | fiddur | leifmadsen: I's appreciate an online version of the built in help, with the usual user-contributed comments etc... |
12:41.03 | leifmadsen | fiddur: it'll happen |
12:41.05 | wonderworld | i need to make money as well. (plus i wouldn't be able to write an asterisk book) |
12:41.05 | Naikrovek | yeah that could be a good idea if it were maintained |
12:41.30 | fiddur | automatically published for released versions off course... |
12:41.33 | fiddur | -f... |
12:41.36 | leifmadsen | Asterisk applications and functions are already in XML format. You can convert that to HTML with an interpreter/parser |
12:42.18 | fiddur | leifmadsen: Yes... and then put up a website, and let users comment it... probably easy to do, but who'll do it? :) |
12:42.24 | leifmadsen | someone will |
12:42.36 | Naikrovek | eventually |
12:43.05 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
12:43.14 | Skeeter- | who wants some money |
12:43.15 | fiddur | leifmadsen: But don't get me wrong. It is a good documentation, and the doxygen code-docs are really good too (at least from the files actually used it :D ) |
12:43.29 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
12:44.13 | wonderworld | isn't there a web-version of the doxygen docs? |
12:44.21 | leifmadsen | http://www.asterisk.org/doxygen/trunk/ |
12:44.23 | fiddur | wonderworld: yes, they are on the web :) |
12:44.45 | fiddur | and, they are available for different versions, as the user-docs would be then... |
12:44.49 | Skeeter- | pm me if you are interested |
12:44.51 | ariel_ | mini-vm seems that it might work, but I can't switch the system to that format just yet. Is there any other scripts that will resend the vm notifications in the current vm app? |
12:45.16 | leifmadsen | ariel_: you can use both at the same time |
12:45.30 | leifmadsen | I'm just using MinivmMWI for MWI, and nothing else |
12:45.40 | ariel_ | any sample |
12:45.46 | leifmadsen | sorry, nothing that I can give out right now |
12:45.53 | leifmadsen | article in the future, but not for a while |
12:46.04 | *** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk) |
12:46.27 | ariel_ | ok is there a direct command that will send the notice to all that have vm? like I set it up via a cron job? |
12:47.28 | [TK]D-Fender | ariel_: I'd go read the instructions for the app leifmadsen just handed you.... |
12:49.17 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
12:49.20 | wonderworld | i am having a problem with transfering calls. i assigned *1 and *2 in features.conf to enable people to transfer calls. it works, but not in a reliable way. every 5th try or so fails. they hear "transfer" they enter the extension but the call is never transfered. any idea how to debug that? |
12:50.00 | wonderworld | people are using SNOM 300 SIP phones |
12:50.44 | ariel_ | [TK]D-Fender: t/y doing that already. Just trying to skip heaving reading right now. |
12:50.49 | [TK]D-Fender | wonderworld: Don't Snom's have a *real* transfer feature? |
12:51.16 | [TK]D-Fender | ariel_: "module unload lazaybastard.so" :p |
12:51.26 | wonderworld | yes i think they have. as i never used hardware-phones before i just did it in the way i always did. |
12:51.31 | [TK]D-Fender | ariel_: Jump at those freebies! |
12:51.31 | wonderworld | maybe i should look into that |
12:51.44 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
12:51.51 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
12:51.51 | ariel_ | [TK]D-Fender: yes sir, do that every day if I can.....;0 |
12:51.58 | [TK]D-Fender | wonderworld: DTMF call-features = suck |
12:52.06 | *** join/#asterisk manxpower (n=EWieling@24.42.221.26) |
12:52.17 | manxpower | ~answers |
12:52.18 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
12:52.23 | tm1985 | Is there someone here that have worked with dial to *21* in dahdi and asterisk? |
12:52.29 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
12:53.11 | manxpower | We are supposed to know what *21* is? |
12:53.31 | Naikrovek | manxpower: you missed that whole conversation |
12:53.39 | voipmonk | looks into one of his Dragon Balls for an answer |
12:53.40 | [TK]D-Fender | tm1985: Thogh I've never tried that specific combo I see no reason for it to work any differently than anything else |
12:53.41 | wonderworld | [TK]D-Fender: do you think playing with inband/outofband transmission of DTMF could improve things? |
12:53.54 | wonderworld | i think i saw such a feature in the phone docs.... |
12:53.59 | [TK]D-Fender | wonderworld: No way should a SIP hardphone ever be doing inband... |
12:54.11 | wonderworld | ok, i'll check what mine do |
12:54.25 | manxpower | all SIP phones support out of band DTMF |
12:54.31 | wonderworld | probably thats it |
12:54.40 | wonderworld | didn't configure them at all |
12:54.42 | tm1985 | So why doesn't it work then |
12:54.43 | manxpower | You can only send inband DTMF over ulaw/alaw so it's pretty pointless over any other codec |
12:54.53 | voipmonk | but its fun to see a bunch of 22222's when you only press one 2 , isnt it? :) |
12:55.16 | voipmonk | or whistle dtmf |
12:55.23 | manxpower | Well, you can SEND inband DTMF over any codec. It just won't come out the other end as DTMF. |
12:55.43 | [TK]D-Fender | tm1985: Your telco is telling you *21* is not valid. why don't you ask them instead? |
12:56.21 | Naikrovek | manxpower: how about g729? DMTF work over g729? |
12:56.29 | [TK]D-Fender | < Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ] |
12:56.31 | wonderworld | i think there was a movie where a guy was able to whistle dtmf. funny scene |
12:56.44 | tm1985 | the wierd thing is that when I use a phone that is using asterisk I can call *21* |
12:56.46 | manxpower | Naikrovek: Of course DTMF works over G729, but only OUT OF BAND DTMF, not inband dtmf |
12:57.16 | [TK]D-Fender | tm1985: this is a PRI, not an analog line. Where do you get the idea that such a number exists in PRI? |
12:57.33 | Naikrovek | in-band = same means of transmission as voice? |
12:57.39 | [TK]D-Fender | tm1985: PRI's don't do analog feature codes. |
12:57.40 | manxpower | Naikrovek: yes |
12:57.42 | Naikrovek | becomes part of the audio stream |
12:57.43 | Naikrovek | okay |
12:58.02 | tm1985 | What does it then |
12:58.03 | manxpower | and since the G729 codec is designed to compress voice VERY well and DTMF is not voice, it will garble voice. |
12:58.08 | wonderworld | captain crunch style |
12:58.09 | Naikrovek | out of band = sip signalling or some other non-compressed method |
12:58.15 | Naikrovek | ? |
12:58.16 | tm1985 | those analog features codes? |
12:58.45 | [TK]D-Fender | tm1985: Sure looks like... |
12:59.15 | manxpower | tm1985: what SPECIFIC feature is *21* supposed to activate? |
12:59.17 | [TK]D-Fender | manxpower: You mean garble DTMF :) |
12:59.26 | wonderworld | tm1985: what kind of pbx are you using for the phones it is working with? |
12:59.29 | manxpower | [TK]D-Fender: that too |
12:59.39 | tm1985 | Our provider offers that for call redirection!! |
12:59.48 | manxpower | tm1985: not over PRI it doesn't. |
13:00.04 | tm1985 | Over what then?? |
13:00.13 | manxpower | tm1985: * codes are for analog lines |
13:00.21 | [TK]D-Fender | tm1985: You don't redirect PRI's.... |
13:00.25 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
13:00.27 | fiddur | tm1985: How is your analog phone connected, the one that *21* works with? |
13:00.44 | tm1985 | It is connected with isdn line |
13:00.53 | [TK]D-Fender | tm1985: go call your telco. Now. |
13:00.56 | manxpower | I see today is typical. |
13:01.02 | manxpower | Have fun, [TK]D-Fender! |
13:01.04 | *** join/#asterisk tjz (n=tjz@220.255.158.226) |
13:01.04 | *** part/#asterisk manxpower (n=EWieling@24.42.221.26) |
13:01.12 | fiddur | tm1985: isdn bri then, not pri? |
13:02.37 | wonderworld | yeah it's a bit strange that the lib is called libPRI, as you need it for BRI as well. should be called libISDN or so |
13:02.57 | wonderworld | tm1985: how many phone lines do you have with your telco? |
13:03.00 | phix | hey |
13:03.02 | tm1985 | whe have a ISDN line so what have to use than |
13:03.35 | fiddur | tm1985: So, now you know what to ask your provider then... |
13:03.55 | phix | Which one should I get out of these? --> Polycom Soundpoint IP 501, POLYCOM SOUNDPOINT IP 301, Linksys SPA 942, Linksys SPA 941 |
13:04.39 | fiddur | phix: My experience of linksys-phones is quite bad :P ...and most people here tends to recommend polycom, although I really like the snom's... |
13:04.57 | Katty | stretches |
13:04.58 | leifmadsen | phix: I like the polycoms |
13:05.04 | leifmadsen | pets Katty |
13:05.19 | *** join/#asterisk b1u3m3th (n=b1u3m3th@office.gtek.biz) |
13:05.26 | Katty | yawns, goes in search of caffeine. |
13:05.35 | leifmadsen | wonderworld: it was called libPRI before it had BRI support :) |
13:05.40 | leifmadsen | BRI support is relatively new |
13:05.45 | wonderworld | hehe |
13:05.48 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
13:05.52 | leifmadsen | Katty: I think I will make a double espresso |
13:06.24 | Katty | leifmadsen: make one for me with ice, lowfat milk, whipped cream, 2 T of pumpkin puree, and pumpkin pie spice on top. |
13:06.35 | leifmadsen | wow... that's crazy |
13:06.36 | kaldemar | phix: IP301 is discontinued, consider 330 or 320 instead |
13:06.36 | Katty | oh! and cinnamon |
13:06.38 | leifmadsen | I just drink mine black :) |
13:06.50 | Katty | pumpkin pie latte is amazing. |
13:07.20 | wonderworld | i drink 60% milk 40% coffee. nothing else. |
13:07.32 | wonderworld | full fat milk of course :) |
13:07.38 | Naikrovek | i also recommend polycoms |
13:07.40 | Katty | leifmadsen: http://www.thismamacooks.com/WindowsLiveWriter/j0438740.jpg |
13:07.45 | Katty | leifmadsen: visual reference. |
13:07.46 | Naikrovek | though i have used snom and they're nice too |
13:07.58 | leifmadsen | Katty: delish :) |
13:08.08 | leifmadsen | Katty: I don't really like pumpkin pie |
13:08.17 | leifmadsen | which is too bad since lots of people seem to |
13:08.26 | Katty | phix: and fyi, a 330 and 320 are basically the same, except the 330 has an extra network port in the back, if you need to 'line out' to another network device like a laptop. |
13:08.31 | [TK]D-Fender | phix: forget the 30X/50X unless you have a killer deal, and then only conisder the 50X |
13:08.33 | phix | fiddur: I can't get snoms here |
13:08.57 | Katty | leifmadsen: that's okay. |
13:09.00 | Katty | leifmadsen: more for me ;) |
13:09.02 | leifmadsen | :D |
13:09.04 | phix | [TK]D-Fender: 501 == 110AU atm |
13:09.09 | Katty | they make mocha lattes |
13:09.15 | Katty | white vanilla lattes. |
13:09.17 | leifmadsen | goes to make this caffienated beverage |
13:09.20 | wonderworld | phix: ehy not? |
13:09.23 | wonderworld | why |
13:09.32 | leifmadsen | get the 550! |
13:09.43 | [TK]D-Fender | phix: how does that compare to the other models? |
13:09.46 | Katty | get the 650 ;P |
13:10.01 | Katty | pats her poor little 501 |
13:10.15 | leifmadsen | Katty: ya, I have a 501 too |
13:10.16 | Katty | i am the keeper of relics. there are still some 500s in this building. |
13:10.18 | leifmadsen | it doesn't have G.722 :( |
13:10.22 | Katty | :< |
13:10.24 | leifmadsen | I have a 7960! |
13:10.29 | Katty | caffeinated beverage |
13:10.30 | Katty | shoo |
13:10.30 | phix | [TK]D-Fender: 41 AUD == SPA941, atm, (ebay so it could increase) |
13:10.39 | Katty | shoos leifmadsen off irc. |
13:10.44 | Naikrovek | i want to obtain a cisco phone to see how they're configured, but i love my polycom 320 |
13:10.58 | Katty | Naikrovek: nice word. obtain. |
13:11.10 | Naikrovek | hehe |
13:11.30 | [TK]D-Fender | phix: 941 is an OK phone... |
13:11.35 | Katty | how does chex get soggy so quickly? |
13:11.39 | Katty | this should be against the law. |
13:11.45 | Naikrovek | Katty: lol |
13:11.54 | Naikrovek | you have lowfat milk or something |
13:12.20 | Katty | don't care for whole milk. feels like i'm drinkin a shake or somethin |
13:12.26 | Naikrovek | 2% all the way |
13:12.30 | Naikrovek | yeah no whole here either |
13:12.31 | jaytee | mornin Katty |
13:12.35 | Katty | morning jaytee (= |
13:12.37 | Katty | hugs jaytee |
13:12.38 | Naikrovek | 1% also tolerable |
13:12.42 | Katty | fat free is awful. |
13:12.44 | jaytee | hugs Katty |
13:12.44 | Katty | can't drink it. |
13:12.47 | Naikrovek | same |
13:12.53 | Katty | same with cream cheese, and regular cheese. |
13:12.56 | Katty | it's just... eww. |
13:13.01 | wonderworld | you can buy fat-free milk? |
13:13.04 | [TK]D-Fender | phix: I would not say that the 501 is worth double the 941... |
13:13.12 | Katty | you can buy fat free anything i think. |
13:13.16 | phix | snom300 |
13:13.16 | [TK]D-Fender | phix: for average use |
13:13.19 | wonderworld | not here |
13:13.20 | [TK]D-Fender | phix: EW! |
13:13.25 | phix | [TK]D-Fender: that is bad? |
13:13.25 | Katty | wonderworld: where's 'here'? |
13:13.28 | wonderworld | wow...thats strange. fat free milk |
13:13.31 | [TK]D-Fender | Snom 300 = puny wastre |
13:13.34 | [TK]D-Fender | waste* |
13:13.38 | wonderworld | germany |
13:13.42 | Katty | ah. right. |
13:13.50 | Naikrovek | phix: i swear to you the polycom 321/331 is superior |
13:13.54 | wonderworld | probably it's available but i never saw it |
13:14.14 | Naikrovek | phix: and cheaper |
13:14.32 | phix | wht about snom m3? |
13:14.47 | Katty | wonderworld: we also have Fat Free sour cream, cream cheese, and yogurts. |
13:15.02 | casnik | really basic question and ya'll will probably just tell me to go read the TfoT (again) but ..... I am trying to just get a softphone to connect to asterisk and register to Asterisk. What all should I pay attention to in configs? |
13:15.06 | Katty | wonderworld: but then people think they can eat more of it, cause it's much lower in calories. |
13:15.11 | phix | Naikrovek: hmmm |
13:15.14 | Katty | wonderworld: and it doesn't really work that way. |
13:15.24 | phix | Naikrovek: where can I buy though from in AU? :) |
13:15.33 | Naikrovek | phix: good question |
13:15.40 | Katty | wonderworld: but instead it just gets them used to eating a larger portion |
13:15.44 | phix | 16.50 for snom300 :) |
13:15.49 | casnik | not worried about external calls or anything because none of that is set up |
13:15.54 | Naikrovek | phix: let me do some poking around and i'll see |
13:16.02 | Katty | wonderworld: do you have any german recipes? (= |
13:16.07 | phix | snom 320 $31 |
13:16.11 | phix | Naikrovek: :D |
13:16.16 | wonderworld | try http://www.chefkoch.de |
13:16.22 | wonderworld | largest german cooking-site |
13:16.31 | wonderworld | i suck in cooking |
13:16.33 | wonderworld | ;) |
13:16.33 | Katty | oh i don't want a german cooking site. |
13:16.36 | Katty | i want family recipes! |
13:16.51 | Katty | besides, i only speak english :/ |
13:16.54 | Katty | translating is difficult. |
13:17.03 | Katty | which is my own fault. |
13:17.28 | [TK]D-Fender | ~101 |
13:17.29 | infobot | it has been said that 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
13:17.35 | [TK]D-Fender | tm1985: ^^^ |
13:17.43 | [TK]D-Fender | tm1985: and go Google it. |
13:18.00 | Katty | there's also a movie on the history channel about telephony |
13:18.06 | Katty | which is kinda neat to watch. |
13:18.11 | [TK]D-Fender | tm1985: BRI is "residential" grade telephony and PRI is meant for larger deployments with multiple DID's, etc |
13:19.30 | Katty | i never knew a bri was considered residential grade. |
13:19.36 | Katty | does that mean it's available for residential? |
13:19.47 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
13:19.51 | Naikrovek | phix: omg shipping to australia is insane |
13:20.22 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:20.26 | Naikrovek | i used to live in sydney, plugging my former glebe address into here is making my mind overheat |
13:20.35 | coppice | the majority of BRI deployment is for small businesses |
13:20.42 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
13:21.43 | Katty | coppice: :< |
13:22.03 | Katty | boo |
13:22.08 | wonderworld | what is the price for a T1 in the US? |
13:22.18 | Katty | i think we pay around 300ish |
13:22.24 | Katty | but i could be wrong. i don't look at the bills. |
13:23.35 | wonderworld | any flat rate included or just for the line? |
13:24.19 | [TK]D-Fender | wonderworld: Depends where and with whom |
13:24.39 | [TK]D-Fender | wonderworld: watch that jump well over $1000 in a lot of places |
13:24.45 | wonderworld | ok, because 300 seems to be pretty expensive |
13:25.09 | [TK]D-Fender | wonderworld: $1000 should seem astronomical then :) |
13:25.14 | [TK]D-Fender | wonderworld: And compared to what? |
13:25.14 | wonderworld | yes it is |
13:25.20 | wonderworld | we pay EUR 99 / month |
13:25.30 | [TK]D-Fender | wonderworld: For a full 30 channel PRI? |
13:25.34 | wonderworld | yes |
13:25.38 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
13:25.43 | [TK]D-Fender | wonderworld: What other charges? |
13:26.03 | wonderworld | the calls are charged seperately |
13:26.08 | wonderworld | 99 is for the line |
13:26.39 | coppice | if you have a call centre (i.e. nothing but incoming calls) is it really 99 per month? |
13:27.18 | wonderworld | yes i think so. i was planing on getting a line in my house to host some dial-in services to make a little extra cash |
13:27.59 | wonderworld | maybe they have a minimum call limit....never looked into that |
13:29.12 | voipmonk | minimum amount of simultaneous calls... |
13:29.18 | voipmonk | im sure |
13:29.40 | voipmonk | you wouldnt go on tv american idol style and ask a few million viewers to dial your did |
13:30.05 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
13:31.12 | wonderworld | well but 99 EUROS just for providing a hardly used line mustn't be such a bad business for the telco... |
13:31.16 | *** join/#asterisk naif (n=naif@93-35-49-25.ip53.fastwebnet.it) |
13:31.20 | *** join/#asterisk oej (n=olle@132.177.253.250) |
13:31.27 | naif | hi all |
13:32.00 | ramindia | any one here success of Audiocodec MP-118 single box with multigateway asterisk register |
13:34.01 | retentiveboy | I'm getting "acl.c:376 ast_get_ip_or_srv: Unable to lookup 'dynamic'" in my logs after starting to use users.conf to setup SIP stations. I've got "users=dynamic" in there so the phones can register along with hassip=yes, registersip=yes, and type=peer among others. Is there some other combination that would fix this? |
13:35.56 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
13:35.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
13:37.44 | naif | Hi, i have my uplink carrier that sometimes make me some nasty joke, like giving me an "Answer" on SIP but then i get in the audio flow (while paying) a never ending ringing. There is some easy method to detect it? I read about callprogress=yes and busydetect=yes but is for PSTN lines (http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf). I need to detect this "fake ringing" after answering in a SIP call. Any idea? |
13:38.06 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
13:38.28 | [TK]D-Fender | naif: No. You are quite screwed |
13:38.31 | Naikrovek | is pissed off. again. |
13:39.01 | naif | Damn, it's something like a fraud. Cheap traffic have this drawback |
13:39.09 | [TK]D-Fender | retentiveboy: that should be host=dynamic |
13:39.12 | naif | i pay for a call that will never be established |
13:39.27 | Katty | Naikrovek: whatsamatter |
13:40.00 | retentiveboy | [TK]D-Fender: ah, been starting at it too long. much thx |
13:40.34 | Naikrovek | this is 2009. have we not gotten far enough along to create some sort of corporate friendly downloading application, let's call it a "downloader" that can reliably resume from interrupted transfers? |
13:41.01 | Naikrovek | i love how one guy wrote bittorrent and yet all these businesses have failed to come up with something better |
13:41.11 | Naikrovek | i'm talkin' to you, microsoft |
13:41.20 | Katty | Naikrovek: ah. well. |
13:41.21 | retentiveboy | [TK]D-Fender: wait, typo on my part. I do have host=dynamic. |
13:41.25 | Naikrovek | their download manager is absolute garbage |
13:41.26 | Katty | Naikrovek: not much you can do about that. |
13:41.32 | Katty | Naikrovek: so perk up buttercup! |
13:41.49 | Naikrovek | it can't throttle, it can't schedule, it can't reliably resume |
13:41.54 | Naikrovek | bittorrent can do all of that |
13:42.00 | Naikrovek | and ONE GUY designed it |
13:42.20 | ramindia | [TK]D-Fender: hi |
13:42.29 | Naikrovek | so, rant is over |
13:42.36 | Naikrovek | happy face |
13:42.51 | Katty | yay! |
13:42.54 | [TK]D-Fender | retentiveboy: that's what I told you... |
13:43.12 | Naikrovek | puts the headphones on, connects to 24/7 Loveline stream, and mellows out. |
13:43.57 | *** join/#asterisk oej (n=olle@132.177.253.250) |
13:44.01 | retentiveboy | [TK]D-Fender: yeah, I mistyped my question. I have host=dynamic, hassip=yes, registersip=yes and type=peer in there and am getting that error. Sorry for the confusion. |
13:44.21 | Naikrovek | i need to integrate cacti with asterisk. hrm. |
13:44.23 | Naikrovek | googles.. |
13:44.36 | [TK]D-Fender | retentiveboy: stations should not have "registersip" |
13:44.49 | *** join/#asterisk xpot (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
13:45.08 | [TK]D-Fender | retentiveboy: by the time you're done pastebin its config & "sip show peer [thepeer]" |
13:45.10 | [TK]D-Fender | ~pb |
13:45.11 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
13:45.11 | [TK]D-Fender | ^^^^^^^^ |
13:46.19 | leifmadsen | Naikrovek: there is a how-to for that. Hold on, I'll find the link I saved. |
13:46.25 | Naikrovek | ooh nice |
13:46.32 | Naikrovek | thanks, leifmadsen |
13:46.37 | Katty | cacti? |
13:46.49 | Katty | pictures big spikey green plants with red leaves. |
13:46.57 | retentiveboy | [TK]D-Fender: will do if removing that doesn't fix it. Thanks. btw, I've been in the code looking for settings that various modules are looking for in users.conf. Should the users.conf sample be updated to include some of these? |
13:47.12 | Naikrovek | cacti = pseudo monitoring. i can monitor and graph bandwidth usage, cpu usage for systems, etc |
13:47.36 | Naikrovek | it doesn't alert if something is unreachable or passes a limit or whatever, just collects info and graphs it |
13:47.56 | retentiveboy | [TK]D-Fender: error's not coming up now. thx |
13:50.39 | retentiveboy | Naikrovek: tried enabling the SNMP agent in * and polling that from Cacti? |
13:50.51 | Naikrovek | howcome TI graphing calculators cost $140, when Casio graphing calcs, that are programmable and do everything a TI calc can do, cost $40 |
13:50.57 | Naikrovek | retentiveboy: yeah working on that now |
13:51.40 | retentiveboy | Naikrovek: I'm curious what you can monitor from there... |
13:51.47 | Naikrovek | we'll see |
13:51.53 | coppice | Naikrovek: schools require TI calculators |
13:52.00 | [TK]D-Fender | Naikrovek: Because TI is in collusion with major schools to force students to buy them for their classes |
13:52.03 | leifmadsen | Naikrovek: well, apparently I didn't bookmark it, and I can't seem to find it |
13:52.13 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
13:52.17 | Naikrovek | leifmadsen: i found this: http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cacti-for-pretty-graphs/ |
13:52.22 | [TK]D-Fender | Naikrovek: ca$h c0w |
13:52.28 | leifmadsen | Naikrovek: ya, there was a newer one actually that I saw the other day |
13:52.51 | Naikrovek | [TK]D-Fender: i guess. the components that the calc is made of cost maybe $35 total |
13:53.02 | mutante | could you help me with this: I have Asterisk configured as client to a SIPgate.de as provider, i can make calls to external phones, just i cant "Playback" a message. this is how i try right now http://pastebin.ca/1568311 |
13:53.58 | casnik | so if I set 2 [Xlite] phones in sip.conf , does that mean they "register" with the entensions I put in there for them? |
13:54.16 | ramindia | [TK]D-Fender: what is the best tool to use.. to identify voice breaks..choppy voice..how can i identify this.. |
13:54.20 | casnik | are they suppose to just be able to call each other |
13:54.20 | retentiveboy | Naikrovek: doc/asterisk-mib.txt |
13:54.29 | Naikrovek | retentiveboy: thank you |
13:55.06 | coppice | seems a strange world where a programmable calculator runs Linux |
13:55.40 | retentiveboy | Naikrovek: I build my * machines with the SNMP agent enabled hoping to get time to hook them up to Cacti. Looks like I'm going to have to get busy :) |
13:56.00 | Naikrovek | retentiveboy: how many * boxes do you have |
13:56.07 | [TK]D-Fender | Naikrovek: Rackets... not just for tennis anymore ;) |
13:56.12 | retentiveboy | Naikrovek: 6 |
13:56.15 | Naikrovek | [TK]D-Fender: no kidding |
13:56.18 | casnik | I have two instances of Xlite running on two desktops , they are getting the congratulations auto greeting from asterisk ... but I am not getting them to call each other >.> |
13:56.28 | [TK]D-Fender | ramindia: Don't know, and please avoid targeting individuals for questions like this |
13:56.47 | Naikrovek | leifmadsen: this one? http://www.voipphreak.ca/2008/10/28/asterisk-snmp-with-cacti-howto-upgraded-for-asterisk-16-and-ubuntu/ |
13:56.54 | ramindia | [TK]D-Fender: got you |
13:56.55 | leifmadsen | Naikrovek: ah, that might have been it! |
13:56.56 | leifmadsen | :) |
13:56.59 | [TK]D-Fender | casnik: Good odds they are fighting for your SIP port on that machine. |
13:57.19 | [TK]D-Fender | casnik: You'll have to run them on separate ports, and configure it to match in their peers |
13:57.51 | casnik | [TK]D-Fender, ok I'll try to figure out how to do that next then |
13:57.53 | casnik | ty |
13:58.02 | wonderworld | mutante: Your announcemen is never played, because "Dial" ends, when the call is finished and one site has hung up |
13:58.29 | wonderworld | mutante: you are looking for the "A"-option of the Dial() command |
13:58.29 | casnik | [TK]D-Fender, in the sip.conf right? |
13:58.55 | Naikrovek | i can't believe how stupidly complex this is. i so hate linux. as linux things go, this isn't complex at all, as real software things go, this is unacceptable |
13:59.00 | [TK]D-Fender | casnik: yes |
13:59.22 | casnik | [TK]D-Fender, cool |
14:00.39 | *** join/#asterisk wathek (n=wathek@41.224.194.132) |
14:01.45 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-149-104.dsl.sfldmi.sbcglobal.net) |
14:02.28 | casnik | [TK]D-Fender, not seeing it where I set up locally connected SIP devices ... no mention of port ... amidoinitrite? |
14:03.12 | [TK]D-Fender | casnik: its up to you to put it in there |
14:03.19 | MarcWeber | Can I make Skype calls from asterisk? |
14:03.27 | [TK]D-Fender | ~skypeforasterisk |
14:03.28 | infobot | rumour has it, skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
14:03.31 | [TK]D-Fender | ^^^^ |
14:03.32 | casnik | so just like port=12000 |
14:03.48 | [TK]D-Fender | casnik: I'd recommend one on 5060, the other on 5061 for instance |
14:04.24 | casnik | really .... so I can just count up from there |
14:05.07 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
14:06.08 | casnik | bah , still gave me call failed , ((( chan_sip.c:14721 handle_request_invite: Call from '' to extension '1001' rejected because extension not found. )) |
14:06.36 | [TK]D-Fender | casnik: that is a DIALPLAN error |
14:06.41 | casnik | yeah |
14:06.43 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
14:06.44 | [TK]D-Fender | casnik: extensions.conf <- |
14:06.50 | casnik | time to do all that |
14:06.53 | MarcWeber | [TK]D-Fender: Thank you! |
14:06.55 | *** join/#asterisk moy (n=moy@mail.e-contact.cl) |
14:07.02 | casnik | at least I read that chapter lol |
14:07.07 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
14:07.20 | [TK]D-Fender | casnik: You'd better master the dialplan, because that is 95% of Asterisk |
14:07.27 | casnik | yeah |
14:07.37 | Katty | decides on stroganoff for lunch. |
14:07.45 | casnik | just wanted to get to where I could connect a phone ... then was gonna go to that |
14:10.15 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
14:12.15 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
14:12.48 | Naikrovek | needs to master the dialplan |
14:14.43 | casnik | needs to L2 dialplan. |
14:16.35 | mutante | wonderworld: thank you, i saw the options to the Dial command in the console, but i have no idea how to write it in a call file |
14:17.50 | wcselby | bmoraca - i was able to successfully get two softphones (on two separate computers) to register to my asterisk server last night, through my 2wire |
14:18.14 | wcselby | bmoraca - I use at&t u-verse internet |
14:21.15 | Naikrovek | wcselby: nice |
14:21.41 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
14:21.43 | wcselby | Naikrovek - he asked me yesterday if I could get two phones connected through a 2wire, I told him I'd check. So I did.... :) |
14:22.40 | mutante | so if i have "exten => _X.,2,Dial(SIP/${EXTEN}@sipgate-out,30,trg)" now, where would i put the "A" option to Dial() |
14:23.11 | wcselby | mutante - with the 'trg' part |
14:23.23 | wcselby | mutante - so it would become ,trgA) |
14:23.35 | wcselby | but I don't think that's the proper spot for options in the Dial() command, let me check |
14:23.38 | mutante | wcselby: ahaa, and the name of the sound to play? |
14:24.32 | wcselby | mutante - you would enter ,Dial(SIP/${EXTEN}@sipgate-out,30,trgA(nameoffiletoplay)) |
14:24.33 | wonderworld | exten => _X.,2,Dial(SIP/${EXTEN}@sipgate-out,30,trgA(mysoundfile)) |
14:24.42 | mutante | thank you :) |
14:25.12 | wonderworld | "mysoundfile" mustn't have a file extension |
14:25.21 | wonderworld | so it's not mysoundfile.wav |
14:25.25 | mutante | yep, i learned that yesterday :) ...trying |
14:25.27 | mutante | i have .gsm files |
14:25.27 | wonderworld | just mysoundfile |
14:26.31 | mutante | arr, nope, still hangs up..:( this would have been to easy...cru |
14:26.56 | wonderworld | check the CLI to see what is happening.... |
14:27.23 | wcselby | did you do a dialplan reload after you made the change to the file? |
14:27.25 | wonderworld | the A option works for sure, i am using it a lot |
14:27.34 | mutante | <PROTECTED> |
14:27.36 | wcselby | i forget that every now and then when doing lots of small changes |
14:27.56 | mutante | wcselby: i did a /etc/init.d/asterisk restart |
14:28.00 | wonderworld | there seems to be an extension missing.... |
14:28.16 | wonderworld | or did you remove it? |
14:28.32 | mutante | ? |
14:28.41 | wonderworld | SIP/..... did you remove the number? |
14:28.44 | mutante | yes |
14:28.47 | wonderworld | ahh ok |
14:29.03 | wcselby | mutante - paste the whole output from the cli for the call to a pastebin |
14:29.04 | mutante | the phone on my desk rings |
14:29.11 | mutante | it just doesnt play the message after pickup |
14:31.05 | wonderworld | increase the verbosity on the cli and post the full call output to a pastebin |
14:31.52 | *** join/#asterisk xrmx__ (n=rm@host103-251-dynamic.15-87-r.retail.telecomitalia.it) |
14:32.40 | *** join/#asterisk csmyth (n=csmyth@ext-52.sagetelecom.net) |
14:34.50 | mutante | http://pastebin.ca/1568366 |
14:34.53 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
14:35.09 | *** join/#asterisk user4545 (n=sipip@p57B1F38A.dip.t-dialin.net) |
14:36.45 | [TK]D-Fender | mutante: [Sep 16 16:28:05] WARNING[1625]: pbx.c:3080 pbx_extension_helper: No application 'SetCallerId,SIPID' for extension (sipout, 10, 1) |
14:37.09 | mutante | so if i dont set a caller id i can still call, but not play sounds? |
14:37.27 | [TK]D-Fender | mutante: I don't see you showing us your configs or the call file... |
14:37.30 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
14:37.58 | user4545 | Hi, my dialplan MENU work nicht...I call one number in Asterisk from outside and i cann't go to next menu, Asterisk don't understand my additions digits |
14:38.12 | user4545 | anybody can help me? |
14:38.21 | Naikrovek | settle down |
14:38.29 | Naikrovek | don't expect a quick response |
14:38.35 | Naikrovek | just wait and see who answers |
14:38.47 | user4545 | <PROTECTED> |
14:38.47 | user4545 | [Sep 16 16:33:52] WARNING[17073]: pbx.c:5656 pbx_builtin_waitexten: Timeout but no rule 't' in context 'von-voip-provider' |
14:38.47 | user4545 | <PROTECTED> |
14:39.04 | wcselby | user4545 - pastebin your extensions.conf file (or at least the relevant parts) and give us a link |
14:39.13 | wcselby | ~pb |
14:39.14 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
14:39.14 | user4545 | one moment |
14:39.15 | *** join/#asterisk kondela (i=kondela@116.68.103.250) |
14:39.22 | [TK]D-Fender | user4545: You took too long to respond and * terminated the call because you didn't have a "t" exten to handle the fact they took too long |
14:39.47 | Naikrovek | the DMTF could have been garbled or lost via the voice encoding as well |
14:39.59 | wcselby | could also be a digit timeout issue |
14:40.01 | *** join/#asterisk wopsy (n=80475@AToulouse-754-1-13-64.w90-55.abo.wanadoo.fr) |
14:40.11 | wcselby | if he's trying to dial a two or three digit number |
14:40.32 | Naikrovek | s/DMTF/DTMF/ |
14:40.47 | user4545 | please http://pastebin.com/m74b7e501 |
14:40.52 | kondela | hi i do have a queue related question |
14:40.57 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
14:41.26 | user4545 | how van I DMTF ON-make ? |
14:41.27 | mutante | [TK]D-Fender: http://pastebin.ca/1568373 |
14:41.30 | user4545 | howcan |
14:41.53 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
14:42.43 | [TK]D-Fender | mutante: "core show function CALLERID" <- and go fix your dialplan |
14:42.43 | wcselby | user4545 - are you using a compressed codec? |
14:43.00 | [TK]D-Fender | mutante: it is dying on that illegal app name |
14:43.04 | user4545 | wcselby: no I have just asterisk instaled |
14:43.05 | kondela | if i use agentlogin(), i cant see the callerid of the caller, because the agent logged in the queue.. |
14:44.21 | mutante | [TK]D-Fender: well ok, how come it still dials though...hmmm |
14:44.28 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
14:45.18 | kondela | hi.. can anyone share with my thoughts..? |
14:45.34 | wonderworld | mutante.... please post the output of a call.... before that: "core set verbose 100" "sip set debug off" |
14:45.52 | wcselby | mutante - exten => _X.,1,SetCallerId,SIPID - should be exten => _X.,1,Set(CALLERID=something) |
14:46.19 | mutante | ok, grr, that line is the one i got as example from the SIP provider page itself :P |
14:47.03 | wonderworld | yes, you have to replace "SIPID" with your actual SIP-ID. the one you got from your provider |
14:47.06 | wcselby | mutante - I think maybe they were giving you guidelines or something |
14:47.20 | mutante | i see, thanks |
14:47.21 | wcselby | as opposed to something you copied and paste |
14:48.27 | wonderworld | mutante: can you talk on the phone when you are called? |
14:48.35 | wonderworld | or is the call directly hung up? |
14:48.41 | *** join/#asterisk kerchunk (n=kerchunk@pool-173-49-10-152.phlapa.fios.verizon.net) |
14:48.51 | mutante | it rings and after pickups is directly hung up |
14:48.52 | garymc | anyone in here got a spare minute to test my sip extension? |
14:49.12 | garymc | it was working earlier, but some guy i test with now doesnt hear nothing. Unless its his sound card? |
14:49.58 | mutante | wonderworld: http://pastebin.ca/1568382 |
14:49.59 | wcselby | garymc - i can make a call to you if that's what you're asking? |
14:50.06 | wcselby | PM me with a number to call |
14:50.09 | *** join/#asterisk heit0050 (n=heit0050@mail2.heitkeconsulting.com) |
14:50.36 | kondela | i do have a general regarding queue behaviour in asterisk.. can someone assist me.. |
14:50.45 | *** join/#asterisk Tim_Toady (n=moi@adsl194-8.kln.forthnet.gr) |
14:52.13 | wonderworld | mutante: the call file is calling you at 0211-something and another side at sipgate-out/10 |
14:52.27 | wonderworld | are you sure sipgate-out/10 is a valid number with your provider? |
14:53.14 | mutante | no, i am not, can i just remove the "Extension: 10" from the call file |
14:53.47 | wonderworld | what do you want to do with your call-file? |
14:53.55 | wonderworld | what triggers it? |
14:54.03 | mutante | i want to call 0211-something and play a message |
14:54.28 | kondela | hello world.. how can i display a caller id , while the agent is logged in using agentlogin() |
14:54.29 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
14:54.32 | wonderworld | ok |
14:54.37 | mutante | after this works we will probably make nagios move a callfile to "outgoing" in the case of an alarm |
14:55.55 | wonderworld | ok, your extensions.conf is not correct for that setting |
14:56.05 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
14:56.07 | wonderworld | your callfile calls 0221-something, which is your desk, i asume? |
14:56.19 | mutante | right, and it does ring |
14:56.25 | wonderworld | after that, it tries to connect you to sipgate-out/10 |
14:56.44 | wonderworld | because you said in the call-file, it sould go to extension 10 after connecting with 0221-something |
14:57.25 | wonderworld | just create an extension called "10" and put a playback in there |
14:57.37 | wonderworld | you don't need to dial to any other party |
14:57.42 | mutante | aha, ok, but "At least one of app or extension must be specified" |
14:57.48 | mutante | ah, ok |
14:59.18 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
14:59.42 | wonderworld | it went into your _X. extension, becuase thats a wildcard for "any number" |
14:59.51 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
14:59.52 | wonderworld | it matches "10" as well |
15:00.02 | kondela | hello... any queue expert here..:) |
15:00.05 | wcselby | kondela - have you tried messing with Set(CALLERID) ? |
15:00.45 | kondela | wcselby.. finally.. thanks. |
15:01.07 | kondela | so you are saying i have to set a caller explicitly..? |
15:01.32 | *** join/#asterisk DigitalFlux-AFK (n=DigitalF@unaffiliated/digitalflux) |
15:01.52 | DigitalFlux-AFK | Hey everybody |
15:02.11 | DigitalFlux-AFK | I need some help regarding how queues are handled in Asterisk .. |
15:03.16 | wcselby | kondela - I'm not sure, I don't really know what's going on with your situation. |
15:03.36 | kondela | wcselby - let me put it straight |
15:04.08 | [TK]D-Fender | kondela: Why start now? I mean you've been in here 20 minutes already and never asked a real question... |
15:04.26 | kondela | well i asked.. may be you didnt see |
15:04.48 | [TK]D-Fender | kondela: Answer : you CAN'T |
15:05.15 | [TK]D-Fender | kondela: Create an external app that monitor AMI to trap the AgentConnect message |
15:05.30 | [TK]D-Fender | kondela: This will require a PC side app |
15:05.48 | *** join/#asterisk plundra (i=404@article.se) |
15:05.54 | kondela | AAHA.. |
15:06.01 | kondela | interesting |
15:06.28 | kondela | so the answer is with agentlogin(), an agent cannot see the callerid |
15:07.31 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-23-22.w90-56.abo.wanadoo.fr) |
15:07.41 | plundra | Ok, this might be a common problem... I've got a few SPA942 setup, calling out via my asterisk is fine, but when calling each other, the mediastream seems to get lost. No sound what so ever. NAT is not used anywhere and all clients have external addresses. |
15:07.51 | kondela | [TK] so the answer is with agentlogin(), an agent cannot see the callerid.. |
15:08.18 | [TK]D-Fender | [11:04]<[TK]D-Fender>kondela: Answer : you CAN'T <- was this somehow not clear? How much more clarification do require? |
15:08.32 | merlin8282 | Hi all ! How can I access data a script sets (such as environment var, or a file, or anything) to work with it in asterisk ? |
15:08.36 | kondela | [TK] though i never used agentcallbacklogin(), i think this allowed us see the callerid.. a m i right..? |
15:08.54 | *** join/#asterisk davidandgoliath (n=David@out.clearnet.com) |
15:09.14 | kondela | [TK] your previous message is clear to me |
15:09.28 | kondela | i think i mis-typed |
15:10.06 | mutante | thank you all guys.. i got it working:) |
15:10.28 | [TK]D-Fender | kondela: AgentLogin is YOU calling into an app. AgentCallbackLogin is setting * to call a local channel to ring the phone. that is a call. |
15:11.16 | [TK]D-Fender | merlin8282: "core show function ENV" , "core show function STAT" |
15:12.16 | kondela | [TK] i got it right.. my question was can the agentcallbacklogin() can present original callerid to the agent. i cant try this, since i dont have any 1.2 * withme now.. do you have any experience this part |
15:13.12 | [TK]D-Fender | kondela: its a call. callee's get CALlERID. |
15:13.21 | kondela | ok.. |
15:13.44 | merlin8282 | [TK]D-Fender: I already know these functions. The proble is, I can't set the environment variable from within my script. |
15:14.04 | [TK]D-Fender | merlin8282: Then go have it set something else. |
15:15.03 | merlin8282 | I don't understand what you mean. |
15:15.12 | kondela | [TK] is there any replacement for agentcallbacklogin() in 1.6/1.4 release.. or an other method to implement the agentcallbacklogin() behaviour in those latest release..? |
15:16.54 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
15:17.41 | leifmadsen | kondela: the functionality is available via dialplan |
15:18.36 | garymc | ok one more, anyone else with a mic can test my asterisk pbx by logging into it with a softphone. Take 2mins of your time? |
15:18.50 | leifmadsen | kondela: http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbacklogin-to-standard-dialplan-methods-part-1/ <-- might be useful |
15:18.51 | [TK]D-Fender | kondela: http://leifmadsen.wordpress.com/tag/addqueuemember/ |
15:18.56 | leifmadsen | heh :) |
15:19.18 | kondela | alright.. alright... thats awesome response.. |
15:19.20 | Naikrovek | garymc: i don't have a softphone, nor the desire to set one up, nor the time, but otherwise i'd be happy to help heh |
15:19.34 | leifmadsen | I still have to finish that article, ugh |
15:19.36 | kondela | i am reading this now |
15:19.39 | leifmadsen | too much work! :) |
15:19.48 | jaytee | TRABAJO! |
15:20.29 | garymc | Naikrovec : Sarcasim? |
15:20.46 | [TK]D-Fender | kondela: All you really need is to seriously read the instructions for AddQueueMember and RemoveQueueMember |
15:21.04 | ruben23 | jaytee: Pinoy.... |
15:21.45 | kondela | aha.. thats what i found, googling while chat.. |
15:23.38 | garymc | anyone? |
15:23.43 | *** join/#asterisk s14ck (n=jtorres@ccscliente156.ifxnetworks.net.ve) |
15:25.29 | *** join/#asterisk rene- (n=fft@200.34.66.137) |
15:25.33 | rene- | hey guys |
15:26.05 | rene- | can i lower my t1 costs if i order my t1s to be dropped at a carrier hotel? |
15:26.35 | Qwell | rene-: maybe. call and ask |
15:28.10 | Naikrovek | garymc: not sarcasm, i'm just busy and i don't dig softphones. have your friend test his sound in some other way |
15:28.21 | Naikrovek | softphones imho are not worth the time |
15:29.47 | wonderworld | twinkle is nice |
15:29.52 | [TK]D-Fender | Naikrovek: No need to use a softphone to test for him |
15:30.10 | Naikrovek | [TK]D-Fender: i know but i'm just too tired to put any effort in anything right now |
15:30.17 | [TK]D-Fender | Naikrovek: He was over-specific for no valid reason. |
15:30.35 | Naikrovek | [TK]D-Fender: well it's a common thing, to do that. |
15:30.39 | [TK]D-Fender | Naikrovek: But not busy enough to simply ignore him ;) |
15:30.47 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-23-22.w90-56.abo.wanadoo.fr) |
15:30.49 | garymc | well you can use asip phone if you wish? |
15:31.17 | Naikrovek | [TK]D-Fender: i'm at this weird place where i'm too busy to take on more activity but too bored to leave it. |
15:31.18 | [TK]D-Fender | garymc: Is that a question? |
15:32.00 | Qwell | [TK]D-Fender: no? |
15:32.14 | garymc | you are all to clever for me |
15:32.16 | [TK]D-Fender | Qwell: :p |
15:32.29 | Naikrovek | sorry garymc, someone will test for you i'm sure, just be patient |
15:32.58 | wonderworld | garymc: i would if setting up my usb-mic wouldn't be such a pain with linux |
15:33.52 | [TK]D-Fender | Naikrovek: http://www.youtube.com/watch?v=C_Y6231uAmo <- Focus on 1:23 |
15:34.03 | [TK]D-Fender | Naikrovek: But completely worth the full view |
15:34.06 | [TK]D-Fender | (listen) |
15:34.15 | user4545 | can me anybody help with it ? http://pastebin.com/m273224f4 .... I cann'nt callthrow.. my Asterisk don't understend DTMF digits |
15:34.20 | Naikrovek | lol |
15:34.25 | Naikrovek | how long will that take |
15:37.46 | [TK]D-Fender | user4545: try "dtmfmode=inband" |
15:38.07 | user4545 | one moment |
15:38.24 | [TK]D-Fender | user4545: actually, some guides I just googled said "info" instead |
15:38.54 | Naikrovek | i wish google handled regexes |
15:39.23 | user4545 | no results |
15:39.44 | user4545 | donn't working |
15:42.53 | [TK]D-Fender | user4545: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk |
15:43.10 | [TK]D-Fender | user4545: if it doesn't work, go contact them |
15:46.28 | user4545 | it's work, but I want different number call throw DTMF |
15:46.56 | [TK]D-Fender | user4545: Please rephrase that... |
15:47.53 | wonderworld | user4545: check if you are using "rfc2833" as signalling method for DTMF in your sip-phone / softphone |
15:47.59 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
15:49.43 | wonderworld | there are different ways of transmitting DTMF. checking the sipgate example conf, they seem to be expecting "rfc2833" |
15:50.48 | *** join/#asterisk jkroon (n=jkroon@dsl-240-185-181.telkomadsl.co.za) |
15:51.42 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
15:57.39 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
16:00.38 | *** join/#asterisk xrmx__ (n=rm@host103-251-dynamic.15-87-r.retail.telecomitalia.it) |
16:01.56 | *** join/#asterisk oej (n=olle@132.177.253.250) |
16:05.21 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
16:05.55 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
16:06.53 | *** join/#asterisk user4545 (n=sipip@dslb-092-074-252-179.pools.arcor-ip.net) |
16:10.07 | *** join/#asterisk wtca (n=wtca@williamt.noc.sonic.net) |
16:10.11 | *** part/#asterisk wtca (n=wtca@williamt.noc.sonic.net) |
16:11.10 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
16:12.50 | *** join/#asterisk errotan (n=errotan@5403E6D6.catv.pool.telekom.hu) |
16:14.59 | wcselby | awesome -> http://www.voip-info.org/wiki/view/Paradise+Tent+for+Camping+and+Holiday+Needs+ |
16:16.33 | Naikrovek | hehe |
16:16.38 | wonderworld | that sucks |
16:16.48 | wcselby | there were links there |
16:16.54 | wcselby | i removed them, obviously |
16:17.27 | wonderworld | some guy or some bot posted that? |
16:17.43 | wcselby | Wed 16 of Sep, 2009 [09:17]dielonput125.60.173.1481 |
16:17.48 | Naikrovek | someone was using it to vent |
16:17.58 | wcselby | no they used it to spam |
16:18.05 | wcselby | i changed it to "No." |
16:18.22 | wcselby | the original -> http://www.voip-info.org/wiki/page_history.php?page_id=5762&preview=1 |
16:19.48 | wonderworld | his website sucks. right sidebar says: "My right sidebar goes here. " |
16:20.19 | wcselby | haha |
16:20.30 | wcselby | the left sidebar contains all the viruses he wants to install on your pc |
16:20.39 | wcselby | :P (j/k, haven't been to the site) |
16:21.26 | wcselby | what was the lawnmower one? |
16:21.28 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
16:21.29 | wcselby | ~lawnmower |
16:21.34 | wcselby | or something like that |
16:21.44 | Naikrovek | boschlawnmower? |
16:21.44 | afink | is there a general ration of seconds/ring? |
16:21.57 | Naikrovek | seconds per ring? |
16:21.59 | wcselby | http://www.voip-info.org/wiki/view/Bosch+lawnmowers |
16:22.25 | afink | like 5 = 1 ring or so |
16:22.44 | Naikrovek | oh a general ratio |
16:22.46 | Naikrovek | you said ration |
16:22.55 | Naikrovek | no idea |
16:23.02 | afink | oh whoops |
16:23.07 | Naikrovek | pretty sure they're more or less the same across the US, but i have never timed it |
16:23.11 | wcselby | i think 5 seconds to ring is about right, depends on the country you're in |
16:23.18 | Naikrovek | i was going to say 3 seconds |
16:23.23 | Naikrovek | but i have no idea if that's right |
16:23.41 | Naikrovek | in australia, their ringing sound sounds like our busy signal |
16:23.46 | Naikrovek | if i recall |
16:24.02 | Naikrovek | i remember thinking that everyone was busy |
16:24.02 | Naikrovek | but their phones were all ringing |
16:24.06 | wcselby | lol |
16:24.18 | Naikrovek | yeah so lots of people thought i was pranking them |
16:25.11 | wonderworld | i did a funny prank a few years ago. |
16:25.20 | wonderworld | connected to pizza ordering service with one another |
16:25.33 | wonderworld | both phones were ringing |
16:25.36 | Qwell | wonderworld: You're an ass. heh |
16:25.41 | Naikrovek | lol |
16:25.51 | wonderworld | it took them a minute to find out, that noone wanted to order |
16:25.55 | wonderworld | quiet funny |
16:26.08 | wcselby | speaking of phone pranks |
16:26.09 | wonderworld | not to...two of course |
16:26.20 | wcselby | did anyone read leifmadsen's blog post about telemarter torture? |
16:26.25 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
16:26.52 | wonderworld | Qwell.... i was young |
16:27.56 | wonderworld | did the same thing with a nazi and a communists party office. funny too |
16:29.40 | Qwell | wonderworld: hope you don't ever need to get a security clearance.. |
16:31.02 | *** join/#asterisk Tim_Toady (n=moi@adsl194-8.kln.forthnet.gr) |
16:31.02 | wonderworld | nah, don't want one. if they have to go thru my call history of the last 10 years for it, i want it even less.... |
16:31.59 | coppice | you can always get one at a clearance sale |
16:32.12 | wcselby | coppice................. |
16:32.15 | wcselby | that was bad |
16:33.51 | leifmadsen | wcselby: I did :) |
16:34.33 | Qwell | leifmadsen: why would you read that nubs post? |
16:34.38 | Naikrovek | hah |
16:34.49 | wcselby | haha @ leifmadsen |
16:34.50 | wcselby | :) |
16:35.17 | leifmadsen | Qwell: BURN |
16:36.23 | russellb | leifmadsen paid off the other 2 authors to get his name on the book ... |
16:36.34 | leifmadsen | that's why it's at the end of the list |
16:36.50 | Qwell | I bribed leifmadsen to get a mention in the book. |
16:36.52 | Qwell | totally did |
16:36.57 | leifmadsen | it's true |
16:37.26 | Qwell | (that stills cracks me up btw) |
16:37.53 | Naikrovek | ... how uh.. how much did that cost |
16:38.03 | Qwell | Naikrovek: about a beer |
16:38.09 | Naikrovek | hrm. |
16:39.33 | wcselby | lol, where's the mention? |
16:39.34 | Naikrovek | so a case would get maybe contributor credit then? |
16:39.37 | wcselby | now I'm curious |
16:39.48 | wcselby | Naikrovek - depends on the type of beer I would imagine |
16:39.50 | Naikrovek | we'll have to know qwell's real name |
16:39.57 | Qwell | not like it's hard to find |
16:40.42 | wcselby | just got an email from my helpdesk that our internet connection was down |
16:40.45 | wcselby | .... |
16:40.55 | *** join/#asterisk DrCarumas (n=Carumas@adslfixo-b3-127-186.telepac.pt) |
16:40.58 | DrCarumas | Hi! |
16:41.08 | Naikrovek | i love how people email me saying email is down |
16:41.13 | Naikrovek | happens nearly weekly |
16:41.40 | wonderworld | exchange server, huh? |
16:41.48 | wcselby | wonderworld - hahahahaha |
16:41.54 | wcselby | i was thinking the same thing |
16:42.05 | Naikrovek | yes, but email has never gone down without the power going down |
16:42.34 | wonderworld | and power is going down once per week? |
16:42.37 | Naikrovek | they're just dumb |
16:42.46 | wcselby | i used to get calls from people saying "my email doesn't work". I tell them to try sending me a test. I'd get it. "Well, you're email isn't down. So what's your specific problem". "Oh, I got an error trying to send so and so an email" |
16:42.49 | Naikrovek | no, power rarely goes out (once or twice in previous 6 months) |
16:43.05 | wcselby | "that doesn't mean email is down" |
16:43.15 | Naikrovek | yeah that's similar to what i get |
16:43.38 | DrCarumas | guys, i'm using a sip provider to place inbound/outbound calls. Sometimes oubound calls wont work for a while and i get this error: Got SIP response 503 "Service Unavailable gkd" back from "ipaddress" . I've google it but not much information. Do you think this could be something with my asterisk (v.1.4.24.1) or is from my voip provider? Thanks in advanced. |
16:44.00 | wcselby | then, since I'm not a BoFH, I'd help them fix their problem and everyone would be happy.......... |
16:44.02 | Naikrovek | DrCarumas: do all outbound calls fail or just some |
16:44.07 | wcselby | honest, that's what happened |
16:44.14 | DrCarumas | Naikrovek, wen this appens all fail |
16:44.35 | DrCarumas | Naikrovek, same error, then after a while everything is back ok. |
16:44.36 | Naikrovek | DrCarumas: do you check with your provider when this happens? where does the SIP 503 come from |
16:44.39 | wcselby | DrCarumas - I'd say you need a secondary, failover provider |
16:45.05 | Naikrovek | yeah this reeks of provider issues |
16:45.23 | DrCarumas | Naikrovek, i've openned a support ticket i'm still waiting but wanted your opinion |
16:45.35 | DrCarumas | wcselby, that's true i realy should |
16:45.41 | wcselby | DrCarumas - I'd say service provider issue. |
16:45.57 | Naikrovek | DrCarumas: my provider does this, well their upstream provider does this, so once in a while, phone calls to a specific area will fail to go through |
16:46.01 | DrCarumas | wcselby, ok i'll wait to see what the ticket resoltin will conclude. |
16:46.14 | wcselby | lol, someone behind me just said "uh oh, is Microsoft down too?" |
16:46.16 | Naikrovek | DrCarumas: this is almost certainly a provider issue, or their network provider |
16:46.43 | DrCarumas | Naikrovek, ok. Thanks for your help and wcselby. |
16:46.50 | wcselby | DrCarumas - np |
16:47.32 | *** join/#asterisk Carlos_PHX (n=carlos@68.108.193.174) |
16:47.51 | *** part/#asterisk Carlos_PHX (n=carlos@68.108.193.174) |
16:53.30 | *** join/#asterisk oej (n=olle@132.177.253.250) |
16:55.26 | ZenBSDi | exten => ${ZenBSDi},1,Background("I got da skillz!") |
16:55.43 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
16:56.14 | wcselby | ZenBSDi - drop the quotes |
16:56.21 | wcselby | :) |
16:56.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
16:59.49 | ZenBSDi | exten => ${wcselby},1,Gotoif($["${ZenBSDi}" = "don't care"]?hatepeoplewhokillmyfnjokes,11 |
17:00.03 | Qwell | That was a joke? |
17:00.56 | ZenBSDi | of course.. cause I got no skill :p |
17:01.04 | wcselby | ZenBSDi - lol |
17:01.28 | wcselby | hey, if it was [TK]D-Fender responding, he wouldn't have been as nice as I was :P |
17:02.01 | ZenBSDi | Yeah and I'd move him to the hangup extension too :p |
17:02.29 | ZenBSDi | Or worst.. run him through an AGI script that dices him up in /dev/null :p |
17:02.42 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
17:02.42 | [TK]D-Fender | ZenBSDi: O RLY? |
17:02.54 | wcselby | lo |
17:02.58 | wcselby | lol even |
17:03.09 | casnik | theres a storm comin! |
17:03.19 | ZenBSDi | :p |
17:06.06 | ZenBSDi | Who AGI scripts in here and who prefers perl or php for agi programming? |
17:06.26 | [TK]D-Fender | ZenBSDi: Yes |
17:06.27 | casnik | If I was at that point I would pcik perl |
17:06.34 | casnik | pick* |
17:06.39 | casnik | but I am still newb |
17:08.16 | Qwell | bah. name one thing you can do in AGI that you can't in pure dialplan |
17:08.43 | ZenBSDi | I wrote a small phpagi script to do credit card processing ... just sucks having to use the weak text to speech stuff thats out there. |
17:09.32 | ZenBSDi | Owell, Control your pbx from a web interface :p |
17:10.47 | ZenBSDi | but thats administration :p .. agi vs dialplan.. well .. I guess it comes down to dynamics. changing passwords or extensions in a database and having it agi check or agi set against those database variables maybe |
17:11.36 | ZenBSDi | I like setting up Asterisk realtime with MySQL .. so I'm odd like that |
17:12.08 | [TK]D-Fender | Qwell: Perform background tasks while playing audio outside of an IVR (non-interruptable) |
17:13.07 | ZenBSDi | heh... nice |
17:13.40 | wonderworld | i did a server-remote-control thingy with AGI/php. but just to learn agi. i think it would have been more complicated within the dialplan, but maybe not.... |
17:14.54 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
17:15.21 | wonderworld | the thing can read me my email, don't know if that would have been possible within the dialplan. there was a lot of text-parsing involved. |
17:16.31 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
17:19.27 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-71-157.link.net.pk) |
17:22.51 | p3nguin | If my phone is on the same LAN as the Asterisk server, but the provider is on the outside of NAT, should the peer be configured for nat=no or nat=yes? |
17:23.05 | p3nguin | peer being the phone |
17:24.39 | IBC_jkenney | Is there any plans for digium to allow hints from the realtime mysql engine? |
17:24.49 | IBC_jkenney | in any new releases of asterisk or asterisk addons |
17:24.49 | p3nguin | And what about for the peer context of the provider? Their config says "; nat=yes ; Uncomment this if your box is behind a NAT" |
17:25.20 | p3nguin | The Asterisk box is the gateway between the internet and the LAN, so is Asterisk behind NAT or not? |
17:27.18 | Naikrovek | p3nguin: i use nat=yes |
17:27.26 | Naikrovek | p3nguin: asterisk box is also router? |
17:27.32 | p3nguin | correct |
17:27.36 | Naikrovek | ew |
17:27.39 | Naikrovek | well if it works... |
17:28.01 | Naikrovek | depending on which interfaces * is listening on, it could be on the LAN, WAN, or both |
17:28.07 | Naikrovek | probablyboth |
17:28.15 | p3nguin | It listens on both. |
17:28.24 | Naikrovek | it straddles both then |
17:28.33 | p3nguin | This provides a public IP address for Asterisk, so I'm thinking Asterisk itself is not considered to be behind NAT. |
17:28.41 | p3nguin | But any phones would be. |
17:28.42 | Naikrovek | if asterisk is compromised and gives a rootshell, attacker has access to your internal network |
17:28.57 | Naikrovek | p3nguin: yes |
17:29.07 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
17:29.30 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:8b:771c:b2c9:8871) |
17:29.32 | cusco | hi |
17:29.38 | [TK]D-Fender | peer entries for Isps should almost always be nat=no |
17:29.38 | p3nguin | And since the phones are talking local addresses to/from Asterisk, I think local phones can also be set to nat=no. |
17:29.42 | [TK]D-Fender | ITSPs |
17:29.55 | Naikrovek | [TK]D-Fender is correct; phones should be nat=yes |
17:30.20 | Qwell | Naikrovek: They aren't behind a NAT |
17:30.29 | Qwell | relative to Asterisk.. |
17:30.38 | Naikrovek | can the phones be reached directly from the internet? |
17:30.41 | cusco | We have several warnings like "Sep 16 18:16:33] WARNING[24867] file.c: File followme/ does not exist in any format" |
17:30.57 | [TK]D-Fender | cusco: Perhaps you should make the requested file exist... |
17:31.39 | Naikrovek | or point it in the right place |
17:32.16 | user4545 | why Sipgate close DTMF |
17:32.16 | cusco | [TK]D-Fender: good point, tho we have manny followme's and can't find one that does not request the filename |
17:32.20 | user4545 | ? |
17:32.22 | cusco | any ideias on how to debug that? |
17:32.26 | p3nguin | Relative to Asterisk the phone are not NATed. Relative to the provider, the phones are behind NAT. That was my entire reason for asking which way it should be configured. |
17:32.52 | [TK]D-Fender | cusco: Well we don't see what is being requested, or confirmation that the file it's looking for should be found, etc |
17:33.05 | [TK]D-Fender | p3nguin: all as "no" |
17:33.27 | Naikrovek | all as no, really |
17:33.40 | *** join/#asterisk el_critter (n=critter@200.8.96.143) |
17:33.44 | el_critter | Hi |
17:33.48 | Naikrovek | iH |
17:33.54 | p3nguin | That was my first thought, but I wanted an expert opinion before I left it that way. |
17:36.00 | superbeef | anybody dealt with this bug? http://pastebin.ca/1568596 |
17:36.18 | superbeef | makes the system load average go to the insanity point... like 120 |
17:36.35 | superbeef | i saw a bug report for asterisk 1.6, but nothing too useful there |
17:36.56 | superbeef | this is on 1.4.26 |
17:37.00 | Naikrovek | 120, that's respectable |
17:37.14 | superbeef | lol, i respect 1.0 more |
17:37.31 | Naikrovek | i've seen some solaris boxes go up to 850 |
17:37.55 | Naikrovek | NOT barf, then come back down |
17:37.55 | superbeef | wow |
17:38.01 | Naikrovek | same reaction i had |
17:38.18 | Naikrovek | but no, i've not seen that. does the bug report give any reason as to the cause |
17:39.24 | Naikrovek | i was tech support at a library software company outside of chicago, customer called saying that their machine was unresponsive |
17:40.28 | Naikrovek | i was able to ssh in, noted the 800 850 850 load average (after much waiting) then reload the appropriate service and it dropped down to 2 |
17:40.31 | superbeef | i can't find the bug report now lol.. looking at some differnet threads |
17:40.52 | Naikrovek | lady on phone said "ooh the server just got quiet" |
17:41.04 | superbeef | haha |
17:41.28 | superbeef | here's the one that seems most like mine |
17:41.29 | superbeef | https://issues.asterisk.org/view.php?id=15900&nbn=1 |
17:41.34 | Naikrovek | yeah i kinda like tech support jobs for that reason. over time, you see everything possible |
17:41.57 | Naikrovek | superbeef: did your iax2 channel fail |
17:42.21 | superbeef | logging goes dead after it gets saturated |
17:42.23 | superbeef | so hard to tell |
17:42.42 | Naikrovek | what fixes it? asterisk restart or machine reboot |
17:42.56 | superbeef | killall -9 asterisk |
17:43.06 | Naikrovek | how often does it happen |
17:43.11 | superbeef | every 2 hours |
17:43.13 | Naikrovek | eek |
17:43.16 | superbeef | yeah |
17:43.20 | superbeef | if not more frequnelty |
17:43.25 | superbeef | i just put this into production last night |
17:43.31 | Naikrovek | oh wow |
17:43.41 | Naikrovek | hard to take it out? easy come easy go |
17:44.01 | superbeef | swap isnt too bad, but other box is asterisk 1.2 |
17:44.07 | dustybin | is it possible to manually input contacts for a polycom phone using the web interface? |
17:44.08 | Naikrovek | what version is this one |
17:44.22 | superbeef | 1.4.26-1 i think |
17:44.31 | p3nguin | I use 1.4.24.1 without any signs of trouble. |
17:44.42 | Naikrovek | dustybin: possible to manually input them via the phone, but not hte webui, or, you can put them into an XML file on the server and the phone will download them next time it reboots |
17:44.46 | p3nguin | core show version |
17:44.59 | superbeef | those errors I have are ring groups calling quees in other PBXs via IAX |
17:45.12 | dustybin | Naikrovek: seems strange why they missed it off the web gui |
17:45.33 | Naikrovek | well if this bug shows up for the bug reporter when an iax2 trunk fails, i would suspect something similar for superbeef |
17:45.39 | dustybin | Naikrovek: im not using tftp or ftp, so i will need to manually put them in... maybe i should start using ftp |
17:45.53 | Naikrovek | dustybin: i thought you were? how many contacts are you going to add |
17:46.02 | dustybin | Naikrovek: not that many, about 15 |
17:46.08 | Naikrovek | ah |
17:46.21 | dustybin | Naikrovek: i configured the phone via the web interface |
17:46.25 | Naikrovek | dustybin: well the web interface is more for admin than daily use by the phone user |
17:46.32 | dustybin | aye ok |
17:46.41 | dustybin | maybe its time to switch to ftp |
17:46.49 | p3nguin | Also, is there any reason I will ever need canreinvite=yes? |
17:47.54 | Naikrovek | dustybin: i use ftp; it's nice. the root of your ftp filesystem will need a polycom/contacts/ folder. set up ftp, place that folder there, add a contact to the phone, reboot the phone, check the folder for a contacts.xml file and then you can see the format of an entry and add additional entries to that. |
17:48.17 | dustybin | thank you :) |
17:48.21 | *** join/#asterisk gardo (n=gardo@121.97.136.60) |
17:48.35 | [TK]D-Fender | dustybin: Didn't I tell you about this... |
17:48.43 | dustybin | hides |
17:48.55 | [TK]D-Fender | dustybin: People configuring Poycom phones outside of a provisioning server should be dragged out an SHOT. |
17:48.58 | superbeef | I don't really know asterisk bug reportined ettitquite, can I respond to this bug report if i'm on 1.4.26? https://issues.asterisk.org/view.php?id=15900&nbn=1 |
17:49.07 | [TK]D-Fender | dustybin: ... and survivors should be shot AGAIN |
17:49.07 | dustybin | :( |
17:49.11 | jaytee | you can run but you can't hide from the [TK]D-Fender |
17:49.26 | dustybin | [FAILS] |
17:49.29 | [TK]D-Fender | jaytee: That's why I go for the knee-caps first :p |
17:49.45 | jaytee | he's 1/4 bloodhound.....see!! his nose is cold!!! |
17:49.47 | dustybin | [TK]D-Fender: i didnt like the idea of installing a ftp server on my box just for 1 phone |
17:50.04 | p3nguin | That's why I said use a tftpd. |
17:50.16 | [TK]D-Fender | dustybin: yippy-kia-yay. You probably already have one, or could in about 1 minute flat |
17:50.29 | [TK]D-Fender | and no, TFTP = bleh |
17:50.36 | jaytee | tftp sucks |
17:50.53 | dustybin | i will install VSFTP |
17:50.55 | p3nguin | Odd. Works for me and for most of Cisco. |
17:50.58 | dustybin | i have used it ages ago |
17:51.06 | jaytee | good move, vsftp works awesome |
17:51.15 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
17:51.19 | dustybin | all the inforation is in the book |
17:51.22 | dustybin | reads |
17:51.43 | jaytee | dustybin, have you read Polycom's whitepaper on configuration? |
17:52.06 | dustybin | i guess running ftp on my local network isnt such a bad thing |
17:52.12 | dustybin | jaytee: no |
17:52.28 | jaytee | highly recommend you do |
17:52.28 | dustybin | ok |
17:54.28 | jaytee | dustybin, if you go here then halfway down the page the last item in the voice section titled "Configuration File Management on Soundpoint IP phones" |
17:54.31 | jaytee | http://www.polycom.com/products/resources/white_papers/index.html |
17:57.06 | dustybin | thanks :D |
17:57.36 | *** join/#asterisk Gnutoo (n=gnutoo@host98-153-dynamic.51-79-r.retail.telecomitalia.it) |
17:57.39 | *** part/#asterisk Gnutoo (n=gnutoo@host98-153-dynamic.51-79-r.retail.telecomitalia.it) |
17:58.57 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
17:59.15 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
18:01.28 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
18:01.39 | *** join/#asterisk eduardovra (i=eduardov@189.42.32.228) |
18:01.46 | grandpapadot | Hey guys, what's the state of presence in 1.6? Does it support userstate via PUBLISH yet? |
18:01.55 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
18:02.15 | [TK]D-Fender | grandpapadot: Not AFAIK |
18:02.51 | grandpapadot | TK: Thanks. Are you guys using the users states with your Polycom's? If so, how? |
18:03.10 | Naikrovek | grandpapadot: yes, via openfire & spark |
18:03.17 | Naikrovek | jabber server & client |
18:03.18 | [TK]D-Fender | grandpapadot: Not usable |
18:03.30 | Naikrovek | but it's not a polycom specific solution |
18:03.32 | [TK]D-Fender | (with *) |
18:03.35 | grandpapadot | Naikrovek: Yea, that's easy, I'm talking about being able to set it via the polycoms. |
18:03.42 | Naikrovek | grandpapadot: no idea |
18:03.44 | grandpapadot | TK: tnx |
18:04.46 | *** join/#asterisk Meaty (n=meaty@office.abi.ca) |
18:04.53 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
18:05.06 | dustybin | the books example of setting up VSFTP is _terrible_ to say the least |
18:05.57 | p3nguin | dustybin: It's simple. I probably have some old confs if they would help. |
18:06.22 | Naikrovek | dustybin: use vsftpd. two minutes you're up, and you probably already have it installed |
18:06.26 | dustybin | i remember setting up VSFTP was quite a mission from past experience! |
18:06.30 | Naikrovek | no no no |
18:06.31 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
18:06.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
18:06.34 | Naikrovek | well |
18:06.37 | Naikrovek | took me 2 minutes |
18:06.50 | dustybin | i did exactly what it said in the book |
18:06.58 | dustybin | i created a user and group: PlcmSpIp |
18:06.59 | p3nguin | That's about a minute longer than it should have taken. You must have stopped to read the comments. |
18:07.01 | raden_work | is there a way to transfer a call and have some sort of status that the transfer went though and also when a call is transfered to have like a beep on the recieving parties line to let them know it transfered |
18:07.09 | raden_work | Naikrovek, howdy |
18:07.23 | Naikrovek | sudo apt-get install vsftpd; vi /etc/vsftpd/vsftpd.conf; vi /etc/vsftpd/ftp_users.conf; /etc/init.d/vsftpd restart |
18:07.32 | Naikrovek | howdy, raden_work |
18:07.47 | Naikrovek | p3nguin: i did, in fact |
18:08.05 | p3nguin | dustybin: I sent you a link to my old confs in a /notice |
18:08.51 | p3nguin | The default vsftpd.conf shouldn't need very much adjustment to get it going. |
18:08.57 | Naikrovek | nope |
18:09.03 | Naikrovek | just enable named users |
18:09.18 | Naikrovek | oh i forgot to "useradd phones; passwd phones" |
18:09.30 | dustybin | p3nguin: where did you send the link? |
18:09.39 | p3nguin | I normally have to add in that I want it to run as a daemon, too. |
18:09.39 | Naikrovek | dustybin: to your irc client |
18:09.46 | dustybin | im running irssi |
18:09.51 | p3nguin | Look on window 1. |
18:09.53 | Naikrovek | dustybin: check the server status window if you have one |
18:09.54 | dustybin | oh yeah |
18:09.57 | dustybin | :) thanks |
18:10.25 | dustybin | im running debian lenny, so maybe things go in different places |
18:10.46 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
18:11.53 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-71-157.link.net.pk) |
18:12.24 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
18:12.25 | garymc | Could logging into Zoiper and reging a SIP Softphone when im in the office on the same connection mess with the firewall on my router? |
18:12.46 | garymc | Cos I got home now and I cant make calls to the office again? |
18:13.32 | garymc | I been getting calls from people in Texas and Isreal all day... again as soon as I get home its not working again :~( |
18:14.49 | [TK]D-Fender | garymc: you weren't outside your work firewall all day, so you are comparing apples & oranges |
18:14.54 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
18:15.44 | garymc | [TK]D-Fender : I wasnt no. But I had people in texas and Isreal and Tunisia login to my PBX as an extension and we made calls perfectly |
18:16.16 | [TK]D-Fender | garymc: says something about your HOME setup, now doesn't it? |
18:16.24 | garymc | Does it? |
18:16.32 | Naikrovek | garymc: is your pbx on your laptop or something |
18:16.44 | garymc | No its a HP proliant Server |
18:16.52 | garymc | in the office with a wEB server and an LTSP |
18:17.03 | Naikrovek | okay so tunisia and israel and texas can call each other still, just you that can't get in |
18:17.15 | garymc | i dont know if they can now |
18:17.30 | garymc | but they could today about an hour before i left the office |
18:19.05 | garymc | but things did start acting weird, cos (i know this aint the channel) but i was logged into extension 202 with my laptop using zoiper. Then before I left my GUI was showing extension 202 logged in when i had unregistered Zoiper and shut down the laptop |
18:20.02 | garymc | it just kept showing as though the phone was there I could call extension 202 and it rang but there was no phone connected to 202? |
18:20.18 | KavanS | garymc, are you trying to run soft phones on LTSP? |
18:20.29 | garymc | KavanS : NO |
18:20.31 | garymc | no |
18:20.35 | KavanS | ok, my bad... |
18:21.24 | garymc | So to me either my router firewall is deciding to not work after a few hours or possibly my GUI is messing with settings somehow? |
18:21.56 | garymc | but i dont get why my GUI would show the third phone connected when it clearly wasnt connected |
18:22.48 | *** join/#asterisk w9sh (n=chatzill@adsl-068-209-117-205.sip.asm.bellsouth.net) |
18:22.59 | *** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar) |
18:23.01 | [TK]D-Fender | garymc: it doesn't miraculously change when you leave. |
18:23.16 | garymc | No thats what i thought, but it is |
18:23.22 | [TK]D-Fender | garymc: So either you specifically changed something after those successful calls, or your home setup is at fault |
18:23.42 | [TK]D-Fender | garymc: and * can't FuBAR your firewall |
18:23.55 | garymc | no could Zoiper? |
18:23.57 | *** part/#asterisk user4545 (n=sipip@dslb-092-074-252-179.pools.arcor-ip.net) |
18:24.15 | garymc | so why is freepbx showing 3 phones connected when only 2 where? |
18:24.32 | [TK]D-Fender | garymc: go look at something real |
18:24.42 | garymc | what you mean? |
18:24.47 | [TK]D-Fender | garymc: go look at something real |
18:25.05 | dustybin | at last, vsftp works with PlcmSpIp username and PlcmSpIp password :D thanks for help! |
18:25.15 | dustybin | the book misses vital instructions |
18:25.26 | [TK]D-Fender | dustybin: Don't. Go setup a completely different suer & strong password |
18:25.32 | [TK]D-Fender | dustybin: NEVER use those defaults |
18:25.35 | dustybin | [TK]D-Fender: its local only! |
18:25.47 | [TK]D-Fender | dustybin: and just enter that in during the boot menu |
18:25.47 | dustybin | 21 will _NEVER_ be open to the outside :D |
18:25.55 | dustybin | [TK]D-Fender: ok .. |
18:25.56 | wonderworld | which open-source iax-softphone would be the most advanced by now? |
18:25.57 | garymc | <[TK]D-Fender>garymc: go look at something real ------------< like what a car? what are you saying? |
18:25.57 | superbeef | are there any good load testing scripts for asterisk? |
18:26.02 | [TK]D-Fender | dustybin: Doesnt' amtter. If another PC gets hacked it opens your server to attack |
18:26.14 | [TK]D-Fender | gay&^$#ing CLI. SIP DEBUG. |
18:26.14 | dustybin | aye good point |
18:26.23 | [TK]D-Fender | garymc: &^$#ing CLI. SIP DEBUG. |
18:26.47 | garymc | oh |
18:26.50 | [TK]D-Fender | superbeef: take a look at "sipp" |
18:29.30 | wcselby | ahhh |
18:29.35 | wcselby | pizza buffet |
18:29.48 | wcselby | and I get a call right in the middle "hey we can't login to the conference bridge" |
18:30.05 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
18:30.18 | wcselby | "so and so had me change all the passwords, they have the spreadsheet with the new ones" |
18:30.22 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
18:30.22 | wcselby | "well they're not heree" |
18:30.22 | garymc | [TK]D-Fender : This was home to office answer phone. I hear nothing I know its broken. http://pastebin.ca/1568668 |
18:30.42 | wcselby | me - "damnit!" |
18:31.10 | superbeef | [TK]D-Fender: thanks..... How do you test T1 cards |
18:32.30 | [TK]D-Fender | garymc: I see noone answered that call |
18:32.49 | garymc | yep the answer phone should have |
18:33.05 | garymc | I know its broken cos I dont hear the answer phone |
18:33.06 | [TK]D-Fender | garymc: Nobody answered the phone... |
18:33.21 | garymc | but shoouldnt i get the answerphone? |
18:33.30 | [TK]D-Fender | garymc: WTF is "answerphone"? |
18:33.37 | garymc | Voicemail |
18:33.40 | [TK]D-Fender | GAH |
18:33.50 | [TK]D-Fender | swats garymc with a disctionary |
18:33.54 | garymc | When it works I hear voicemail |
18:33.56 | [TK]D-Fender | dictionary even |
18:34.13 | [TK]D-Fender | garymc: Fine. Maybe your home internet connection is crap |
18:34.14 | garymc | Im F&*KING ENGLISH!!! you know the language we all spek here |
18:34.21 | garymc | *SPEAK |
18:34.22 | dustybin | [TK]D-Fender: is there also a way to change the username from: PlcmSpIp to something else? |
18:34.37 | Naikrovek | dustybin: yes |
18:34.37 | [TK]D-Fender | garymc: "answerphone" is not a valid term |
18:34.39 | dustybin | ace |
18:34.49 | Naikrovek | [TK]D-Fender: in britain it's acceptable, which is where he is |
18:34.52 | dustybin | i will use username: polycom password: strong |
18:34.53 | Skeeter- | ~answerphone |
18:34.57 | garymc | [TK]D-Fender : My internet at home is better than office! |
18:34.58 | dustybin | and make sure the shell cannot login |
18:35.01 | [TK]D-Fender | dustybin: rebooth the phone. Enter setup. Changer the boot server IP, user & pass threre |
18:35.06 | dustybin | thanks |
18:35.16 | [TK]D-Fender | garymc: Apparently only FASTER. But you might be getting FILTERED |
18:35.18 | Naikrovek | yup what [TK]D-Fender said |
18:35.34 | [TK]D-Fender | garymc: Or your home router could be a flaming pile of shit |
18:35.38 | garymc | Well it worked last night when I got my Co worker to set the Router to DMZ |
18:35.50 | Naikrovek | there's the clue |
18:35.50 | garymc | Calls worked fine |
18:36.10 | garymc | Yeah DMZ. I tutrnt DMZ off today and set the ports and it worked all day! |
18:36.49 | Naikrovek | garymc: that was at the office, yes? |
18:37.02 | wcselby | no, I was able to login from MY office to HIS server (whereever that is) |
18:37.09 | wcselby | and everythign worked just fine |
18:37.13 | Naikrovek | ah |
18:37.17 | wcselby | garymc, you want me to try to login again? |
18:37.18 | Naikrovek | but not any more |
18:37.20 | Naikrovek | ? |
18:37.22 | *** join/#asterisk |omni| (n=rob@67.185.91.139) |
18:37.31 | wcselby | Naikrovek - evidently. i don't know, I went to lunch since then |
18:37.33 | garymc | Naikrovek : But I had people in texas, Tunisa and Isreal log in and we made calls to each other |
18:37.37 | Naikrovek | yeah |
18:37.41 | *** join/#asterisk Takapa (i=vegard@junior.svanberg.no) |
18:38.01 | |omni| | anyone having a problem with dahdi channels under 1.6.1.5 and dahdi 2.2.0 ? |
18:38.05 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
18:38.05 | *** mode/#asterisk [+o mog] by ChanServ |
18:38.05 | [TK]D-Fender | garymc: And you then left everything alone, went home, tried to log in the same as them and it doesn't work? |
18:38.08 | wcselby | i wonder if I'm logging my chats |
18:38.21 | wcselby | i can find the info to relog into his server |
18:38.26 | |omni| | I'm getting this crap [Sep 16 09:57:32] ERROR[10808]: chan_dahdi.c:10760 dahdi_pri_error: No more room in scheduler |
18:38.26 | wcselby | bleh, re-register with his server |
18:38.26 | |omni| | [Sep 16 09:57:32] ERROR[10808]: chan_dahdi.c:10760 dahdi_pri_error: Asked to delete sched id -1??? |
18:38.35 | |omni| | on my PRI..and it completely stops responding |
18:38.47 | garymc | [TK]D-Fender : Just one thing..... Freepbx (I know other channel) was showing 3 phones connected when there was only 2 |
18:39.04 | Naikrovek | garymc: don't trust the FOP |
18:39.04 | |omni| | so..I came from zaptel..noticed that previous default was to reset PRI chans every hour..new dahdi default is not..so I have them reset |
18:39.04 | [TK]D-Fender | garymc: Please confirm the exact situation I jsut told you |
18:39.21 | |omni| | works most of the time but usually in the a.m. after low or no call transaction is just dies |
18:39.23 | garymc | Yes I touched nothing apart from make calls |
18:39.50 | garymc | but the GUI started showing my Zoiper Extension connected when it wasnt |
18:39.52 | [TK]D-Fender | garymc: then your home setup clearly ahs issues |
18:40.01 | garymc | no it doesnt |
18:40.11 | garymc | definatley my home setup doesnt |
18:40.13 | [TK]D-Fender | garymc: Yes, it does |
18:40.16 | garymc | oh ok |
18:40.25 | garymc | how does it |
18:40.39 | Naikrovek | [TK]D-Fender: his independent office pbx stops working when he leaves; nothing else changes. i think the timing is just a coincidence. someone is unplugging yoru server I bet |
18:40.45 | garymc | have you got a test server I could make a call too from my home that you know works? |
18:40.47 | wcselby | garymc - I just logged in using the info you gave me earlier, and dialed ext 201 - and went to voicemail |
18:40.48 | [TK]D-Fender | garymc: You seem to have problems with the most basic aspects of the scientific process |
18:41.11 | garymc | wcselby : NO WAY! |
18:41.13 | wcselby | garymc - well, it rang first, then went to voicemail |
18:41.14 | Naikrovek | wcselby: ah hah! okay now we know that garymc's system is up |
18:41.27 | wcselby | garymc - YES WAY |
18:41.32 | wcselby | garymc- ;) |
18:41.33 | Naikrovek | garymc: i'm with [TK]D-Fender now; your have outgoing ports blocked |
18:41.44 | [TK]D-Fender | garymc: Everything works for other remote users. You leave the server as it was. You go home. You places calls. YOURS don't work. Answer : your home setup is fucked up. It is just you and you are in a pathetic state of denial. |
18:41.45 | garymc | shit! I hate it that [TK]D-fender has just .......... told me again!!! |
18:42.00 | wcselby | lol |
18:42.05 | garymc | lol |
18:42.12 | Naikrovek | it does suck when you're mad and arguing and you realize that your point is incorrect |
18:42.20 | Naikrovek | [TK]D-Fender has shown me the way a few times |
18:42.22 | garymc | yes :#( |
18:42.27 | Naikrovek | eh, it happens |
18:42.32 | [TK]D-Fender | garymc: And your home ISP could be completely fucking you over and it wouldn't even be a setting you have any control over |
18:42.33 | Naikrovek | learn from it |
18:42.44 | garymc | fukc sake |
18:42.50 | Naikrovek | maybe he could SSH into his * server and do some tunnelling |
18:43.00 | Naikrovek | maybe.. not sure if you can tunnel that many ports |
18:43.05 | wcselby | [TK]D-Fender knows what he's speaking of |
18:43.11 | Naikrovek | yeah he does |
18:43.14 | wcselby | Naikrovek - you could do it with an iax phone |
18:43.25 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
18:43.28 | Naikrovek | it sucks when he tells me i'm wrong when i'm sure i'm not, then it turns out i am wrong |
18:43.31 | wcselby | iax is only one port to worry about, yes? |
18:43.34 | garymc | appologises for ever doubting the mighty [TK]D-Fender :P |
18:43.38 | Naikrovek | wcselby: yes |
18:43.45 | [TK]D-Fender | garymc: ISPs that offer VoIP services as well often purposefully interfere with other VoIP services jsut to blackmail you into using theirs |
18:43.53 | Naikrovek | yeah |
18:43.54 | Naikrovek | they do |
18:43.59 | Naikrovek | BT is bad about that I bet |
18:44.03 | Naikrovek | they're a bunch of tools |
18:44.07 | garymc | possibly |
18:44.10 | Naikrovek | tried to patent hyperlinks a few years back |
18:44.12 | Naikrovek | idiots |
18:44.27 | garymc | I can only get BT in my area |
18:44.36 | garymc | at home |
18:44.40 | [TK]D-Fender | garymc: go setup a VPN to the office and see if that works. |
18:44.43 | garymc | so we use BT in office too |
18:44.47 | Naikrovek | well ask them if they're blocking things |
18:44.48 | wcselby | you could always call up and ask politely for them to open ports for you.......... |
18:44.52 | Naikrovek | ah yes vpn |
18:44.59 | Naikrovek | vpn vpn vpnvpn |
18:45.01 | wcselby | .... |
18:45.09 | wcselby | laughs quietly to himself |
18:45.16 | [TK]D-Fender | garymc: And just because you use BT in both doesn't mean they get the same treatment either |
18:45.42 | garymc | yeah one is business BT and the other is Residential |
18:45.48 | garymc | fuk sake! |
18:45.48 | Naikrovek | yeah, comcast is like that. business customers can do what they want, home customers have certain ports blocked so they can't do certain things |
18:45.55 | [TK]D-Fender | garymc: and I never say you do a proper UDP port test across your range. |
18:46.11 | [TK]D-Fender | saw* |
18:46.12 | garymc | my home range? |
18:46.12 | Naikrovek | i totally understand garymc's mood right now |
18:46.18 | wcselby | at&t u-verse, you have to call and get 2nd tier support to remove the block on port 25 so that you can use an email server other than theirs |
18:46.23 | [TK]D-Fender | garymc: both |
18:46.32 | Naikrovek | garymc: grab .. ooh what's it called. .. nmap |
18:46.36 | wcselby | but they will remove, if you convince them you aren't stupid and aren't going to become a spambot |
18:46.48 | Naikrovek | and see if you can scan your voip server from home |
18:46.56 | garymc | right ok |
18:47.22 | Naikrovek | scan TCP & UDP, ports 1024-20000. it will take some time |
18:47.38 | Naikrovek | probably overkill but you'll see what you need to see |
18:48.17 | *** join/#asterisk sn00p- (i=sn00p@c-66-41-139-23.hsd1.mn.comcast.net) |
18:48.27 | sn00p- | DO I need any hardware for asterisk to work? |
18:48.37 | superbeef | Is DAHDI echo cancelation super resource expensive? |
18:48.39 | Naikrovek | sn00p-: you need a linux box to run it on |
18:48.55 | sn00p- | Naikrovek, yea I know but do I need any phone hardware? |
18:48.56 | Naikrovek | superbeef: probably done in hardware, no not super intensive |
18:49.05 | sn00p- | like a pbx switch or something |
18:49.05 | Naikrovek | sn00p-: not necessarily |
18:49.28 | wcselby | sn00p- - you can do it all completely with a linux box running asterisk and a softphone |
18:49.35 | superbeef | Naikrovek: Sadly I have a Sangoma A101 and not the D, so software echo cancelation is enabled... I'm still getting load averages much higher than I should for 5 or 6 calls |
18:49.38 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
18:49.42 | sn00p- | Naikrovek, so is it possible for me to set up asterisk and be ale to send SMS? |
18:49.46 | Naikrovek | sn00p-: you can use softphones and if you want to call out you'll need an IP voice provider |
18:49.49 | wcselby | sn00p- - but it's be a pretty small server. it depends on what you want to do |
18:49.51 | *** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
18:50.16 | sn00p- | wcselby, all I want it to do is send SMS messages on the asterisk server |
18:50.18 | Naikrovek | sn00p-: who are you trying to send SMS to |
18:50.24 | sn00p- | Friends |
18:50.27 | Naikrovek | sn00p-: what provider(s) |
18:50.35 | [TK]D-Fender | snoo* is NOT an SMS server |
18:50.44 | [TK]D-Fender | sn00p-: * is NOT an SMS server |
18:50.45 | wcselby | sn00p- - i haven't tried sending sms, but I think there's an app for that |
18:51.03 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
18:51.04 | [TK]D-Fender | wcselby: Only works on EU PRI's |
18:51.25 | Naikrovek | sn00p-: there are pages online that you can use to do that. usually providers have their own page where you can send to customers of that provider |
18:51.25 | sn00p- | wcselby, do you recall the app for SMS ? |
18:51.38 | sn00p- | Naikrovek, yea But I want my own |
18:51.38 | [TK]D-Fender | sn00p-: You have an EU PRI? |
18:51.58 | Naikrovek | he's in minnesota, based on his hostname |
18:52.22 | [TK]D-Fender | Naikrovek: thats what it says alright... |
18:52.33 | [TK]D-Fender | Naikrovek: but that doesn't mean it counts... |
18:52.36 | sn00p- | So there is something seperate for sms server? |
18:52.38 | Naikrovek | so probably not... oh wait EU doesn't mean europe probably |
18:52.46 | [TK]D-Fender | sn00p-: Not Asterisk <- |
18:53.02 | Naikrovek | yeah this isn't something asterisk was meant to handle; not natively anyway |
18:53.04 | wcselby | [TK]D-Fender i thought I had read that chan_mobile will do that with a mobile phone connected via bluetooth |
18:53.09 | [TK]D-Fender | Naikrovek: And yes, I am asking if he had a European PRI |
18:53.30 | Naikrovek | any solution would involve some goofy hack that wraps around http posts or bluetooth phone |
18:53.48 | wcselby | yeah, that |
18:53.58 | *** part/#asterisk sn00p- (i=sn00p@c-66-41-139-23.hsd1.mn.comcast.net) |
18:54.00 | *** join/#asterisk i9 (n=arthurh@70.56.139.60) |
18:54.04 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
18:54.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
18:54.05 | [TK]D-Fender | BAI BAI |
18:54.08 | Naikrovek | not even a goodbye or a thanks |
18:54.13 | wcselby | hahaha |
18:54.19 | [TK]D-Fender | Naikrovek: Consider it a small mercy |
18:54.21 | Naikrovek | yeah |
18:54.23 | Naikrovek | i do |
18:54.25 | Naikrovek | i was happy abou tit |
18:54.27 | Naikrovek | about it |
18:54.35 | wcselby | you're not happy about tit? |
18:54.39 | Naikrovek | lol |
18:54.45 | Naikrovek | my favorite bird |
18:55.32 | Naikrovek | here's something that made me laugh, if anyone is bored: http://www.collegehumor.com/article:1791517 |
18:55.44 | wcselby | Naikrovek - you might be able to do something clever with google voice / jabber and asterisk |
18:56.05 | *** join/#asterisk wathek (n=wathek@41.224.194.132) |
18:56.09 | wathek | hey all |
18:56.14 | wcselby | just an idea that popped into my head, no idea if it would work at all |
18:56.32 | Naikrovek | wcselby: for SMS? could do IM that way perhaps |
18:56.44 | Naikrovek | wathek: yahey |
18:56.47 | wathek | is it possible to configure nicknames as extensions instead of numbers ? |
18:57.03 | wcselby | wathek - yes, but then what? |
18:57.19 | Naikrovek | wathek: i think so, but keeping names and numbers separated, uncoupled, is better |
18:57.24 | wcselby | wathek - you can do something like 'exten => nickname,1,NoOp() |
18:57.29 | wathek | wcselby, it's easier to keep in minde than numbers |
18:57.40 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
18:57.40 | wcselby | wathek - you have phones that can dial names instead of numbers? |
18:57.42 | Naikrovek | wathek: how do you dial a name on a phone though |
18:57.42 | wathek | wcselby, ok thank you |
18:57.52 | wathek | Naikrovek, lol |
18:57.56 | wcselby | i've heard some phones can do it |
18:58.03 | wathek | no but I'm just configuring just SIP |
18:58.11 | wathek | Naikrovek, yep I think it's possible |
18:58.19 | Chainsaw | My Cisco 7960s have a URL/Number softkey. |
18:58.34 | wcselby | well, are you talking dialplan extensions or just sip usernames? |
18:58.39 | wcselby | because sip usernames is really simple |
18:58.41 | [TK]D-Fender | wathek: if you aren'te expecting to use those in an IVR, sure... |
18:58.58 | [TK]D-Fender | wcselby: You don't dial usernames.... |
18:59.19 | wathek | wcselby, just SIP |
18:59.28 | wcselby | [TK]D-Fender - I know that, but he said he's just configuring SIP |
18:59.29 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:e0b2:150:ec3e:2a25) |
18:59.40 | Naikrovek | could do exten => 123,1,Dial(SIP/DaveyMakinCopies); |
18:59.51 | wcselby | wathek - ^^ what Naikrovek said |
19:00.15 | wathek | and then users could put DaveyMakinCopies to call that number ? |
19:00.17 | Naikrovek | then you could name your devices, and use the exten => lines for extension to phone napping |
19:00.23 | Naikrovek | wathek: that, or 123 |
19:00.28 | wathek | cool |
19:00.29 | wathek | thank you |
19:00.37 | wcselby | in your sip.conf file, instead of naming the phones [201], you can name them [wathek]. then you dial using Dial(SIP/wathek) |
19:00.49 | Naikrovek | yeah |
19:00.53 | wathek | ok |
19:00.55 | wathek | that's cool |
19:01.06 | wathek | have to find now how to limit the call duration |
19:01.33 | wcselby | as in, once a call reaches it's limit you want to cut it off? |
19:01.39 | Naikrovek | that would actually make sense in a way, you could keep track of MAC addresses that way. [0004abcdef10] - then exten => 444,1,Dial(SIP/0004abcdef10); |
19:01.55 | wathek | wcselby, yep |
19:02.23 | Naikrovek | how long do you want to limit to |
19:02.34 | wathek | 10minutesz |
19:02.40 | wcselby | [TK]D-Fender - didn't you help someone with that the other day? |
19:03.16 | i9 | Hey guys, we're looking for a small group of closed beta testers for the latest internal beta of iSymphony -- with Asterisk 1.6 support -- if anyone's interested in participating, shoot an e-mail to isymphony-beta@i9technologies.com with environmental specs.. |
19:03.34 | Naikrovek | i9: would love to but i broke it on my laptop |
19:03.38 | Naikrovek | client won't launch anymore |
19:03.40 | Naikrovek | yay |
19:03.49 | Naikrovek | won't uninstall completely either |
19:03.50 | Naikrovek | yay |
19:03.53 | Naikrovek | heh |
19:03.55 | i9 | Naikrovek, want some help? |
19:03.56 | wcselby | Naikrovek - lol |
19:04.00 | Naikrovek | no, i gave up on it |
19:04.03 | wcselby | i remember that day |
19:04.08 | Naikrovek | i'll install it in a virtual machine and play there |
19:04.27 | Naikrovek | if i ever get any time |
19:04.27 | wcselby | wathek - limit call durations - http://forums.whirlpool.net.au/forum-replies-archive.cfm/776539.html |
19:04.31 | wathek | wcselby, call thank you so much |
19:04.57 | *** join/#asterisk flohack (n=fhackenb@84.115.131.198) |
19:05.05 | [TK]D-Fender | wcselby: Yup, that would be yesterday morning |
19:05.13 | wcselby | wathek - those are for older versions of asterisk, but should get you started on the right path |
19:05.19 | i9 | Naikrovek, it's all self contained. 86ing the directory and a registry entry should get rid of it in it's entirety -- give us a ring or shoot an e-mail and we'll be happy to help.. we just need 1.6 testers at the moment |
19:05.24 | wathek | wcselby, ok |
19:05.53 | Naikrovek | i9 yeah it's based on eclipse, which i'm very familiar with. will have to nuke the registry and try again then |
19:06.18 | superbeef | So does asterisk spawn a process for every phone call? |
19:06.29 | i9 | Naikrovek, No, the ONLY registry entry we put in there is for the uninstaller (to appear in add/remove programs) |
19:06.31 | wcselby | superbeef - i don't think so..... |
19:06.32 | Naikrovek | superbeef: don't think so... |
19:06.37 | superbeef | Hmm |
19:06.39 | Naikrovek | i9: okay |
19:06.41 | i9 | Naikrovek, so not nuking the registry, just removing that entry |
19:06.47 | Naikrovek | i9: yes i know that |
19:06.48 | superbeef | well i have liek 15 asterisk instances running |
19:06.56 | Naikrovek | i9: you keep plugging for beta testers i'll work on this |
19:06.58 | wcselby | i9, Naikrovek, lol. I think you both had this convo like two weeks ago |
19:07.08 | superbeef | more like 20 |
19:07.09 | Naikrovek | that was seanmh, not i9 |
19:07.18 | i9 | haha.. |
19:07.30 | wcselby | Naikrovek - ahh, lol. same convo as last time though |
19:07.35 | Naikrovek | pretty much |
19:07.37 | [TK]D-Fender | i9: I suspect if you help Naikrovek with his issues he might be happy to give it a look... |
19:07.38 | seanmh | I'm here :D |
19:07.43 | i9 | Naikrovek, No more plugging -- one post is enough I think in #freepbx, #asterisk and #trixbox ;) |
19:08.10 | Naikrovek | [TK]D-Fender: i'm CERTAIN that my issues are my own fault |
19:08.18 | wcselby | i9 - using 1.4.26. Will probably be building a 1.6 cluster in a couple months. how long do you plan on running your beta? |
19:08.23 | Naikrovek | [TK]D-Fender: i just need to figure out what is going on and fix it |
19:08.30 | [TK]D-Fender | Naikrovek: And I'm still pretty sure you'd be happy for them to be solved :) |
19:08.34 | Naikrovek | yeah |
19:08.39 | Naikrovek | but low on priority list |
19:08.41 | [TK]D-Fender | Naikrovek: My way both of you win |
19:09.02 | i9 | wcselby, It's likely we'll be out of beta in a few months, General release should be within 30 days.. even if you're running 1.4, hop in.. there's quite a bit of bug-fixing and new stuff.. |
19:09.22 | wcselby | i9 - i'll send an email when I get a chance this week. :) |
19:10.15 | i9 | wcselby, awesome -- note in the e-mail that you're an irc contact and we'll throw you up top |
19:10.44 | *** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-bnbwwkrzpnjhwcxt) |
19:10.44 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
19:12.01 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
19:12.02 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-okopfrwoucuizora) |
19:12.40 | [TK]D-Fender | OMG DIGIUM RAIDING PARTY! |
19:13.08 | [TK]D-Fender | hides his contraband elecom gear |
19:13.11 | [TK]D-Fender | telecom* |
19:14.04 | Katty | oh man, the sleepies are getting me |
19:14.56 | [TK]D-Fender | The Dream Police... reside in my head! |
19:15.17 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
19:15.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
19:16.16 | Naikrovek | isymphony people: how the hell does admin/secret not work when: <SystemAdmin userName="admin" password="secret"/> |
19:17.25 | Naikrovek | no time to investigate this right now; i suppose i'll call one day |
19:17.40 | Naikrovek | thought i'd try again and maybe the gods i don't believe in would smile on me today |
19:17.43 | Naikrovek | nope |
19:18.31 | dustybin | i keep on getting: Could not contact boot server on my polycom |
19:18.37 | dustybin | the menus are confusing |
19:18.50 | dustybin | however, i did select ftp and put in my ip and username + password |
19:18.59 | Naikrovek | dustybin: your authorization is wrong, or your ftp server isn't running |
19:19.09 | [TK]D-Fender | dustybin: And confirmed that your server matches? |
19:19.16 | [TK]D-Fender | dustybin: logged in with another client? |
19:19.37 | Naikrovek | yeah use cmdline ftp client to test |
19:19.45 | Naikrovek | use correct hostname, username, and password, verify that it works |
19:22.03 | seanmh | Naikrovek: hover over the server icon in the administration section next to server and let me know what it says |
19:22.13 | garymc | Right heres some news. BT reckon they arnt blocking nothing and i need to unblock the ports on my home HUB! |
19:22.17 | *** join/#asterisk raden (n=jon@69.179.99.17) |
19:22.29 | garymc | so doen that and still not working :S |
19:22.31 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
19:22.44 | [TK]D-Fender | garymc: And its still just you. |
19:22.57 | [TK]D-Fender | spins up some Eric Carmen, jsut for garymc |
19:23.25 | jaytee | Katty, it was probably from the sour cream in the stroganoff |
19:24.25 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
19:24.27 | Katty | jaytee: it was the cream cheese, actually (= |
19:24.40 | wcselby | garymc - have you tried using xlite to connect from home, just to be sure? |
19:24.48 | wcselby | garymc - that's what I was using to connect to you |
19:25.39 | jaytee | Katty, I follow a similar recipe to the one you posted but I use a can of mushroom gravy and a can of Campbell's Beefy Mushroom soup instead. |
19:26.19 | [TK]D-Fender | wcselby: Does it still work for you trying to call in to his office? |
19:26.38 | wcselby | it did oh....i dunno, 20 minutes ago? |
19:26.38 | p3nguin | He's still on that? Wow. |
19:26.52 | [TK]D-Fender | wcselby: Ok, that confirms it |
19:27.33 | wcselby | and it still works now |
19:28.00 | *** join/#asterisk merkurie (n=merkurie@192.153.163.45) |
19:28.05 | Katty | jaytee: yum. i might try that for some additional deep flavor (= |
19:29.14 | merkurie | anyone else ever run into a problem where they have a router, with some sip devices inside behind nat, asterisk server on the public internet side, but the router always hands out the same source port number when the sip clients connect to ast/udp/5060? even though the sip clients all have different ips inside nat? |
19:29.32 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
19:29.41 | jaytee | Katty, when I make it I often go all out and substitute sirloin tips I've broiled rare on the grill and then cut into cubes instead of ground beef/turkey |
19:29.55 | ayeso | Anyone using SS7 with asterisk? |
19:29.57 | Chainsaw | merkurie: Some routers try to be clever about SIP and rewrite packet headers. |
19:30.09 | Chainsaw | merkurie: You may have to disable this "feature". |
19:30.41 | *** join/#asterisk Micc (n=dotirc@c-98-225-59-171.hsd1.wa.comcast.net) |
19:31.25 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
19:31.37 | merkurie | Chainsaw, dd-wrt? |
19:31.40 | Katty | jaytee: that sounds delicious |
19:31.46 | dustybin | grrrrr because i put bin/false on the end of the passwd file for polycom, it failed |
19:32.04 | jaytee | we really need to build a comprehensive list of router manufacturers whose products try to rewrite SIP packet headers so that we can more efficiently purge them when the violent phase of the revolution begins |
19:32.12 | Micc | can I make two mailboxes notify a single sip account? |
19:32.24 | Chainsaw | merkurie: I don't know every router that was ever made by heart. You'll have to check your settings and see whether anything involves SIP header rewriting. |
19:32.29 | p3nguin | dustybin: You sure that's why? You shouldn't need a login shell to get files from the ftpd. |
19:32.30 | dustybin | how can one reboot a polycom 321 without keep on pulling out the power cord? |
19:32.47 | dustybin | p3nguin: it needs /nologin, not /false |
19:32.57 | merkurie | Chainsaw, kk, thanks |
19:33.04 | wcselby | dustybin - isn't there a restart command in the settings menu? |
19:33.12 | wcselby | maybe advanced settings? |
19:33.17 | wcselby | i know the 601 has one |
19:33.23 | jaytee | dustybin , you can reboot from the Menu key |
19:33.59 | p3nguin | dustybin: I doubt that's the case. Chances are that your false shell just wasn't listed in /etc/shells. |
19:34.06 | p3nguin | dustybin: Try pressing the 4, 6, 8 and * keys simultaneously. |
19:34.26 | jaytee | dustybin, press Menu, choose Settings then Advanced (enter password) and then option 3 Restart Phone |
19:34.57 | Naikrovek | dustybin: menu, 3, 1, 4, yes softkey |
19:35.16 | dustybin | thanks :D |
19:35.36 | dustybin | here we go, hopefully it will use ftp now |
19:35.57 | dustybin | i've named my polycom 'sod' |
19:35.58 | dustybin | :D |
19:36.03 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
19:36.13 | dustybin | 'could not contact boot server..' |
19:36.19 | dustybin | checks ftp with a client again |
19:36.20 | wcselby | Micc - if you find out, let me know...but i think it depends on the type of sip device you want to notify |
19:36.25 | dustybin | shit i changed the password |
19:36.27 | dustybin | of course |
19:36.31 | Naikrovek | lol |
19:37.28 | dustybin | its _SO_ easy to make a mistake, one tiny little error |
19:37.35 | Naikrovek | yeah |
19:37.36 | dustybin | one tiny digit |
19:37.41 | Naikrovek | but once it's set up it'll work forever |
19:37.46 | Naikrovek | until you change a passwd again |
19:37.53 | p3nguin | like four bits or so? |
19:37.58 | dustybin | i think system admins need to pay close attention to detail if they want to be good admins |
19:38.05 | bmoraca | dustybin: the phone will also give you that error if it doesn't find a file it can download, such as its MAC.cfg file |
19:38.26 | Naikrovek | bmoraca: true, but he can see what it is looking for via the ftp server log |
19:38.35 | bmoraca | right |
19:39.00 | wcselby | bmoraca - you read earlier that I was able to get 2 softphones over my at&t 2wire? |
19:39.06 | wcselby | from 2 separate computers |
19:39.29 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
19:39.37 | bmoraca | wcselby: i can get them to register fine, but they do not stay registered. after 30 seconds or so, they are no longer available for incoming calls |
19:39.51 | bmoraca | or, rather, 5 minutes might be a closer estimate |
19:40.01 | wcselby | hmmm....okay, I didn't last that long |
19:40.08 | wcselby | maybe a minute or so total? |
19:40.08 | bmoraca | they work for 5 minutes, but then the 2wire loses its NAT and the phone no longer worked |
19:40.23 | wcselby | i got both hooked up, then dialed out on both, which both worked. |
19:40.29 | wcselby | then I shut down, cause I was going to bed |
19:40.39 | wcselby | i'll try again tonight, let them run |
19:40.42 | p3nguin | Sounds like you need a new network appliance if NAT stops working every five minutes. |
19:40.44 | wcselby | overnight, then test in the morning |
19:40.55 | bmoraca | dialing out was never the issue...it was dialing IN to the phone that was the issue |
19:41.14 | bmoraca | p3nguin: indeed. this is why i was complaining about SOHO gear last night (one of the reasons, anyway) |
19:41.17 | wcselby | i know - i'll check that also |
19:41.28 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
19:41.53 | p3nguin | If NAT stops working, that means computers are losing their ability to connect to the internet, as well. Correct? |
19:42.12 | ayeso | p3nguin: no |
19:42.23 | bmoraca | p3nguin: read what i wrote again. NAT doesn't stop working, the router just does not preserve the phone's NAT |
19:42.39 | garymc | wcselby i will try xlite now |
19:42.42 | p3nguin | That wasn't what I saw in the above-listed text. |
19:42.49 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
19:43.27 | p3nguin | "then the 2wire loses its NAT", where 2Wire indicates the network appliance. |
19:43.33 | bmoraca | wcselby: FWIW, the 2wire gateway for Uverse doesn't seem to have this issue. |
19:43.45 | bmoraca | p3nguin: right, however "its" refers to the phone |
19:44.00 | p3nguin | Ah, improper grammar usage caught me again. |
19:44.48 | bmoraca | there was nothing improper about my grammar. within the context of my statement, it was very clear that i was talking about phones |
19:44.48 | p3nguin | I beg to differ. |
19:44.48 | wcselby | bmoraca - then I won't be much help, I've got at&t uverse |
19:45.27 | p3nguin | Regardless, I won't be of any help with the situation, so I'll refrain from additional comment. |
19:45.33 | bmoraca | p3nguin: whatever. you have fun with that. meanwhile, i'll have fun making money off my hosted pbx customers. |
19:45.42 | p3nguin | laughs |
19:45.49 | *** part/#asterisk deeperror (n=deeperro@adsl-76-226-149-104.dsl.sfldmi.sbcglobal.net) |
19:46.24 | bmoraca | wcselby: i've got uverse at home as well, and that 2wire seems to be far superior to the original 2wires AT&T sold, which is what this particular customer has |
19:46.33 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
19:46.57 | wcselby | bmoraca - :( |
19:47.01 | bmoraca | wcselby: however, i sold him on a new router and a discrete DSL modem, which should resolve the issue. |
19:47.11 | wcselby | cool :) |
19:47.58 | wcselby | one of the AA's at my client wants to buy a CS70N headset - anyone ever use this? (from plantronics) |
19:48.20 | wcselby | well, anyone ever use this successfully with asterisk / polycom phones ? |
19:48.34 | bmoraca | not that one in particular, but plantronics tend to be very good |
19:48.49 | wcselby | yeah, they usually buy the CS55s here |
19:49.01 | bmoraca | wcselby: if you have doubts about compatibility, tell them to get the one with the lifter and it'll be a non-issue |
19:49.11 | wcselby | yeah, that's what they're doing |
19:49.13 | wcselby | cool |
19:49.35 | bmoraca | from what i remember, though, polycoms tend to be compatible with almost any headset...Ciscos, on the other hand, are not |
19:49.54 | p3nguin | wcselby: It has a good customer rating, if that means anything. |
19:50.36 | jaytee | we use the Jabra GN2010 ST headset but it's not wireless, it uses a cord |
19:51.03 | jaytee | but it works great on Polycom 330's and 550's |
19:51.05 | p3nguin | Hmm. Only one person rated it on plantronics web site, though. |
19:51.39 | *** join/#asterisk evil_gordita (n=evilgord@ip24-254-160-77.rn.hr.cox.net) |
19:52.06 | Meaty | Hi Every Body! |
19:52.09 | Meaty | I want use Realtime SIP with ODBC. I have set configurations in extconfig.conf and res_odbc.conf. I add some rows in table with type = peer. When i make "sip show peers" i have only my peers in sip.conf. But when i make "realtime load sippeers username test", i see the colums values for my peers with username "test". Am I suposed to see the peer "test" when i make "sip show peers" ? |
19:54.20 | *** join/#asterisk [netman] (n=netman@216.Red-88-17-240.dynamicIP.rima-tde.net) |
19:54.51 | wcselby | <Meaty> Hi Every Body! <---- Hi Doctor Nick! |
19:55.05 | wcselby | sorry, i can't help with your issue |
19:56.14 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
19:56.16 | doolittlework | hi there i am stuck with the monitor command, [TK]D-Fender said i must use "g" in the dial command as an option to continue with the dialplan if the called party drops the channel. cant seem to get this to work i just get a ingage tone. on the extension that made the call, any help welcom |
19:56.20 | doolittlework | e |
19:57.53 | wcselby | doolittlework - please pb the relevant parts of your extensions.conf and also the cli output of a call that exhibits what you're talking about |
19:57.59 | wcselby | ~pb |
19:57.59 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
19:57.59 | dustybin | in the polycom SETUP menu, i am following DHCP MENU > Boot Server: Static |
19:57.59 | [TK]D-Fender | doolittlework: show me the FAILURE |
19:58.20 | dustybin | there is also a option for: IP Gateway ? |
19:58.43 | [TK]D-Fender | dustybin: Not DHCP.... |
19:59.01 | dustybin | this menu is confusing to navigate |
19:59.11 | *** join/#asterisk ebroad (n=EB@72.11.213.195) |
20:00.10 | dustybin | [TK]D-Fender: i need DHCP enabled so the phone is given a IP address? |
20:00.28 | *** join/#asterisk el_critter (n=critter@190.78.48.45) |
20:00.34 | p3nguin | makes sense to me |
20:00.34 | dustybin | the first option is: DHCP Client: Enabled |
20:00.41 | dustybin | if i press DOWN |
20:00.48 | dustybin | DHCP Menu |
20:00.53 | SuPrSluG | any way to tone down a moh file in musiconhold.conf ? the files are wav |
20:00.56 | dustybin | down again |
20:00.59 | dustybin | IP Gateway |
20:01.04 | dustybin | 00.00.00.00 |
20:01.11 | dustybin | down again |
20:01.14 | dustybin | Server Menu |
20:01.16 | SuPrSluG | similar to quietmp3 |
20:01.31 | p3nguin | If you're getting an IP address via DHCP, wouldn't that configuration contain the gateway setting? |
20:01.52 | dustybin | p3nguin: yes it should |
20:02.10 | dustybin | i am entering the Server Menu |
20:02.17 | dustybin | Server type: FTP |
20:02.31 | dustybin | Adress: = my server IP |
20:03.00 | wcselby | http://austin.craigslist.org/cpg/1377692273.html <--- lol |
20:03.07 | [TK]D-Fender | SuPrSluG: Either use mpg123, or resample them yourself |
20:03.51 | SuPrSluG | k, thanks. |
20:04.11 | maour | i want to present somewhere about ip/pbx , anyone has any document/link/pdf/slide !!? |
20:05.02 | [TK]D-Fender | maour: www.google.com |
20:05.10 | [TK]D-Fender | ~wikis |
20:05.11 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
20:05.11 | ayeso | heheh |
20:05.13 | [TK]D-Fender | ^ |
20:05.16 | maour | :) |
20:05.30 | maour | i mean some thing like a presentation |
20:05.50 | maour | i googled a lot ! nothing usefull |
20:06.32 | Corydon76-dig | Why don't you write your own? |
20:06.43 | [TK]D-Fender | maour: http://www.google.ca/#hl=en&q=asterisk+pbx+presentation+powerpoint&meta=&fp=b664983305b547ce |
20:06.43 | Corydon76-dig | That's what the rest of us have to do |
20:07.04 | maour | Hm! :D |
20:07.15 | dustybin | would the polycom give the same message: could not contact boot server, even if it meant there was something wrong with the .cfg files? |
20:07.23 | doolittlework | wcselby:http://pastebin.com/m4c789f44 |
20:07.51 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
20:08.11 | Corydon76-dig | I don't relish the idea of somebody else taking my presentation that I worked hard to create, changing only the author name, and presenting it as their own |
20:08.51 | [TK]D-Fender | dustybin: has nothig to do with the cfg's |
20:08.52 | wcselby | doolittlework - so once the called party hangs up, you go to 4,Hangup.....? show me a log from the cli of a call doing what you don't think it should be doing.... |
20:09.23 | doolittlework | the first one works records the file but http://pastebin.com/m7cf6151 this i swhat i did to see if the calls go to next step in dialplan |
20:09.39 | dustybin | [TK]D-Fender: what im trying to say is, if the phone did log into my ftp server, however, it couldnt find a config, would it give a error message saying: could not locate config ? |
20:09.44 | Naikrovek | SuPrSluG: you can use sox to normalize to a lower volume i believe |
20:09.56 | doolittlework | i should hear the vm-busy message but i just get ingage tone |
20:09.58 | wcselby | dustybin - check your /var/log/vsftpd.log |
20:10.03 | Naikrovek | SuPrSluG: if it supports mp3 on your system. otherwise use lame probably |
20:10.04 | [TK]D-Fender | dustybin: check your perms,e tx |
20:10.11 | ayeso | Using asterisk with SIP for signaling. Can I use a separate interface for media then the one used for signaling? |
20:10.12 | wcselby | doolittlework - i still need to see the cli output from during a call |
20:10.26 | dustybin | vsftp is not logging for some reason |
20:10.27 | SuPrSluG | Naikrovek: mp3 = too much overhead. |
20:10.36 | wcselby | dustybin - is it running? |
20:10.38 | Naikrovek | oh sorry misread |
20:10.42 | dustybin | ohh it is!!!!!!!!! |
20:10.42 | Naikrovek | yes sox will do it for you |
20:11.15 | wcselby | dustybin - from a computer on the same network as your phone, try to login to the ftp server using the same credentials you're putting in for the phone. |
20:11.15 | [TK]D-Fender | ayeso: No |
20:11.29 | doolittlework | k i just setting up putty sesion to box |
20:11.52 | SuPrSluG | actually I called and it sounds fine. they must be whiners |
20:12.01 | ayeso | [TK]D-Fender: didn't think so... |
20:12.31 | wcselby | SuPrSluG - I had to resample some wav files for a client once, they said it was too loud. sounded fine to me, they wanted it lower so it went lower. |
20:12.38 | dustybin | my polycom has made communication with VSFTP :D |
20:12.50 | wcselby | SuPrSluG - then they said it was too low, so I had to go back up. |
20:12.50 | dustybin | my god, its left some new files behind |
20:12.56 | wcselby | dustybin - lol |
20:12.58 | [TK]D-Fender | dustybin: So far so good |
20:13.10 | wcselby | dustybin - that's what polycom's do |
20:13.27 | Naikrovek | dustybin: that's how you check the phone log |
20:13.31 | Naikrovek | it uploads the log to the ftp server |
20:13.40 | Naikrovek | as well as any contacts (which i think is what started all this) |
20:13.47 | SuPrSluG | they can send their own files if they want it. this is multi-tennant |
20:14.27 | dustybin | i can see the problem now |
20:14.36 | dustybin | its full of FAIL |
20:14.43 | Naikrovek | whoa there, cowboy |
20:14.48 | Naikrovek | you smacktalkin' my polycoms |
20:14.49 | wcselby | haha |
20:14.50 | Naikrovek | ? |
20:15.35 | SuPrSluG | dustybin: <mac>.cfg tells polycom phones where to find sip.cfg and phone.cfg files and directory.xml if desired. |
20:15.53 | dustybin | my configs are named wrong |
20:15.58 | dustybin | i should be using -directory.xml |
20:16.08 | dustybin | -license.cfg |
20:16.29 | dustybin | bootrom.ld' FAILED on attempt 1 (addr 1 of 1) |
20:16.35 | dustybin | im lacking a good howto |
20:16.40 | dustybin | again, google was wrong |
20:16.43 | Naikrovek | dustybin: we are your howto |
20:16.45 | SuPrSluG | if defined in <mac>.cfg it doesn't matter what you call em |
20:16.54 | Naikrovek | well the filenames for the firmware matter |
20:17.18 | dustybin | do i need to download something from here: |
20:17.19 | dustybin | http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip330_320.html |
20:17.24 | dustybin | even though i have a 321 |
20:17.37 | [TK]D-Fender | dustybin: What did you extract into your provisioning folder? |
20:17.39 | Naikrovek | dustybin: download the split VVX SIP firmware and bootrom for your model phone, just dump those files into the root directory of your ftp server (the directory where everything else is) |
20:18.09 | dustybin | [TK]D-Fender: not a lot at the moment, just some rough configs, i will delete them now |
20:18.38 | SuPrSluG | dustybin: a mac.cfg should look like http://pastebin.com/m687e0a95 |
20:18.40 | [TK]D-Fender | dustybin: Sounds like a good reason not to find firmware there... |
20:18.52 | dustybin | eeek |
20:18.56 | [TK]D-Fender | dustybin: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
20:19.36 | SuPrSluG | 321's have the lastest firmware no need to upgrade firmware |
20:19.37 | [TK]D-Fender | dustybin: Go download and extract http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_vvx_3_1_3RevC_release_sig_combined.zip |
20:19.51 | [TK]D-Fender | dustybin: unless your phones hav 3.2.0 |
20:19.51 | dustybin | i will wget it into the ftp root |
20:20.06 | SuPrSluG | 3.2.0 is out? |
20:20.21 | [TK]D-Fender | SuPrSluG: last month |
20:21.01 | dustybin | [TK]D-Fender: im not sure if my phone is 3.1 or 3.2 |
20:21.31 | [TK]D-Fender | dustybin: menu>status>platform>application>main |
20:21.33 | SuPrSluG | menu -> status -> platform |
20:21.59 | dustybin | 3.1.3.0507 |
20:22.28 | dustybin | that is a big .zip file |
20:22.37 | [TK]D-Fender | dustybin: dl the fil I gave you and extract to your provisioning folder. |
20:22.42 | [TK]D-Fender | dustybin: and we start from there |
20:22.54 | dustybin | ok :D |
20:23.50 | doolittlework | wcselby: just bare with me my networkcard on server is a bit dodgy |
20:24.17 | wcselby | doolittlework - k, but I'm only here another 20-30 minutes |
20:24.20 | dustybin | ok its unzip |
20:24.56 | dustybin | images of jellyfish |
20:24.57 | Kobaz | how do i 'lock' CALLERID(name)... i accept a call, and then do a bunch of processing... by the time i actually send the call to a phone via Dial... the callerid name get's clobbered by something else |
20:24.58 | dustybin | strange |
20:25.07 | Kobaz | this is for inbound calls on dahdi via t1 |
20:25.26 | wcselby | Kobaz - are you messing with CALLERID(name) at all during processing? |
20:26.21 | Kobaz | wcselby: yes, i set it |
20:26.37 | Kobaz | i force callerid name depending on some factors |
20:26.41 | Kobaz | but i set the name |
20:26.50 | Kobaz | and then dial the phone... and then the phone gets Unknown |
20:26.58 | wcselby | hmmm |
20:27.01 | Kobaz | it only happens when i get a call with a blank callerid name |
20:27.07 | wcselby | pb relevant parts of your extensions.conf |
20:27.10 | *** join/#asterisk s519 (n=steve@87-194-151-213.bethere.co.uk) |
20:27.14 | dustybin | -r--r--r-- 1 polycom polycom 68467899 2009-05-27 23:00 sip.ld <-- is this the firmware |
20:27.21 | wcselby | well if the callerid(name) is blank, then what's wrong with Unknown? |
20:27.57 | SuPrSluG | dustybin: yes |
20:28.29 | wcselby | pb relevant parts of your extensions.conf |
20:28.38 | Naikrovek | dustybin: if you downloaded the combined firmware |
20:28.40 | SuPrSluG | there are also sip.ld for each phone model |
20:28.43 | Naikrovek | dustybin: split works better |
20:28.48 | *** part/#asterisk korihor (n=korihor@190.77.83.180) |
20:28.51 | Kobaz | http://pastebin.ca/1568840 |
20:29.08 | Kobaz | wcselby: the name has to be the name i set it to be... not unknown |
20:29.10 | Kobaz | i have a name |
20:29.19 | Kobaz | i set the name in CALLERID(name)... but it doesn't 'take' |
20:29.28 | Kobaz | and it happens only when i get an empty callerid name |
20:29.39 | p3nguin | How would I go about telling Cisco 7900 series phones to pull files from a subdirectory of the tftpd rather than the root of it? |
20:30.06 | Naikrovek | p3nguin: not sure you can |
20:30.45 | [TK]D-Fender | Kobaz: no quotes |
20:30.55 | Kobaz | [TK]D-Fender: k |
20:31.21 | wcselby | Kobaz - let me see your extensions.conf where you're setting it |
20:31.58 | wcselby | but [TK]D-Fender is probably right |
20:32.21 | bpgoldsb | 'MYSQL(Fetch fetchid ${resultid} var1 var2 ... varN) -- Fetches a single row from a result set contained in ${result_identifier}. Assigns returned fields to ${var1} ... ${varn}. ${fetchid} is set TRUEif additional rows exist in result set.' |
20:32.32 | Kobaz | http://pastebin.ca/1568845 |
20:32.36 | wcselby | p3nguin - i don't think you can |
20:32.47 | bpgoldsb | I'm getting fetchid set to true, even though I'm only getting 1 row back. shouldn't that not be the case? |
20:33.16 | jaytee | Troy Dale West of Poulan, GA is a mullet-wearing brain dead inbred redneck racist piece of shit. Just had to get that off my chest. |
20:33.33 | Kobaz | [TK]D-Fender: i've never had a problem using quotes |
20:33.40 | Kobaz | [TK]D-Fender: i don't think the quotes are the problem |
20:33.43 | p3nguin | I seem to recall seeing someone's post on some site about using "local/<configs>" rather than the root, but I don't know what to search for to find the correct info about doing it. |
20:33.55 | [TK]D-Fender | Kobaz: you no longer DIAL or do anything else productive in that |
20:34.05 | [TK]D-Fender | Kobaz: and I don't see the FAILED ATTEMPT with back |
20:34.06 | Kobaz | [TK]D-Fender: that's later |
20:34.07 | [TK]D-Fender | up |
20:34.08 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
20:34.13 | Kobaz | [TK]D-Fender: it's not a failed attempt |
20:34.23 | [TK]D-Fender | Kobaz: I know. What aren't I seeing it? |
20:34.26 | Kobaz | [TK]D-Fender: like... the dial is successful |
20:34.27 | [TK]D-Fender | Why* |
20:34.30 | Kobaz | heh |
20:34.39 | Kobaz | find fine... it's not useful |
20:35.02 | jaytee | quittin time |
20:37.32 | wcselby | who the hell is Troy Dale West? |
20:37.54 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
20:38.25 | Naikrovek | who the hell cares? |
20:38.36 | wcselby | evidently jaytee does |
20:38.51 | SuPrSluG | kanye's forgotten twin |
20:39.13 | Naikrovek | twin? |
20:39.20 | Naikrovek | dual douchebaggery? |
20:39.20 | wcselby | http://www.ajc.com/news/clayton/army-reservist-beaten-in-138917.html?imw=Y |
20:39.22 | wcselby | ahh |
20:40.02 | Naikrovek | oh what an ass, a man, beating a woman, yelling slurs, in front of her daughter |
20:40.10 | wcselby | yeah, i agree |
20:40.15 | Naikrovek | there are fewer things you can do that are more traumatic to a child |
20:40.19 | Naikrovek | very few |
20:41.20 | Naikrovek | in my utopia people like that are shot on site |
20:41.31 | Kobaz | [TK]D-Fender, wcselby : http://pastebin.ca/1568858 |
20:42.58 | [TK]D-Fender | Kobaz: where do i see you NoOp-ing it after having set it? |
20:43.08 | Kobaz | hmm |
20:43.10 | Kobaz | intersting |
20:43.12 | Kobaz | i'll try that |
20:44.13 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
20:44.22 | [TK]D-Fender | reaches for his ClueBat (tm) |
20:44.42 | AndyML | so - can someone have a conversation with me about the differences in the different 1.6.X releases? |
20:45.02 | [TK]D-Fender | ~asteriskversioning |
20:45.02 | infobot | asteriskversioning is, like, Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
20:45.02 | dustybin | IT WORKED!!!!!!!!!!!!!!!!!! |
20:45.06 | [TK]D-Fender | ^^^^^^^^^^ |
20:45.17 | AndyML | it looks like there is a 1.6.0.current and 1.6.1.current. - thanks [TK]D-Fender |
20:45.29 | AndyML | are you using either [TK]D-Fender? |
20:45.32 | Kobaz | [TK]D-Fender: heh |
20:45.43 | [TK]D-Fender | andI'm on 1.6.0 at home |
20:45.49 | [TK]D-Fender | AndyML: I'm on 1.6.0 at home |
20:45.53 | Naikrovek | dustybin: gratz |
20:46.02 | AndyML | k. i'll read this and come back. tnx |
20:46.06 | Kobaz | [TK]D-Fender: i was having issues before... i had to insert a Wait() when processing the call...because i would answer the call... and then pri q931 callerid name would come in, and clobber the name |
20:46.13 | dustybin | THANKS!! Naikrovek your configs work perfectly |
20:46.20 | Naikrovek | awesoem |
20:46.26 | [TK]D-Fender | Kobaz: Yes, I have seen some retarded PRI's do this... |
20:46.29 | Naikrovek | awsum |
20:46.36 | dustybin | i now have 2 lines with the same SIP because of the webgui config |
20:46.51 | Naikrovek | dustybin: yeah you can handle two calls on each line as well |
20:46.55 | Naikrovek | polycoms are awesome |
20:47.10 | [TK]D-Fender | dustybin: delete the "-phone" config and pull the power on the phone hard. |
20:47.24 | [TK]D-Fender | dustybin: that should kill your overrides. |
20:47.26 | Kobaz | [TK]D-Fender: but i think i'm having the same type of issue... when callerid is empty... i get a reset of callerid name in from the pri *after* i set callerid(name) |
20:47.35 | Naikrovek | is away |
20:47.42 | Naikrovek | fake status update |
20:47.43 | Kobaz | that's my assumption... i'll need to turn on pri debug |
20:47.47 | [TK]D-Fender | Kobaz: How much time does your call take to go through processing before that set? |
20:47.56 | dustybin | [TK]D-Fender: ok! |
20:48.25 | doolittlework | wcselby: still there? |
20:48.31 | wcselby | for now |
20:48.46 | Kobaz | [TK]D-Fender: shouldn't be more than a hundred ms |
20:48.54 | Kobaz | [TK]D-Fender: it runs by pretty quick |
20:49.05 | [TK]D-Fender | Kobaz: Time for some artificial delay.. |
20:49.07 | doolittlework | k working fine if the connected calls is Sip, overflows and plays the message |
20:49.20 | Kobaz | [TK]D-Fender: i already have a .5 sec delay on initial call pickup |
20:49.21 | *** join/#asterisk scalex000 (n=chatzill@190.80.201.96) |
20:49.24 | Kobaz | do i need more? |
20:49.27 | doolittlework | wcselby: for some reason it does now wanna work with zap |
20:49.30 | scalex000 | hello |
20:49.38 | [TK]D-Fender | Kobaz: evidence seems to say "yes" |
20:49.42 | Kobaz | heh |
20:49.47 | Kobaz | i even tried 1.5sec |
20:49.49 | Kobaz | i'll try two |
20:49.58 | Kobaz | oh... i know |
20:50.06 | Kobaz | if the callerid is empty... i'll wait 2 more seconds |
20:50.10 | Kobaz | otherwise it works fine |
20:50.13 | wcselby | doolittlework - okay.......? |
20:50.16 | scalex000 | I need to ask something about how polycom sip work and how to make dialplan. |
20:50.20 | doolittlework | wcselby: does the g in the dial command only work with sip2sip calls? |
20:50.29 | p3nguin | naikrovek, wcselby: Cisco says, "Configuration files reside in a TFTP server subdirectory (you can specify the location of this subdirectory with the tftp_cfg_dir parameter)." Now I just need to figure ou where that parameter belongs. :) |
20:51.05 | wcselby | doolittlework - where have you shown me anything about the call that's not working? |
20:51.23 | [TK]D-Fender | wcselby: You learn quick young Padawan... |
20:51.28 | wcselby | doolittlework - as far as I know the g option should work on Zap / dahdi calls |
20:51.42 | [TK]D-Fender | it works REGARDLESS of the called channel |
20:51.48 | wcselby | [TK]D-Fender - the more I help people, the more I understand your pain |
20:51.54 | wcselby | :) |
20:52.05 | [TK]D-Fender | is a well of infinite sorrow... |
20:53.19 | wcselby | doolittlework - until I see what's actually happening, I can't really suggest anything. please paste the cli output (with verbose set to 30 or higher) of a call that's not doing what you think it should be doing. |
20:54.04 | doolittlework | i am batling to get the zap card to work for outbound |
20:54.25 | wcselby | doolittlework - resolve your underlying issues before you try to resolve these easier issues |
20:54.42 | wcselby | dodgy network card, zap card not doing what you want, etc |
20:56.33 | doolittlework | hey i am new to linux, i am trying here, big learning curve if one has been stuck betwwen windows and gates |
20:57.07 | wcselby | may I suggest a new keyword for infobot - ~useful - Useful information to pastebin to help troubleshoot your issues - cli output (with verbose set to 30 or higher); relevant sections of .conf files; relevant DEBUG (SIP, ZAP, etc) info. That'd be a good start anyways |
20:57.48 | wcselby | doolittlework - I understand, but it's difficult to tell if the issues you're having are due to the underlying issues you've mentioned or if they're associated with asterisk |
20:58.05 | wcselby | doolittlework - especially when you won't paste any of the requested information |
20:58.12 | [TK]D-Fender | http://punditkitchen.files.wordpress.com/2009/02/political-pictures-bill-gates-campaign-fail.jpg |
20:58.22 | *** join/#asterisk trebaum (n=trebaum@ip68-8-175-208.sd.sd.cox.net) |
20:58.48 | [TK]D-Fender | wcselby: I have on for that.... |
20:58.53 | [TK]D-Fender | one* |
20:59.09 | wcselby | [TK]D-Fender - share please |
20:59.10 | [TK]D-Fender | ~wmmfpb |
20:59.11 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
20:59.12 | Kobaz | okay |
20:59.19 | wcselby | hahahahaha |
20:59.19 | Kobaz | [TK]D-Fender: i got a culprit call |
20:59.20 | Kobaz | [TK]D-Fender: http://pastebin.ca/1568877 |
20:59.20 | doolittlework | i know i will get my facts straight then come back, if i set the cli to 30 there is a huge load of data to type ,so i will get network sorted and come back, thx for listening and advise |
20:59.32 | wcselby | np |
20:59.37 | wcselby | i'll be back tomorrow, probably |
20:59.39 | Kobaz | [TK]D-Fender: no callerid... and i SET callerid(name)... and then on the phone it's unknown |
20:59.50 | *** join/#asterisk ibercom (i=d9d85043@gateway/web/freenode/x-szjxljykkvapmpyp) |
21:00.06 | wcselby | time to hit the road, night all |
21:00.17 | Kobaz | [Sep 16 16:52:47] VERBOSE[20007] logger.c: -- Executing [s@handleIncomingCall:30] NoOp("DAHDI/1-1", "DEBUG: 1253134364.11925 Callerid Name: TEST Aseracare") in new stack |
21:00.29 | Kobaz | that's my noop of the callerid(name) |
21:00.34 | Kobaz | before i send it to the dialer |
21:01.31 | trebaum | Does anyone here have any experience working with Colt Telecom in the NL? |
21:01.52 | [TK]D-Fender | Kobaz: I don't see the call that goes out after that |
21:02.32 | Kobaz | [TK]D-Fender: i got the middle of a previous call in there... |
21:03.00 | dustybin | ever since i deleted all the information on the webgui, i have lost my extensions!! |
21:03.06 | [TK]D-Fender | Kobaz: Never any complete evidence... |
21:03.21 | [TK]D-Fender | dustybin: Excellent, now you can do them right... |
21:04.02 | dustybin | yes, they were in the wrong place |
21:04.03 | Kobaz | [TK]D-Fender: heh... it's really hard to get a nice debug here... there's a bazillion calls coming in at the same time |
21:04.10 | dustybin | i just created a dir called polycom |
21:04.15 | dustybin | inside that is the .cfg |
21:04.23 | dustybin | and other dirs, like contacts, logs, overrides |
21:04.45 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
21:04.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
21:06.00 | doolittlework | can one record 24 fxo on a tdm24xxp card using GenuineIntel Intel(R) Core(TM)2 CPU 6700 @ 2.66GHz with 2 gig memory? |
21:06.40 | [TK]D-Fender | doolittlework: Sure. |
21:07.05 | doolittlework | [TK]D-Fender will the memory be enough? |
21:07.17 | [TK]D-Fender | doolittlework: * doesn't save to RAM you know.. |
21:07.54 | doolittlework | [TK]D-Fender: but surely takes up memory and cpu power to record 24 sim calls |
21:08.57 | [TK]D-Fender | doolittlework: And we had * & T1 card before the P4 was out... |
21:09.21 | doolittlework | what spec was your pc [TK]D-Fender? |
21:09.47 | doolittlework | t1 30 or 20 channels |
21:09.53 | [TK]D-Fender | doolittlework: Its not the spec of the server, its the bandwidth of your pipe :p |
21:09.55 | doolittlework | uk or brit? |
21:10.02 | [TK]D-Fender | UK doesn't have T1 |
21:10.02 | doolittlework | usa or brit? |
21:10.11 | Kobaz | [TK]D-Fender: heh... debugging output from asterisk is so hard to work with... theres no unique identifier for each call for non-dialplan output |
21:10.12 | [TK]D-Fender | doolittlework: C) None of the above |
21:10.13 | doolittlework | ok e1 right |
21:10.25 | doolittlework | where u from [TK]D-Fender? |
21:12.41 | ayeso | Has anyone here ever reported a bug to the development team? Were they responsive? |
21:12.56 | ibercom | Voicemail with IMAP need a lot of connections to imap server. Is it possible minimize the connections number ? |
21:13.42 | ibercom | I have 500 extensions/voicemail users. |
21:13.54 | doolittlework | can one use the Chanspy command to spy on a specific sip extension? |
21:16.10 | [TK]D-Fender | doolittlework: ChanSpy spies on a CHANNEL |
21:17.32 | doolittlework | will extenspy work? |
21:18.31 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
21:18.32 | ayeso | Anyone know anything about the memory leak in the current version of comedian mail? |
21:18.58 | grandpapadot | ayeso: svn? |
21:19.27 | ayeso | Asterisk 1.6.1.1 |
21:22.33 | ayeso | It leaks when calling functions to leave messages, Iv run it through ValGrind, but still analyzing the output.... i have to reboot * every night because of it. |
21:23.04 | dustybin | im unsure what this file does: 0004f2251870-phone.cfg my log is trying to find it |
21:23.54 | ayeso | This is using local storage, I have not tried IMAP |
21:25.15 | [TK]D-Fender | dustybin: that would hold settings manually overridden diectly in the phone |
21:25.18 | *** join/#asterisk ZX81 (n=ZX81@121.74.228.111) |
21:25.40 | [TK]D-Fender | ayeso: 1.6.1.6 = current |
21:26.02 | [TK]D-Fender | ayeso: When you're 5 vers behind, you know what the automatic answer is |
21:26.09 | ayeso | [TK]D-Fender: Grinded 1.6.1.6 as well same issue |
21:26.12 | [TK]D-Fender | doolittlework: both will work. |
21:27.13 | ayeso | [TK]D-Fender: I haven't diffed app_voicemail.c in 1.6.1.6 and 1.6.1.1 but I'm willing to bet there is no change. |
21:28.30 | ZX81 | Was there a fire at Digium? |
21:28.59 | dustybin | ZX81: VIC 20 |
21:29.06 | ZX81 | :) |
21:29.33 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
21:29.55 | dustybin | there seems to be a problem with sip communication to asterisk |
21:30.09 | dustybin | when the phone boots, asterisk doesnt say anything at all |
21:30.12 | dustybin | and i have no extensions |
21:30.13 | ayeso | dustybin: sip set debug on |
21:30.29 | dustybin | ayeso: asterisk isnt the problem |
21:30.36 | dustybin | it was working ok before |
21:30.54 | [TK]D-Fender | dustybin: Because we flushed your reg info <- |
21:30.58 | dustybin | there is some kind of problem with either, server.cfg |
21:31.05 | [TK]D-Fender | dustybin: as I said, its time to do it RIGHT now. |
21:31.27 | dustybin | [TK]D-Fender: i have setup a reg.cfg file with my SIP auth settings |
21:31.39 | dustybin | but the polycom isnt reading it or using it |
21:31.43 | [TK]D-Fender | dustybin: where do I see that that file matters? |
21:32.00 | dustybin | i will paste my configs |
21:33.25 | dustybin | http://paste.debian.net/46718/plain/46718 |
21:35.27 | dustybin | http://paste.debian.net/46719/plain/46719 |
21:37.16 | dustybin | http://paste.debian.net/46720/plain/46720 |
21:37.21 | scalex000 | hello |
21:37.22 | dustybin | those are the 3 important one |
21:37.51 | scalex000 | TK: Can I use ignorepat with polycom sip phone? |
21:39.10 | Kobaz | [TK]D-Fender: http://pastebin.ca/1568921 finnaly got a clean one |
21:39.12 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
21:39.17 | [TK]D-Fender | scalex000: No, only Zaptel/DAHDI FXS |
21:39.21 | Kobaz | [TK]D-Fender: no callerid... i set callerid name,.. and it's unkown kn the phone |
21:39.47 | scalex000 | ok |
21:39.53 | scalex000 | thanks |
21:40.27 | Kobaz | [TK]D-Fender: my debugging shows callerid(name) is what it should be until the end |
21:40.37 | scalex000 | Tk: do you know how to create password before dial international call, any examples |
21:40.58 | [TK]D-Fender | scalex000: "core show application read" , "core show application authenticate" |
21:41.09 | scalex000 | ok |
21:42.06 | Kobaz | [TK]D-Fender: should i answer the call as late as possible? |
21:42.20 | Kobaz | right now i do an answer, and continue to do some processing |
21:42.41 | Kobaz | sometimes i need to play tracks and stuff before sending it to a phone. so i do an answer... but now that i think about it... i can do it at the track spot |
21:43.14 | [TK]D-Fender | Kobaz: I'd delay it 2 sec to start |
21:43.33 | Kobaz | [TK]D-Fender: 2... okay |
21:44.44 | [TK]D-Fender | dustybin: I recommend you trash those configs and start based on the ones int he provisioning pack |
21:45.09 | dustybin | ok |
21:46.13 | dustybin | goes back to the drawing board, this requires a fresh cup of tea |
21:47.19 | dustybin | [TK]D-Fender: the zip file didnt have a example server.cfg |
21:47.34 | [TK]D-Fender | dustybin: that is not a standard file |
21:47.39 | dustybin | ok |
21:48.04 | Kobaz | <PROTECTED> |
21:48.07 | Kobaz | <PROTECTED> |
21:48.10 | Kobaz | same problem |
21:48.17 | Kobaz | [TK]D-Fender: is the provider killing me? |
21:48.32 | Kobaz | or is this an asterisk/config issue |
21:48.41 | [TK]D-Fender | Kobaz: as a test doa long series of consecutive wait/Noop's to conunt the arrivat |
21:48.43 | [TK]D-Fender | l |
21:49.04 | Kobaz | cocout? |
21:49.16 | Kobaz | coconut |
21:49.30 | [TK]D-Fender | count |
21:49.41 | *** join/#asterisk xpot-mobile (n=james@173.8.94.1) |
21:50.44 | Kobaz | i wait 8 seconds.. same problem |
21:50.45 | Kobaz | heh |
21:51.25 | Kobaz | i have some test did's.. i can do some simple tests |
21:51.36 | [TK]D-Fender | Kobaz: Time when it DOES come in. relative to entry / answer |
21:52.17 | Kobaz | k |
21:54.51 | *** part/#asterisk ZX81 (n=ZX81@121.74.228.111) |
21:56.05 | *** join/#asterisk MaliutaLap (i=nikolai@203.39.87.98) |
21:57.02 | *** join/#asterisk jaytee (n=jaytee@unaffiliated/jaytee) |
21:57.17 | *** part/#asterisk ibercom (i=d9d85043@gateway/web/freenode/x-szjxljykkvapmpyp) |
22:00.19 | dustybin | phone1.cfg is a horrible file to edit |
22:00.26 | dustybin | everything is one big wrap |
22:00.45 | p3nguin | eh... is it supposed to be that way? |
22:00.48 | [TK]D-Fender | dustybin: don't like wrap then :) |
22:00.53 | [TK]D-Fender | p3nguin: Yes |
22:01.11 | MaliutaLap | mo'ning Mr 'Fender |
22:01.39 | dustybin | at last!! i have communication!!! |
22:01.41 | doolittlework | what is the variable for noop''ing the sip extension number that inisiates the call? |
22:02.10 | [TK]D-Fender | doolittlework: You can already see that in CLI just by the channel |
22:02.21 | dustybin | my phone says 'John Doe' |
22:02.23 | dustybin | who the hell is that |
22:02.30 | doolittlework | i wanna use the variable in the dialplan |
22:02.42 | [TK]D-Fender | doolittlework: ${CHANNEL} |
22:02.53 | MaliutaLap | dustybin: you've been haxor3D! ;P |
22:03.23 | MaliutaLap | dustybin: probably has something to do with how you configured the SIP, "John Doe" is the example name used in thebook |
22:03.26 | doolittlework | [TK]D-Fender: do you need to set it up in your sip.conf? |
22:03.37 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
22:03.54 | [TK]D-Fender | doolittlework: ${CHANNEL} <-- this is the CHANNEL that is created based on the peer creating the call |
22:05.05 | doolittlework | thanks |
22:06.27 | Kobaz | okay |
22:06.28 | Kobaz | well |
22:06.33 | Kobaz | i simplified everything |
22:06.38 | Kobaz | this is so fscked up |
22:06.55 | Kobaz | http://pastebin.ca/1568952 |
22:07.41 | Kobaz | http://pastebin.ca/1568954 dialplan |
22:08.50 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
22:08.52 | Kobaz | so umm |
22:09.01 | Kobaz | what could possibly be the problem |
22:09.08 | mercutioviz | Dude, did I read that tweet correctly? Was there a fire at digium hq?! |
22:10.20 | Chainsaw | mercutioviz: There was, and everyone is okay. |
22:10.39 | mercutioviz | okay, good |
22:10.46 | mercutioviz | never been there, never seen it, just a pic |
22:10.53 | Chainsaw | mercutioviz: This has turned into an impromptu news article: http://www.venturevoip.com/news.php?rssid=2238 |
22:11.29 | doolittlework | what am i doing wrong with this variable Set(CALLFILE = ${AGENTINFO}-${TIMESTAMP}-${UNIQUEID}), i only get Set("SIP/1000-0a0f3e70", "CALLFILE = --1253138940.31" only the unique id works |
22:13.10 | *** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
22:20.12 | Kobaz | do de do |
22:21.03 | [TK]D-Fender | doolittlework: First I don't know about, second - "core show function CHANNEL" |
22:21.16 | Kobaz | heh so |
22:21.18 | Kobaz | [TK]D-Fender: any idea? |
22:21.31 | [TK]D-Fender | Kobaz: Dunno |
22:21.32 | Kobaz | i hate to be pestering everyone.. but... heh... this is kinda bad |
22:27.25 | MaliutaLap | Kobaz: have I missed something, it seems to be doing what you want |
22:28.15 | Kobaz | MaliutaLap: the phone gets 'Unknown' as the callerid |
22:28.20 | Kobaz | if there is no callerid on the incoming call |
22:28.24 | Kobaz | it's not what i want |
22:28.46 | Kobaz | it's like... if there is no callerid on the incoming call... the callerid is locked |
22:29.42 | MaliutaLap | Kobaz: and you don't have "asrecieved" in the dahdi conf anywhere? |
22:30.25 | Kobaz | nope |
22:32.00 | *** join/#asterisk ebroad (n=EB@72.11.213.195) |
22:32.25 | ebroad | i need a recommendation |
22:32.42 | ebroad | im looking for a good cheap hard phone that supports video |
22:32.55 | ebroad | *hardphone |
22:33.05 | Chainsaw | Good, cheap, video support. Pick any two. |
22:33.06 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
22:33.20 | ebroad | hehe |
22:33.37 | ebroad | lets try cheap and video support |
22:33.39 | MaliutaLap | 's true |
22:34.06 | Chainsaw | ebroad: I'm sure GrandStream will do something resembling a videophone. |
22:34.28 | Chainsaw | GXV-3000 or so. |
22:35.14 | ebroad | looks nice |
22:35.21 | ebroad | anybody using one? |
22:35.37 | Chainsaw | Sure, on your 1970s retro Ikea desk. |
22:35.47 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
22:36.03 | ebroad | eh, this is for home |
22:36.30 | ebroad | i wouldn't roll these out in the enterprise |
22:36.59 | mmlj4 | smart man |
22:37.54 | doolittlework | [TK]D-Fender i am lost again how do i datestamp a monitored file? |
22:38.37 | [TK]D-Fender | doolittlework: "core show function STRFTIME" |
22:39.00 | doolittlework | [TK]D-Fender: is that not to set your system time? |
22:39.48 | [TK]D-Fender | doolittlework: "core show function STRFTIME" <------ |
22:45.31 | *** join/#asterisk pthreadd (n=thread@85.138.26.234) |
22:46.31 | pthreadd | hello everyone |
22:47.14 | pthreadd | im having a problem making a zap trunk to work with a te210p dual-span card |
22:47.22 | pthreadd | i can receive calls |
22:47.38 | pthreadd | but im unable to make calls through any of the PRI links |
22:47.55 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
22:48.35 | pthreadd | and all i can see in documentations is just create a zap trunk with zap channel nr etc etc and everything will be fine |
22:49.04 | pthreadd | but it isn't working anyway |
22:49.20 | pthreadd | if any of you can point me some usefull documentation about this topic |
22:49.41 | [TK]D-Fender | pthreadd: Show us your configs and call attempt |
22:49.42 | [TK]D-Fender | ~pb |
22:49.43 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
22:49.44 | [TK]D-Fender | ^^^^^^^ |
22:50.21 | pthreadd | [TK]D-Fender zaptel and zapata? |
22:50.42 | [TK]D-Fender | pthreadd: Yes, and CLI output of the failed attempt |
22:50.47 | pthreadd | ok |
22:50.51 | pthreadd | just a sec |
22:51.04 | pthreadd | ill give the link in a minute |
22:51.20 | doolittlework | thx [TK]D-Fender: learned something new |
22:53.24 | el_critter | Hi, how can I use dahdi_monitor to determine my PSTN busy pattern? |
22:57.10 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
22:57.38 | *** join/#asterisk garymc (n=garymc@host86-162-159-166.range86-162.btcentralplus.com) |
22:58.25 | garymc | [TK]D-Fender Managed to open them ports on my home router....... it works the bomb now ;) |
22:59.16 | [TK]D-Fender | garymc: And I was sure you'd need this beaten into your head a few hundred thousand more times... |
22:59.41 | garymc | You where only right :S |
22:59.54 | garymc | I got a few more beatings in me yet ;) |
23:00.03 | garymc | before i go down |
23:00.06 | garymc | :S |
23:00.47 | garymc | Well just wanted to come in and thank you...... Thank you very much for the help :) |
23:01.57 | garymc | Good night |
23:02.01 | *** part/#asterisk garymc (n=garymc@host86-162-159-166.range86-162.btcentralplus.com) |
23:02.04 | [TK]D-Fender | garymc: Good, now hopefully you can move on to newer and more interesting things to break |
23:02.10 | pthreadd | [TK]D-Fender http://pastebin.com/d4a72354b |
23:02.18 | pthreadd | only one PRI is configured at the moment |
23:03.19 | [TK]D-Fender | pthreadd: And you should probably be dialing a GROUP, and not channel 1 fixed directly |
23:03.45 | [TK]D-Fender | pthreadd: "G1" instead of 1 |
23:03.49 | [TK]D-Fender | "1" |
23:03.51 | pthreadd | i've tried it |
23:03.51 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
23:04.00 | pthreadd | let me try again |
23:04.05 | pthreadd | maybe i've used g0 |
23:04.09 | pthreadd | instead of g1 |
23:04.12 | pthreadd | sec |
23:04.23 | [TK]D-Fender | pthreadd: correct this, then before your call also pastebin "zap show channels" zap show status" "pri show span 1" |
23:04.38 | pthreadd | ok |
23:07.34 | pthreadd | http://pastebin.com/dd0278b5 |
23:07.37 | pthreadd | there it is |
23:08.51 | [TK]D-Fender | pthreadd: looks pretty good. New call attempt please |
23:09.07 | pthreadd | k.. ill paste it in a minute |
23:12.06 | *** join/#asterisk thegoat (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net) |
23:12.10 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
23:13.23 | pthreadd | http://pastebin.com/mfb59716 |
23:13.42 | *** join/#asterisk raden (n=tanning@69-179-99-17.stat.centurytel.net) |
23:14.01 | raden | http://pastebin.com/m7424f693 <<< can someone tell me if this is vitelity or me |
23:14.06 | raden | im getting very annoyed |
23:16.16 | [TK]D-Fender | pthreadd: "pri debug span 1" , "set verbose 10", try again |
23:16.22 | pthreadd | okey |
23:16.25 | pthreadd | sec |
23:18.54 | raden | [TK]D-Fender, can u traceroute inbound21.vitelity.net or outbound.vitelity.net without error ? |
23:19.17 | *** join/#asterisk propellerhead (n=yogurt2u@host251.200-82-124.telecom.net.ar) |
23:20.26 | doolittlework | pthreadd: have u set your pridialplan and prilocaldialplan? |
23:20.36 | doolittlework | also try overlapdial=yes |
23:21.41 | doolittlework | why would wanna do a traceroute IP if you working with isdn pri, or am i missing the plot? |
23:22.49 | pthreadd | http://pastebin.com/d6c57af0a |
23:23.34 | [TK]D-Fender | pthreadd: Its not even trying and I don't see the debug I should be. |
23:23.39 | [TK]D-Fender | pthreadd: restart * completely |
23:23.42 | [TK]D-Fender | pthreadd: and retry |
23:23.47 | pthreadd | ok |
23:25.18 | pthreadd | doolittlework i think pridialplan is in its default value.. national |
23:25.37 | pthreadd | or maybe not |
23:25.44 | pthreadd | its unknown |
23:25.59 | pthreadd | sec ill restart the asterisk and paste again the debug |
23:27.47 | doolittlework | [TK]D-Fender: can one save Mixmonitor files to custom directories in/var/spool/asterik/monitor folder? |
23:28.47 | [TK]D-Fender | doolittlework: You tell it where to save them.. |
23:30.40 | raden | [TK]D-Fender, can u explain why with vitelity id get a temporary failure in name resalution at the hop nearest them ? |
23:30.50 | [TK]D-Fender | raden: Nope. |
23:30.52 | raden | http://pastebin.com/m7424f693 |
23:31.18 | raden | even doing a route via ip i get the same issue |
23:31.41 | raden | somedays its there and somedays its not |
23:32.05 | pthreadd | [TK]D-Fender the output is similar to the previous one |
23:32.14 | pthreadd | i think it isn't dumping anything new |
23:32.23 | pthreadd | should i increase verbosity? |
23:33.51 | [TK]D-Fender | pthreadd: something is very wrong if it isn't trying to dial at all.... |
23:34.01 | [TK]D-Fender | pthreadd: And you've completely restarted * |
23:34.06 | pthreadd | yes |
23:34.17 | pthreadd | i dont understand this because i can receive calls |
23:34.24 | pthreadd | from the pri |
23:34.42 | pthreadd | i just can make calls |
23:34.51 | pthreadd | can't* |
23:35.14 | [TK]D-Fender | pthreadd: Show me an incoming call |
23:35.22 | pthreadd | ok.. wait |
23:38.32 | *** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk) |
23:40.35 | bmoraca | raden: that name lookup failure is not for the second to last hop, it's FOR VITELITY, and all it means is that their PTR record isn't setup properly |
23:40.53 | bmoraca | raden: the second to last hop doesn't resolve because they probably disabled ICMP on it. |
23:42.34 | pthreadd | [TK]D-Fender http://pastebin.com/d7821f252 |
23:42.45 | p3nguin | A trace to outbound.vitelity.net has every host replying, though. |
23:42.57 | pthreadd | calling, ringing, hangup |
23:43.05 | pthreadd | worked perfectly |
23:43.08 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:43.24 | pthreadd | there's no problems in inbound calls |
23:43.33 | [TK]D-Fender | pthreadd: You are receiving as Zap and dialing as DAHDI. Try dialing as Zap |
23:43.45 | pthreadd | how do i do that? |
23:43.56 | [TK]D-Fender | ~freepbx |
23:43.57 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
23:43.57 | bmoraca | p3nguin: your route to their network may be different than his. mine is also different. i get there in 9 hops and all nodes respond as well. |
23:43.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
23:44.16 | pthreadd | hummm k |
23:44.22 | pthreadd | thanks a lot for your time |
23:44.28 | bmoraca | pthreadd: in amportal.conf, disable DAHDI compatibility mode |
23:44.29 | pthreadd | ill try freepbx now :) |
23:44.33 | p3nguin | Yeah. It takes me 5 hops just to get out of my ISP. |
23:44.40 | pthreadd | bmoraca ok thanks |
23:47.46 | *** part/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
23:48.43 | raden | bmoraca, is that wy i would have registration issues ? |
23:48.54 | bmoraca | raden: probably not, no. |
23:48.59 | raden | p3nguin, is that bad |
23:49.13 | raden | bmoraca, cause i only get that when i have registration issues |
23:49.27 | p3nguin | raden: No. The route from me to vitelity is fine. |
23:49.41 | raden | how about inbound20 and inbound 19 ? |
23:51.07 | bmoraca | raden: the only relevant part of your traceroute is the fact that you were able to get there with reasonable latency and that none of the hops between you and them had issues. what are the problems you're aving? |
23:52.02 | raden | calls not going out or coming in |
23:52.13 | raden | callcentric been up weeks without issue |
23:52.26 | raden | vitelity everytime i turn around unreachable |
23:52.27 | bmoraca | are you registered with vitelity? what does sip debug say? |
23:52.42 | raden | its working again out of the blue |
23:52.50 | raden | next time ill copy everything |
23:53.08 | raden | yeah i registered now but i lost registration for like 20 minutes 2 times today |
23:53.14 | raden | callcentric up all day |
23:53.16 | bmoraca | run sip debug next time it stops working. that'll give a better picture of what's actually happening |
23:53.19 | pthreadd | thanks a lot guys |
23:53.23 | pthreadd | problem solved |
23:53.24 | el_critter | how can I use dahdi_monitor to determine my PSTN busy pattern? |
23:53.25 | doolittlework | [TK]D-Fender: |
23:53.25 | raden | bmoraca, thanks i will :) |
23:53.35 | raden | i have to go later guys |
23:53.50 | pthreadd | [TK]D-Fender special thanks to you.. you were a precious help :) |
23:53.52 | bmoraca | have fun |
23:53.56 | pthreadd | bmoraca thanks for the tip |
23:54.38 | bmoraca | pthreadd: no problem...freepbx defaults that to ON in 2.5 and higher, i think |
23:55.14 | doolittlework | [TK]D-Fender:can one cp a file from one folder to another using an ivr, say for instance when they press one to cp a file from /var/spool/asterisk/monitor to /var/spool/asterisk/monitor/processed? |
23:56.00 | bmoraca | doolittlework: http://www.lmgtfy.com/?q=Asterisk+cmd+System |
23:56.11 | doolittlework | thx |
23:57.27 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
23:59.51 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |