| 00:06.09 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 00:09.36 | Katty | looks in |
| 00:13.25 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
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| 00:14.28 | Katty | anything going on? |
| 00:14.52 | ChannelZ | is back from the dentist |
| 00:14.59 | Katty | how'd that go? |
| 00:15.20 | ChannelZ | I'm $1000 poorer and can't feel the right side of my face, but other than that.. |
| 00:15.28 | ChannelZ | I left work early, so that's worth something I guess |
| 00:15.34 | Katty | on the road to recovery? |
| 00:15.45 | ChannelZ | heh yeah had some fillings |
| 00:15.56 | Katty | were you hurting? |
| 00:16.19 | ChannelZ | Yes and no; Two weeks ago yes, and I went it and found out I have to have my wisdom teeth pulled. GRRR |
| 00:16.23 | *** join/#asterisk fabay (n=fabay@190.50.240.134) |
| 00:16.46 | Katty | that's very normal. |
| 00:17.08 | ChannelZ | but I got some antibiotics and besides one of my back wisdom teeth being a little sore if I bite down on it, I'm OK |
| 00:17.24 | Katty | nods |
| 00:17.33 | Katty | wisdom teeth pain will come and go for several months |
| 00:18.26 | Katty | but if you need them out, you need them out |
| 00:18.38 | Katty | it can cause serious damage if you don' |
| 00:18.39 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:18.39 | ChannelZ | yeah.. only one is actually bothering me but.. |
| 00:18.39 | Katty | don't |
| 00:18.47 | Katty | did he show you the xrays? |
| 00:18.56 | ChannelZ | no I didn't really look |
| 00:19.05 | Katty | ah. |
| 00:19.22 | Katty | well it will be pretty obvious on the xrays if you have issues |
| 00:19.26 | Katty | like side ways wisdom teeth. |
| 00:19.55 | ChannelZ | one of my uppers is actually broken for a long time |
| 00:20.25 | Katty | http://upload.wikimedia.org/wikipedia/commons/d/d0/Impacted_wisdom_teeth.jpg <- take note of toofers on right |
| 00:20.43 | ChannelZ | jesus it's completely horizontal |
| 00:20.51 | Katty | that's actually very common |
| 00:20.58 | Katty | extremely common |
| 00:21.10 | Katty | in fact nearly everyone i know, has their wisdom teeth out, and they were sideways. |
| 00:21.14 | ChannelZ | we've got a design flaw |
| 00:21.17 | Katty | aye. |
| 00:21.18 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 00:21.47 | Katty | something tells me our eating habits used to be a bit different |
| 00:21.47 | Katty | including the size of our mouths! |
| 00:21.55 | *** part/#asterisk fabay (n=fabay@190.50.240.134) |
| 00:21.58 | Katty | more and more often, people are born without wisdom teeth |
| 00:22.25 | Katty | 25% chance now. |
| 00:23.17 | Katty | tho i wouldn't really call it evolution |
| 00:23.27 | Katty | evolution is the weaker dying off before they have a chance to breed |
| 00:23.40 | Katty | and i highly doubt some wisdom teeth are going to keep anyone from uhh...breeding these days. |
| 00:25.31 | drmessano | I have an appendix AND a table of contents |
| 00:25.43 | Katty | i lost my appendix. |
| 00:25.55 | drmessano | Did you rebuild and reindex? |
| 00:26.01 | Katty | no :< |
| 00:26.40 | drmessano | Perhaps the transaction logs would have been useful |
| 00:26.51 | Katty | yes, perhaps )= |
| 00:26.56 | Katty | i must have had logging turned off |
| 00:28.34 | drmessano | A simple replay of the logs could have prevented this. Also your constant gravitation towards NT Backup of *nix based platforms is incorrect, unhealthy, and destructive. Seek help. |
| 00:30.32 | Katty | ^_- |
| 00:30.34 | Katty | nt backup? |
| 00:30.49 | jblack | Katty: I put up a pic of myself. wanna see it? |
| 00:30.58 | Katty | sure (= |
| 00:31.02 | jblack | http://i258.photobucket.com/albums/hh260/lawngnome8273/untitled40.jpg |
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| 00:31.03 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:31.15 | ChannelZ | yikes |
| 00:31.23 | Katty | hahaha |
| 00:31.33 | ChannelZ | Is that Engrish or just irresponsible parenting? |
| 00:31.51 | jblack | dunno. Found that by googling for "fat people" |
| 00:32.04 | jblack | at images, of course |
| 00:32.24 | jblack | it's first if you search for 'fat fucks'. Don't know why... |
| 00:32.55 | jblack | rotfl http://img330.imageshack.us/img330/9974/anorexic3gv.jpg |
| 00:33.00 | jblack | I SO want that tshirt! |
| 00:33.30 | drmessano | Yeah.. "jblack" GIS got me this: http://cdn.kysdc.com/wp-content/uploads//2009/08/ll-cool-j-black-wifebeater.jpg |
| 00:34.01 | ChannelZ | http://adult.engrish.com/2007/07/16/wake-up-call/ |
| 00:34.08 | jblack | damn. busted. |
| 00:35.18 | jblack | Anyone wanna see a picture of "nasty fat granny porn" ? |
| 00:35.41 | ChannelZ | no |
| 00:35.42 | Katty | http://i.imgur.com/SzRNK.jpg <- which one is jblack? |
| 00:36.22 | jblack | I have to pick amongst them? |
| 00:36.41 | Katty | not really. i just thought it was cute. |
| 00:36.45 | jblack | I suppose reddit was closest |
| 00:36.55 | Katty | is reading reddit. |
| 00:37.41 | jblack | my god. why does google index this crap/ |
| 00:38.27 | jblack | feels physically sick now |
| 00:38.41 | jblack | What is reddit anyways |
| 00:38.55 | Katty | it's a bunch of articles people have linked |
| 00:38.59 | Katty | and you up vote it, or down vote it |
| 00:39.05 | Katty | so the most popular topics are at the top... |
| 00:40.06 | drmessano | Its like Digg without being Digg |
| 00:40.46 | *** join/#asterisk chendy (n=chatzill@113.91.37.179) |
| 00:41.50 | jblack | http://cdn.holytaco.com/www/ is cool |
| 00:42.18 | ian6 | ... 404? |
| 00:42.21 | ian6 | realcool :P |
| 00:43.21 | Katty | http://d.yimg.com/a/p/afp/20091020/capt.photo_1256082525540-1-0.jpg |
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| 00:54.05 | jblack | http://www.youtube.com/watch?v=mScGC7nFDxM&feature=topvideos |
| 00:54.14 | jblack | I thought this was a joke, but I'm not sure |
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| 01:41.24 | Carlos_PHX | I'm really impressed with Laptop Magazine's thorough understanding of networking and VoIP: |
| 01:41.26 | Carlos_PHX | "Latency, known as "packet loss" in benchmarking terms, almost disappeared on a Vonage VoIP account." |
| 01:42.17 | trogs | haha |
| 01:44.07 | Carlos_PHX | This is why sometimes VoIP sounds bad, the data is squiggly: http://www.hawkingtech.com/downloads.php?type=diagram&file_id=3175 |
| 01:46.36 | drmessano | lol |
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| 01:51.36 | Ardnat | eloo |
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| 02:12.27 | trogs | some of the data now bypases the modem, i see! |
| 02:43.44 | p3nguin_ | carlos_phx: That's good to know. All this time, I thought latency and packet loss were totally different things. Good thing I can rely on Laptop Magazine to set me straight! |
| 02:44.19 | *** join/#asterisk kyleh (n=kyleh@newguy1.cns.ksu.edu) |
| 02:44.32 | kyleh | hello |
| 02:46.35 | p3nguin_ | I got to thinking, though. Maybe it's different on laptops than it is on enterprise class servers and other commercial telephony equipment. |
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| 03:04.22 | MaliutaLap | if I want to use more than one soundfile (one after the other) in a loop, and as a background ... what is the best way? |
| 03:05.25 | Corydon76-dig | As a background? MOH |
| 03:06.01 | MaliutaLap | Corydon76-dig: as in Background() type stuff |
| 03:06.01 | Corydon76-dig | Pretty much the only way to do background sound |
| 03:06.39 | MaliutaLap | so you hear "the number you dialed is no longer in use" but secretly can enter a code to let you get DISA() access |
| 03:07.35 | Corydon76-dig | Well, there's no looping there, either |
| 03:07.54 | Corydon76-dig | But you can chain soundfiles together in Background with the '&' character |
| 03:08.38 | Corydon76-dig | Just remember that as soon as you press ANY digit, Background stops playback |
| 03:08.45 | MaliutaLap | Corydon76-dig: that is part a ... the next part requires it to loop |
| 03:09.02 | Corydon76-dig | Again, there's no looping in Background |
| 03:09.04 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 03:09.06 | MaliutaLap | alternately how long does the congestion recording go for? |
| 03:09.13 | Corydon76-dig | But your next step after Background could be a Goto |
| 03:09.42 | Corydon76-dig | What congestion recording? |
| 03:10.04 | MaliutaLap | Corydon76-dig: that would loop the Background, but wouldn't it prevent it from getting to the DISA() line? |
| 03:10.38 | Corydon76-dig | Nope... once you start entering digits, you move back to the dialplan, to wait for more digits |
| 03:11.06 | MaliutaLap | hmm |
| 03:11.07 | Corydon76-dig | and thus a jump occurs when the extension is complete |
| 03:16.24 | drmessano | Art Garfunkel can hold a note longer than I can. For some reason, this bothers me |
| 03:16.29 | MaliutaLap | thwacks forehead |
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| 03:46.03 | c|oneman | Are there VoIP services that support forwarding SMS messages (when call forwarding is enabled?) |
| 04:02.55 | kyleh | does 1.6 not have the postgres_cdr.sql in contrib/scripts ? |
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| 04:22.05 | c|oneman | Does anyone know if a VoIP provider that supports forwarding text messages |
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| 05:05.27 | joako | I need a SIP provider that can sustain 100+ channels but my ACD is very low and I believe ASR as well. Does anyone know of a carrier that can support this sort of traffic? |
| 05:15.07 | manxpower | joako: Gafachi |
| 05:15.41 | joako | You don't think their would be an issue with low ACD/ASR on Gafachi w/ their standard rates? |
| 05:16.10 | manxpower | joako: As I understand it they provide service to telemarketers, call centers, customer service operations, etc. |
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| 06:01.33 | knctrnl | ok i got some good tips for the DCap last night. Anyone else have any tips for my DCAP exam tomorrow? |
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| 07:11.55 | thisismyname | good morning |
| 07:12.01 | thisismyname | everybody |
| 07:20.11 | angryuser | joako, which country ? |
| 07:23.27 | ChannelZ | yawns |
| 07:25.17 | joako | angryuser: USA |
| 07:25.44 | angryuser | joako, sorry i know only some in europe |
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| 08:40.52 | Polysics | hi all |
| 08:41.03 | Polysics | wow, i managed to make my flash based phone work :-) |
| 08:41.20 | Polysics | now one AMI related thing |
| 08:41.46 | Polysics | i figured out that to properly log stuff, decrease credit, and other work, on a queue call, i need to detect a few AMI events |
| 08:42.33 | Polysics | could i equally do that with AGI? i basically need to detect from whom the call was, to which user it was, and for how long |
| 08:42.58 | Polysics | rest of variables are already set from an earlier AGI |
| 08:44.03 | Get_The_Fish | not sure what you are asking here.... do you have the queue configured to generate the event, and now you need to capture the output of the AMI events? |
| 08:44.27 | Polysics | the queue throws the events to the AMI interface, yes |
| 08:44.50 | Polysics | i was looking into avoiding to have a sort of listening daemon, and doing the actions reactively via AGI |
| 08:45.17 | Get_The_Fish | ah gotcha |
| 08:45.38 | Polysics | i have already built a PoC with PHP that logs events |
| 08:45.50 | Get_The_Fish | great question actually, never thought of that. I have always done the listening daemon approach. |
| 08:45.56 | Polysics | but it has to be a continuously running thing |
| 08:46.05 | Polysics | well, AMI has it good things |
| 08:46.28 | Polysics | but in this particular case i only need some variables which i sort of "know" are already in the channel :-) |
| 08:46.53 | Polysics | i am not monitoring anything here, just logging from who, to who and duration |
| 08:47.04 | Polysics | + the unique channel ID |
| 08:48.32 | Get_The_Fish | dammit now you have me thinking :) |
| 08:48.36 | Polysics | which i have set earlier |
| 08:50.38 | Get_The_Fish | I would assume that you could do that with AMI, just dont know how |
| 08:50.41 | Get_The_Fish | oops AGI |
| 08:51.04 | Polysics | with a queue involved it changes some things |
| 08:51.10 | Get_The_Fish | yeah, it does |
| 08:51.26 | Polysics | the first thing i'll build is a variable dump for the channel |
| 08:51.41 | Polysics | just have to figure out how to get DeadAGI to work with 1.6 |
| 08:51.58 | Get_The_Fish | cause now that you mention it I dont think that you get that info on a channel variable, queue information that is |
| 08:55.15 | *** join/#asterisk Ashura (n=ashura@89.119.206.194) |
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| 08:59.47 | Get_The_Fish | well, check this: http://www.asterisk.org/docs/asterisk/trunk/applications/queue?type=applications&value=Queue |
| 09:00.05 | Get_The_Fish | AGI: Will setup an AGI script to be executed on the calling party's channel once they are connected to a queue member. |
| 09:00.13 | Get_The_Fish | perhaps that will do what you are looking for |
| 09:00.39 | Get_The_Fish | (btw, new asterisk site with docs is awesome, the drop down application and function listing is a lifesaver |
| 09:01.55 | grharry | Hi, I've got a CISCO 7940 which I've updated with the sip firmware and set it up to connect to asterisk however sip show peers shows the cisco peer as "110 (Unspecified) D N A 5060 UNKNOWN" and DEBUG shows "Trying to put 'SIP/2.0 40' onto UDP socket destined for 10.10.10.2:51113" while tcpdump shows "ICMP 10.10.10.2 udp port 51113" ... Any ideas ?? please ??? |
| 09:02.32 | *** join/#asterisk errotan (n=errotan@81.0.115.122) |
| 09:03.03 | grharry | Sorry .... for the typo tcpdump shows "ICMP 10.10.10.2 udp port 51113 unreachable" ... Any ideas ?? |
| 09:03.35 | Get_The_Fish | I assume that the 7940 is 10.10.10.2 right |
| 09:03.42 | grharry | yes |
| 09:03.47 | Get_The_Fish | just checking |
| 09:03.54 | Get_The_Fish | and can you ping the phone? |
| 09:03.57 | grharry | ye |
| 09:03.59 | grharry | s |
| 09:04.14 | Get_The_Fish | whats the phone saying? |
| 09:04.33 | Get_The_Fish | (been way way too long since I dealt with a cisco phone) |
| 09:05.10 | grharry | when I call out it times out and gives the busy signal |
| 09:06.22 | grharry | It seems that asterisk does not authorize the peer cause of the 511113 unreachable udp port. |
| 09:06.26 | Get_The_Fish | have you tried to register with asterisk with a softphone |
| 09:07.13 | grharry | asterisk is ok with a linksys PAP2 device and it woks perfect ... |
| 09:07.28 | grharry | no not softphone |
| 09:07.45 | Get_The_Fish | thats fine, just really wanted to try another device |
| 09:08.07 | Get_The_Fish | ast and the phone are on the same subnet right? |
| 09:08.14 | grharry | yep |
| 09:08.26 | Get_The_Fish | iptables on the ast box? |
| 09:08.35 | grharry | 10.10.10.70 being asterisk and 10.10.10.2 cisco |
| 09:08.36 | grharry | no |
| 09:08.41 | grharry | no iptables |
| 09:08.41 | Get_The_Fish | k |
| 09:09.12 | grharry | is there a way to make this 5111xxx port something more lower ??? |
| 09:09.49 | grharry | It's on the cisco side I am sure !!1 |
| 09:10.01 | grharry | The problem is there . |
| 09:10.07 | Get_The_Fish | I wonder if qualify=yes has anything to do with it... |
| 09:10.21 | grharry | Let me try no |
| 09:10.27 | Get_The_Fish | I doubt it, cause I think qualify just sends SIP NOTIFY packets. |
| 09:11.07 | Get_The_Fish | and you can icmp ping the phone (not from the asterisk app)? |
| 09:12.31 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 09:12.42 | grharry | yes ... from the shell ... by the way qualify=no did nothing |
| 09:13.14 | grharry | "Ignoring SIP message because of retransmit (REGISTER Seqno 202, ours 202)" |
| 09:13.36 | grharry | "Trying to put 'SIP/2.0 40' onto UDP socket destined for 10.10.10.2:51113" |
| 09:13.59 | Get_The_Fish | what version of asterisk? |
| 09:14.22 | grharry | 1.6.1.6 |
| 09:16.31 | Get_The_Fish | have you tried setting the port=5060 in sip.conf? |
| 09:16.52 | grharry | U mean on asterisk right ?? |
| 09:17.20 | Get_The_Fish | right |
| 09:17.51 | Get_The_Fish | also, just out of curiosity, you have allow=0.0.0.0 for the phone (peer) setting in sip.conf for this peer, right? |
| 09:18.27 | grharry | both yes |
| 09:18.46 | Get_The_Fish | try port=5060 for this phone' |
| 09:18.49 | Get_The_Fish | in sip.conf |
| 09:19.30 | *** join/#asterisk puzzled (n=foobar@83.163.53.136) |
| 09:19.57 | grharry | Its already there |
| 09:20.18 | Get_The_Fish | er? |
| 09:20.26 | Get_The_Fish | thats a little wierd... |
| 09:20.35 | grharry | I knos |
| 09:20.43 | grharry | sorry know !! |
| 09:20.47 | Get_The_Fish | are you getting a register packet from the phone? |
| 09:21.15 | grharry | Should I ngrep ?? |
| 09:22.21 | Get_The_Fish | the best thing, in my opinion, is to use a span (or monitor or mirror or whatever stupid nomenclature your switch uses) port on both hosts and wireshark it... but yea, ngrep that |
| 09:22.21 | grharry | " SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 10.10.10.2:5060;" |
| 09:22.30 | Get_The_Fish | username and password?? |
| 09:23.02 | grharry | all set |
| 09:23.10 | Get_The_Fish | cause that is sip getting to asterisk, ast telling it to f#ck off |
| 09:23.24 | grharry | :-) |
| 09:23.58 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:24.25 | grharry | and cisco insists "REGISTER sip:10.10.10.70 SIP/2.0..Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z...." |
| 09:25.26 | Get_The_Fish | "SIP/2.0 401 Unauthorized" thats the issue right there |
| 09:25.52 | Get_The_Fish | I dont think that this is it, but what codecs are you allowing? |
| 09:25.58 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:26.15 | grharry | I have compiled all into asterisk |
| 09:27.30 | Get_The_Fish | yeah, but what are you allowing in sip.conf |
| 09:27.45 | grharry | oh oh oh oh .... hold |
| 09:28.28 | Get_The_Fish | humor me, put allow=all AFTER deny=all in sip.conf for this phone and restart asterisk (its overkill but I wanna be sure) |
| 09:30.36 | *** join/#asterisk puzzled (n=foobar@83.163.53.136) |
| 09:31.09 | *** join/#asterisk shadebob (n=chatzill@41.92.9.214) |
| 09:31.15 | shadebob | hi all |
| 09:31.24 | grharry | same sh&t |
| 09:31.46 | Get_The_Fish | pastebin your sip.conf for me please |
| 09:31.51 | Get_The_Fish | and the ngrep |
| 09:32.05 | Get_The_Fish | sup shadebob |
| 09:32.49 | grharry | I am sure it's this stupid cisco dev let me try to reflash it and I get back latter ... Very much obliged ... I be back in a sec |
| 09:33.55 | Get_The_Fish | I would check the sniffs for the username and password that you see on the wire if possible...and yeah I am not a big fan of anything cisco btw, my personal opinion |
| 09:34.00 | Get_The_Fish | np |
| 09:35.15 | Get_The_Fish | uh, one other thing, is host=dynamic set for this phone in sip.conf? |
| 09:35.53 | Get_The_Fish | I would look at the device's sip.conf config before I did anything else... see if that linksys or even a softphone will register using the same credentials |
| 09:36.22 | shadebob | I have some performance issues with "logoff" cmd with AJAM + Apache (proxypass). http://pastebin.com/d4e4dfe87 . Any way to boost AJAM performance ? |
| 09:36.38 | grharry | ok !! thanks |
| 09:36.45 | Get_The_Fish | anytime |
| 09:36.52 | Get_The_Fish | sorry I couldnt be more help |
| 09:37.00 | grharry | let me first sniff for username and pass ... |
| 09:37.15 | *** join/#asterisk QaDeS (n=mklaus@213.157.13.70) |
| 09:38.01 | Get_The_Fish | shadebob, 1.6 right? |
| 09:38.18 | shadebob | Get_The_Fish : 1.4 |
| 09:39.05 | Get_The_Fish | interesting... what are you using thats popping ajam? |
| 09:39.54 | shadebob | Check some db values, originate, transfer .... |
| 09:40.05 | *** join/#asterisk shinao1 (n=shinao1@41.219.249.104) |
| 09:40.48 | Get_The_Fish | "/callcenter/ol_cc_form2.inc.php?action=isPredictive&campaig" thats really what I was interested in knowing about :) (I run a call center) |
| 09:41.00 | shadebob | before I used manager API with PHP |
| 09:41.09 | shadebob | but AJAM seem to be more "reactive" |
| 09:41.28 | shadebob | isPredictive ;) |
| 09:41.44 | shadebob | isHeadache ! |
| 09:42.05 | Get_The_Fish | not that I know of on AJAM performance. I know 1.6 has this integrated into the asterisk "mini web server" or whatever |
| 09:42.22 | shadebob | 1.4 too. |
| 09:42.34 | Get_The_Fish | didnt know that |
| 09:42.52 | Get_The_Fish | so, confused here... why apache then? |
| 09:45.14 | shadebob | To make some functions ike : http://pastebin.com/m5649b5c5 |
| 09:45.44 | shadebob | Maybe i'm on a bad way ... |
| 09:47.22 | Get_The_Fish | "Maybe i'm on a bad way" I dont understand |
| 09:48.11 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 09:48.13 | shadebob | Sorry for my bad english. AJAM through Apache is not a good thing ? |
| 09:48.56 | Get_The_Fish | no problem... I couldnt say if it is or not, havent tried it....I was just asking... |
| 09:49.34 | shadebob | Are you using AJAM or AMI API ? |
| 09:49.34 | Get_The_Fish | are you seeing the log off event in the asterisk cli? |
| 09:49.37 | Get_The_Fish | AMI |
| 09:49.45 | Get_The_Fish | considering AJAM though |
| 09:49.50 | Get_The_Fish | done some testing with it |
| 09:49.56 | shadebob | before me too ... without problem :) |
| 09:50.30 | shadebob | but with AJAM ... it's an another thing ! And poor documentation ... |
| 09:52.32 | *** join/#asterisk af_ (n=getsmart@88-149-240-204.dynamic.ngi.it) |
| 09:52.38 | Get_The_Fish | I see to remember this with vicidial in the past, buddy of mine ran into the same issue with logoff |
| 09:53.02 | Get_The_Fish | but I think that was AMI |
| 10:19.02 | *** join/#asterisk mazott (n=chatzill@151.95.162.144) |
| 10:20.33 | mazott | hello, can anyone pls confirm read this message? |
| 10:22.09 | Get_The_Fish | mazott, yes |
| 10:23.01 | mazott | Get_The_Fish, thank you - absolute IRC beginner |
| 10:23.04 | *** part/#asterisk Weedle (n=Weedle@ausphreak/Weedle) |
| 10:25.33 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 10:28.11 | mazott | Get_The_Fish, is this a siutable place to ask technical questions about Asterisk? |
| 10:28.52 | kaldemar | yes, just go ahead and ask. |
| 10:30.15 | Gido-E | mazott are you new to opensource? |
| 10:31.42 | mazott | not really, my first Slackware installation was done on Feb 1995, but I'm new to Asterisk |
| 10:32.10 | Get_The_Fish | mazott, this is where you find all the pros |
| 10:32.20 | Get_The_Fish | (me not being one of them :)) |
| 10:32.45 | *** join/#asterisk rgouveia (n=rgouveia@169.89.54.77.rev.vodafone.pt) |
| 10:33.24 | mazott | that probably too conservative |
| 10:35.26 | rgouveia | hi all ... does anyone know if the openvox card a400p has hardware echo cancellation? it seems to be compatible with tdm400p according to the specs |
| 10:36.52 | mazott | Do you know, by any chance, what a Vodafone Station is? I've got to interface it to something able to extendit to a number of IP phone extensions |
| 10:37.56 | kaldemar | rgouveia: no it doesn't. and neither does TDM400P. |
| 10:38.33 | mazott | so I was looking into an Asterisk-based solution to get it done |
| 10:38.37 | kaldemar | TDM410 added HW EC. |
| 10:39.08 | *** join/#asterisk Smiley_Polecat (n=ippon@87-194-132-25.bethere.co.uk) |
| 10:39.24 | Smiley_Polecat | Hello! n00b in the room <--- |
| 10:39.57 | Smiley_Polecat | has anyone got a minute to discuss how I can get some support or find it for myself? |
| 10:40.01 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 10:40.27 | rgouveia | kaldemar: I see. I was asking because I was recommended that software echo could be a problem. is that true? |
| 10:41.22 | kaldemar | rgouveia: yes. you will most likely suffer less with HW cancellation. |
| 10:44.01 | mazott | The Voidafone staion is a CPE capable of placing VOIP calls on the PSTN network through an intranet, using a conventional multifrequency phone connected to the station via RJ11 |
| 10:44.16 | rgouveia | kaldemar: and is that included with asterisk? |
| 10:45.28 | kaldemar | rgouveia: there is software echo cancellation included in asterisk. you can use a 3rd party one too if asterisk's own algorithms don't work out for you, for example oslec. |
| 10:46.27 | kaldemar | mazott: what VoIP protocol(s) does it support? |
| 10:48.06 | rgouveia | kaldemar: thanks for your attention and help, appreciated. |
| 10:48.37 | kaldemar | no problem |
| 10:50.25 | mazott | Kaldemar: that a good question but I think is not needed to know about, since the Vodafone station is a complete black box, let me elaborate slightly more my question |
| 10:51.40 | kaldemar | if you need to interface the box with asterisk, you definitely need to know what protocols it supports. |
| 10:52.20 | Zeeek | asterisk users who have a Google Wave account, there are several waves about asterisk and voip: One is http://tr.im/vucwave |
| 10:54.30 | mazott | kaldemar: yes, assume the following: the typical client get tha station, connect it to the the adsl, connect a multifrequency phone set to the station to the appropriate TJ11 plug and starts to place PSTN calls through a VOIP gateway run by Vodafone in its comms cloud |
| 10:55.15 | mazott | kaldemar, sorry for the typo, not TY11 but RJ11 |
| 10:56.00 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 10:56.14 | mazott | the station is actually VOIP-connected to the gateway and know the voip protocol |
| 10:57.33 | kaldemar | mazott: so the box is an ATA. if you can access its configuration you might be able to replace the whole thing. then you wouldn't have to buy any analog hardware and could use pure VoIP all the way to the provider. |
| 10:59.00 | kaldemar | situation now: analog phone -(analog)-> vodaphone station -(VoIP)-> provider |
| 10:59.49 | kaldemar | your choice #1: phones -(VoIP, analog...)-> asterisk -(VoIP)-> provider |
| 11:00.44 | kaldemar | #2: phones -(VoIP, analog...)-> asterisk -(analog)-> vodaphone station -(VoIP)-> provider |
| 11:01.04 | mazott | kaldemar: I would like to switch that Voip, converted by means of an ATA, to a set of internal IP phones |
| 11:01.11 | kaldemar | in #2 you would need more analog hardware. |
| 11:02.43 | kaldemar | first try to get rid of the vodaphone station and just replace it with asterisk. since the calls go as VoIP to the provider, there's not much sense in going from VoIP to analog and back to VoIP if it can be avoided. |
| 11:03.15 | kaldemar | sure you can buy another ATA between asterisk and the vodaphone station, but it's pretty ugly. |
| 11:03.43 | garymc | anyone know why my polycoms are not showing the Zero at the begining of a call that comes in? |
| 11:04.02 | mazott | Kaldema: agreed, but the black box terminates the provider network and I have no control over it |
| 11:04.30 | kaldemar | garymc: prove that there is a zero they should show. i.e. pastebin a sip debug of a call. |
| 11:04.55 | garymc | yeah ok sorry |
| 11:05.34 | garymc | Ok i call into my pbx with my mobile it begins 07904 but on the phone it shows 7904 |
| 11:05.46 | garymc | just a little confusing to the eye, if you know what i mean |
| 11:05.52 | Chainsaw | And the sip debug? |
| 11:06.53 | mazott | kaldemar: I also agree on the ugly thing, but I'd need a cost effective solution to provide 4 internal extensions to that ATA-converted line, an Asterisk solution could be an overkill? |
| 11:07.38 | Zeeek | I think this is the one for Asterisk: https://wave.google.com/wave/#restored:wave:googlewave.com!w%252B6kYj-3vsH.3 |
| 11:08.34 | garymc | http://pastebin.ca/1639743 |
| 11:09.34 | kaldemar | mazott: don't think so. you can use a pretty cheap pc and get an unbeatable level of flexibility and features. |
| 11:09.49 | garymc | it doesnt show a zero, but to call it i have to put a zero there, im not getting it. Is there a way i can make the zero show? |
| 11:10.22 | mazott | kaldemar, yes i trust that and the reason I'm asking ;-) |
| 11:11.12 | kaldemar | garymc: that's not a sip debug either, but it happened to show the reason. there's no 0 coming in from your pri. |
| 11:12.16 | garymc | so can i add one, or do i just get used to it? |
| 11:12.22 | mazott | so, the flexibility I'm looking for is to manage 4 additional extensions in a way that the can handle the only analog line provided by the ATA device |
| 11:12.24 | kaldemar | garymc: the zero requirement to call out if in your own configuration. asterisk doesn't require anything you don't tell it to. fix your dialplan. if you're still using freepbx, continue in #freepbx please. |
| 11:12.40 | Smiley_Polecat | chatroom: anyone know about IVR systems? me and my mate's business is dying and we could really use some support. we're trying to intergrate an IVR system into our existing setup and we are totally lost. as out business is dying we can't really afford to buy one so i'm spenting nights trying to work it out for myself! |
| 11:12.59 | kaldemar | garymc: with asterisk, you can do whatever you want with the number. freepbx might be keeping you back though. |
| 11:13.54 | Smiley_Polecat | i dont need a walkthrough per say. I just need someone to help me understand the technical aspects of what it's all about and what I might need. I'm having little luck with google and forums :( |
| 11:14.58 | kaldemar | mazott: you can do that with asterisk. only negative thing is that when a call comes in, you don't necessarily know who it is for. |
| 11:15.53 | kaldemar | Smiley_Polecat: your needs really depend on how you want the IVR to be reachable to your clients. |
| 11:16.33 | mazott | Kaldemar: that would not be a problem |
| 11:16.53 | kaldemar | Smiley_Polecat: at minimum, all you need is a PC hardware. |
| 11:17.30 | kaldemar | mazott: just a matter of deciding where to throw the calls when they come in on a line. |
| 11:18.30 | mazott | Kaldemar; even if all 4 extensions already engaged in conversations between them, the incoming call would be place in wait, that what would be the wish |
| 11:19.01 | *** join/#asterisk lftsy (n=lftsy@88.191.80.8) |
| 11:21.28 | mazott | kaldemar: I figure that when the call comes in, all the extensionsare ringing and the first of them to take it is the one put in conversation |
| 11:23.09 | lftsy | Hello, I have 2 customers with Asterisk servers and for outgoing calls, one received "183 Session Progress" with SDP and the other not, what do I have to change to have it all the time? Thanks |
| 11:24.09 | drcarumas | Smiley_Polecat, this a good starting point. http://www.voip-info.org/ |
| 11:25.26 | kaldemar | mazott: the number of calls is limited to the number of RJ11 plugs on the vodaphone station then. |
| 11:25.26 | drcarumas | Smiley_Polecat, it as almost everything about asterisk, functions, variables.... |
| 11:25.35 | kaldemar | drcarumas: if you can spot the parts that are outdated and invalid. :) |
| 11:25.43 | kaldemar | ~book |
| 11:25.44 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:25.55 | kaldemar | Smiley_Polecat: ^^ that is a good read also. |
| 11:26.01 | drcarumas | kaldemar, true |
| 11:26.02 | mazott | Kaldemar, yes, it would be available one external line only, from the perspective of the internal extensions |
| 11:26.42 | kaldemar | mazott: so that's only one call at a time. other incoming tries would get a busy tone. |
| 11:27.34 | mazott | Kaldemar: yes, absolutely |
| 11:27.39 | kaldemar | mazott: if you're not obliged to use the vodaphone station, ditch it and use some VoIP provider instead. you'll get multiple simultaneous calls and don't need any telephony hardware. |
| 11:28.43 | mazott | Kaldemar: agreed but unfortunately the Vodafone station id marketed by Vodafone as a whole package, take it or loose it |
| 11:29.50 | mazott | tha analog ATA adapted line should be treated by Asterisk as PSTN-like line, can this be possible using a FXO card? |
| 11:31.57 | mazott | Kaldemar;... or, ifa FXO intfc not suitable, a FXS one instead? |
| 11:32.52 | kaldemar | mazott: FXO is what you need if you plan to plug in to a phone port. so either an FXO card or another ATA that speaks SIP to asterisk. |
| 11:32.52 | *** join/#asterisk moos3 (n=rgenthne@cpe-76-179-253-210.maine.res.rr.com) |
| 11:33.11 | moos3 | can anyone help set up te220 pci-e card |
| 11:35.17 | moos3 | the box is running cent5-64bit |
| 11:35.32 | moos3 | I have everything installed |
| 11:36.47 | mazott | Kaldemar: So which type, FXS or FXO, should I use to connect the analog plug of the ATA device, in order to get that line routed by Asterisk to the 4 internal extensions contending the line? |
| 11:37.14 | kaldemar | mazott: FXO |
| 11:38.21 | mazott | Kaldemar; all in all, can then Asterisk accomplish this job? |
| 11:39.19 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:39.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:42.13 | kaldemar | mazott: yes. |
| 11:44.46 | Smiley_Polecat | kaldemar: thanks for your response. Right now we have an analogue PBX which goes through the phone line. Am i right in saying that we would need one phone line PER CALL going into the PC handling the IVR if we continue to use this method? |
| 11:45.13 | Smiley_Polecat | --sorry, i mean one INTERNAL phoneline per call |
| 11:45.50 | Smiley_Polecat | and if this is the case it seems we should change our method of recieving calls to VOIP? |
| 11:46.14 | kaldemar | Smiley_Polecat: yes. and yes. |
| 11:46.14 | mazott | Kaldemar: my plan would be then to put FXS ports for the internal analog phones and one FXO port to use the ATA-provided external line. |
| 11:47.14 | Smiley_Polecat | kaldemar, drcarumas: thankyou both. I'll begin my study :) |
| 11:47.17 | kaldemar | mazott: sounds like a plan. however, you could go for VoIP phones too. |
| 11:48.08 | mazott | Suppose also, in my planend setup, an external call coming in and all internal phones already engaged among them in "private" conversations. Would the incoming call be put in a hold state? |
| 11:52.32 | kaldemar | mazott: that's something you decide in your dialplan implementation. by default, asterisk does nothing with a call. but yes, you can put a call in a queue that plays music to the caller. |
| 11:55.00 | *** join/#asterisk shadebob (n=chatzill@41.92.8.83) |
| 11:55.02 | shadebob | hi all |
| 11:55.38 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 11:55.40 | mazott | Kaldemar, yes, that was I meant in my priimitive, with respect of telephony systems, technical terms. I understand then that provision for such queue handling is what Asterisk provides also off its shelf |
| 11:56.24 | kaldemar | mazott: yes it does. |
| 11:57.33 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 11:57.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:58.25 | mazott | Kaldemar, thank you very much for sorting my thoughts |
| 12:02.41 | *** join/#asterisk anonymouz666 (n=anonymou@187.28.37.118) |
| 12:03.56 | shadebob | If I open an AJAM connection with Logon, make some AMI cmd, will AJAM automatically close socket or I MUST logoff with AJAM cmd ? |
| 12:06.39 | garymc | anyone know anything about snom m3 dect phone. I cant seem to get it to connect to my asterisk server |
| 12:07.20 | garymc | my SNOM has an ip of 192.168.1.74 and my asterisk box ip is 192.168.0.29 |
| 12:08.05 | garymc | in the snom settingsmy registra is set to 192.168.0.29 and my outbound proxy is set to 192.168.0.29 |
| 12:08.11 | garymc | is this correct? |
| 12:10.43 | shadebob | garymc : seem to be correct |
| 12:11.00 | garymc | yes but it keeps saying Registrating then goes to error |
| 12:11.05 | garymc | cant suss it out |
| 12:11.35 | shadebob | any output in asterisk CLI ? |
| 12:11.52 | garymc | I have it plugged into my BT business hub in another building. So it gets a IP off the HUB. But the Asterisk box gets an IP from the hub too? |
| 12:12.23 | garymc | I have other phone connected via RJ45 through an ethernet switch. Would that make much differnce? |
| 12:12.24 | shadebob | any output in asterisk CLI ? |
| 12:12.29 | garymc | Nothing |
| 12:12.48 | garymc | the phone fails to connect |
| 12:12.54 | garymc | it wont register |
| 12:13.48 | Gugge | and you have your routing between 192.168.0 and 192.168.1 setup correct ? |
| 12:17.55 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 12:20.48 | *** join/#asterisk zorp75ck (n=zorp75ck@LindaW.otc.psu.edu) |
| 12:21.47 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:24.28 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) |
| 12:27.16 | garymc | not sure, i think so |
| 12:27.22 | garymc | everything else works |
| 12:27.26 | garymc | how would i check that? |
| 12:27.37 | *** join/#asterisk Lanh (n=wibble@thuy.databug.co.uk) |
| 12:28.00 | kaldemar | garymc: try to ping the phone from the asterisk box. |
| 12:28.32 | garymc | by typing "ping 192.168.1.74" ? |
| 12:29.08 | garymc | yeah thats sending info back to me seems to be working |
| 12:30.16 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
| 12:31.31 | kaldemar | garymc: show a sip debug and prove that asterisk isn't getting any of the registration messages. if that's the case, there's something in between blocking the sip traffic. |
| 12:31.44 | Smiley_Polecat | kaldemar: i'm looking at this package supplied by BT (as an existing customer) http://business.bt.com/broadband-and-internet/internet-communication/broadband-voice#overview .............can you advise me if this is what we should be looking at to prepair for an IVR system? -thankyou :))))) |
| 12:31.49 | garymc | should i just do a asterisk -rvvvvvv |
| 12:33.52 | [TK]D-Fender | Smiley_Polecat: To prepare... go install & learn * |
| 12:33.58 | [TK]D-Fender | ~book |
| 12:33.59 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:34.00 | [TK]D-Fender | ^^^^^ |
| 12:34.53 | garymc | I put in the actual static ip to the world and the phone is showing ok now? |
| 12:34.55 | garymc | Weird |
| 12:35.14 | *** join/#asterisk robl^laptop (n=robl@m445336d0.tmodns.net) |
| 12:35.44 | kaldemar | Smiley_Polecat: that's one choice |
| 12:35.57 | *** join/#asterisk anonymouz666 (n=anonymou@187.28.37.118) |
| 12:36.06 | kaldemar | [TK]D-Fender: he's already been given the docs :) |
| 12:36.14 | kaldemar | ~itsp-list |
| 12:36.15 | infobot | itsp-list is probably Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 12:37.24 | Smiley_Polecat | yes thankyou i'm very inexperienced and am doing my best to understand the terminology and the theory behind this stuff. certainly with what we have in place right now an IVR server cannot be used. but thankyou both for your continued support. you are giving us hope here :) |
| 12:38.27 | [TK]D-Fender | Smiley_Polecat: What do you have installed right now? |
| 12:39.14 | *** join/#asterisk qu1ckkkk (i=c636cae2@gateway/web/freenode/x-sxhrrsjkjjpzltbb) |
| 12:43.03 | TSM2 | is it planned for asterisk to have the ability to update the CID when a call transfer has been completed, or is there a config issue with my * |
| 12:43.05 | kaldemar | [TK]D-Fender: nada. he was asking about his options to build an IVR system earlier and is starting to study asterisk. |
| 12:47.55 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 12:49.38 | *** join/#asterisk anonymouz666 (n=anonymou@187.28.37.118) |
| 12:52.29 | [TK]D-Fender | kaldemar: Sure thing... |
| 12:53.30 | Smiley_Polecat | [TK]D-Fender: presently we have a Siemens analog based PBX system that comes thorugh the adsl line. |
| 12:54.17 | [TK]D-Fender | Smiley_Polecat: What does your anaolg PBX have to do with ADSL? |
| 12:54.31 | drcarumas | guys, i'm currently developing an outbound asterisk solution and I'm trying to improve the dialplan i've been developing. I have a lot of while's, and alot of variable handling. Do you think it is best to handle variables, mysql conections, whiles, etc, with AGI scripts than actually using the dialplan itself? Do i get more performance from AGI ? Right now my structure is very scalable i wont go for AGI only if i realy need more control of things. |
| 12:56.22 | [TK]D-Fender | drcarumas: AGI is a larger load on *. Judge the complexity payoff before going that route |
| 12:56.47 | [TK]D-Fender | drcarumas: Odds are if you can do everything you need in dialplan, leave it there |
| 12:57.00 | drcarumas | [TK]D-Fender, i tought that too. :) |
| 12:57.21 | drcarumas | so, you think only should leave dialplan if i cant do something on it? |
| 12:57.45 | [TK]D-Fender | drcarumas: Once you run into a thing or two you really feel should be done outside, then look to move as much of the other complex stuff along-with to average out the load request. |
| 12:57.52 | [TK]D-Fender | drcarumas: Generally, yes |
| 12:57.59 | *** join/#asterisk rgouveia (n=rgouveia@77.54.89.169) |
| 12:58.49 | drcarumas | OK thank you. Just trying to know if i'm on the correct path . |
| 12:58.49 | Smiley_Polecat | [TK]D-Fender, i'm sorry. i could be wronge. we're on adsl. we have BeThere boardband and an PBX system supplied by Opus. In all honesty they are connected somehow but my understanding of it all is so limited. |
| 12:59.30 | Zeeek | The name of [TK]D-Fender was invoked in vain many times at Astricon! |
| 13:00.04 | [TK]D-Fender | IF YOU SEEK AMY! |
| 13:00.08 | [TK]D-Fender | :p |
| 13:00.15 | *** join/#asterisk kazaa_lite (n=msaleem@94-193-98-124.zone7.bethere.co.uk) |
| 13:00.27 | kazaa_lite | hi all... how to reload all xml files in asterisk? |
| 13:00.42 | *** join/#asterisk shadebob (n=chatzill@41.92.28.199) |
| 13:00.50 | [TK]D-Fender | drcarumas: Yes... the road to Hell is indeed paved with good intentions |
| 13:01.01 | [TK]D-Fender | kazaa_lite: * doesn't use XML |
| 13:01.35 | kazaa_lite | ehhhh... soory i mean configuration files |
| 13:01.41 | drcarumas | [TK]D-Fender, AhahaHa! Indeed!!! :P |
| 13:02.00 | [TK]D-Fender | kazaa_lite: Most = "reload" |
| 13:02.01 | Smiley_Polecat | [TK]D-Fender: what i was told by the person that supplies our current PBX system (which allows us to have a few simultanious calls) is that for us to intergrate with an IVR system we would need 1 internal line per caller. as we plan to have up to 100 callers I can see 100 internal lines going into a server being a good idea. |
| 13:02.26 | Smiley_Polecat | cant* |
| 13:02.29 | [TK]D-Fender | kazaa_lite:Voicemail, dahdi, features, etc need to be specificalyl reloaded or a complete restart of * |
| 13:02.42 | kazaa_lite | ahan |
| 13:02.57 | [TK]D-Fender | Smiley_Polecat: I have 100 DID's on my system, and no, you certainly do not require 1 # per user |
| 13:03.10 | [TK]D-Fender | Smiley_Polecat: it is common for certain deplloyments though |
| 13:04.50 | [TK]D-Fender | Smiley_Polecat: I think part of that IVR remark they made is to allow * to sit in fornt and be able to pass the call to a specific user on your existing system |
| 13:05.06 | [TK]D-Fender | Smiley_Polecat: How many simultaneous calls can it currently support to the outside, and how many phones on it? |
| 13:06.17 | Smiley_Polecat | [TK]D-Fender: 4 calls outgoing. 2 simultanious on hold, on 'voicemail' (as described by the operator) |
| 13:07.03 | Smiley_Polecat | we're still in the stoneages here :$ |
| 13:07.06 | [TK]D-Fender | Smiley_Polecat: Hrm... you will want to be 100% certain of its charateristics before getting started... |
| 13:08.13 | Katty | hmm. |
| 13:08.18 | [TK]D-Fender | Katty: Mew. |
| 13:08.29 | Katty | hi |
| 13:08.46 | *** join/#asterisk _ShrikE (n=_ShrikE@74.185.215.60) |
| 13:09.07 | *** join/#asterisk raden (n=chatzill@66-168-15-19.dhcp.stpt.wi.charter.com) |
| 13:09.23 | *** part/#asterisk raden (n=chatzill@66-168-15-19.dhcp.stpt.wi.charter.com) |
| 13:10.06 | Smiley_Polecat | [TK]D-Fender: when someone calls us a maximum of 2 people can listen the to robot giving them options for extentions. once they make a selection they are put thorugh to us. a maximum of 4 people can reach our phones even though we have 5. and whilst 4 people may be on the phone to us; 2 slots are then available on the 'voicemail' where they will listen to opitons |
| 13:11.15 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 13:11.53 | [TK]D-Fender | Smiley_Polecat: OK, youve got a 2-port IVR / VM unit on it. |
| 13:12.14 | [TK]D-Fender | Smiley_Polecat: does each user have a network jack near their phone? |
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| 13:12.35 | *** part/#asterisk grharry (n=root@ppp-94-65-203-98.home.otenet.gr) |
| 13:13.32 | Smiley_Polecat | [TK]D-Fender: everyone has access to ethernet via our router if that's what you mean. but the phones are running on that standard phone line with 2 copper wires. |
| 13:13.54 | [TK]D-Fender | Smiley_Polecat: I jsut asked if you had a jack pretty much next to it.. |
| 13:15.46 | *** join/#asterisk viq_ (n=viq@unaffiliated/viq) |
| 13:16.24 | Smiley_Polecat | [TK]D-Fender: what do you mean by 'jack' (sorry)? the phones have ports for ethernet and there is ethernet cables available. presently i don't think they are connected to the network. |
| 13:16.50 | *** join/#asterisk krunalpatel (n=chatzill@122.169.94.60) |
| 13:17.01 | krunalpatel | hello |
| 13:17.37 | krunalpatel | I am facing an issue with skype for asterisk |
| 13:18.03 | Smiley_Polecat | [TK]D-Fender: forget it i'm a fool. yes we've got network jacks :$ |
| 13:18.21 | Smiley_Polecat | [TK]D-Fender: plenty unused |
| 13:19.31 | Smiley_Polecat | [TK]D-Fender: and yes network jacks on the phones |
| 13:22.15 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 13:22.47 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 13:23.12 | [TK]D-Fender | Smiley_Polecat: OK... while we haven't confirmed this, its extremely likely that you'd require an interface card for * to sit in front of your current PBX as an IVR. with only 5 phones it'd be a lot more effective to scrap your entire previos PBX, and buy 5 phones for use with *. |
| 13:23.14 | *** join/#asterisk brad_mssw (n=brad@216.155.101.90) |
| 13:24.16 | [TK]D-Fender | Smiley_Polecat: Attempting to "integrate" a solution like yours is not usually cost effective and pidgeon-holes you functionality-wise |
| 13:26.57 | Smiley_Polecat | [TK]D-Fender: certainly feeling that way. hopefully my studies will allow me to utilize * for such. if not I was looking at http://business.bt.com/broadband-and-internet/internet-communication/broadband-voice#overview as a means of getting something new to intergrate with an IVR (which i still have to design) |
| 13:27.54 | [TK]D-Fender | Smiley_Polecat: * really isn't that hard... its worth it... |
| 13:28.21 | [TK]D-Fender | Smiley_Polecat: You may even be able to resell your old system to recoup a portion of your upgrade costs |
| 13:31.58 | *** join/#asterisk cxk287 (n=zorp75ck@LindaW.otc.psu.edu) |
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| 13:34.45 | *** join/#asterisk fainsys (n=fainsys@c-98-242-73-30.hsd1.ga.comcast.net) |
| 13:38.12 | fainsys | looing form some hep with Mixmonitor, I think |
| 13:38.55 | fainsys | anyone know where the message "User hit '*1' to record call. " that is generated in the asterisk CLI comes from |
| 13:39.07 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:40.05 | [TK]D-Fender | fainsys: res_features.so |
| 13:41.25 | fainsys | is there a way to inject a command when recording is started or stopped. I want to send a notification to a third party when a recording is happening |
| 13:42.05 | fainsys | i saw that it can be done following a recording as part of MixMonitor |
| 13:42.33 | [TK]D-Fender | fainsys: You'd have to do some pretty serious hacks of the source... |
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| 13:51.49 | Smiley_Polecat | [TK]D-Fender: ok, i think i understand what you're saying. what we should be looking at is a large broadband pipe with a hosted phone number; which links to an IVR system on * on our server. From here it can then, base on the phonecall needs, act as a PBX switch and redirect the calls, if necessary, to an operator within our office. |
| 13:53.19 | [TK]D-Fender | Smiley_Polecat: What are the specs on your ADSL? |
| 13:54.59 | *** join/#asterisk pta200 (n=paolo@goose.specialai.com) |
| 13:55.12 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 13:55.16 | TSM2 | is it planned for asterisk to have the ability to update the CID when a call transfer has been completed, or is there a config issue with my * |
| 13:56.22 | [TK]D-Fender | TSM2: No. |
| 13:56.43 | Smiley_Polecat | [TK]D-Fender: 12megabit down (so about 1.3 megabytes -ish), 1.11 megabit up. and the box i have available i've been told is too weak; is our old fileserver 2.6ghz intel (lol) |
| 13:56.53 | *** part/#asterisk smtx (n=smtx@p50998557.dip0.t-ipconnect.de) |
| 13:56.58 | [TK]D-Fender | Smiley_Polecat: Certainly no need for a bigger connection. |
| 13:57.18 | [TK]D-Fender | Smiley_Polecat: And your fileserver is OVERKILL for *'s needs |
| 13:57.27 | [TK]D-Fender | Smiley_Polecat: You've got a green light to get started |
| 13:57.48 | [TK]D-Fender | Smiley_Polecat: get cracking... |
| 13:58.26 | pta200 | Is there some kind of method to determine the number of audio buffers necessary in a a meeme conference based on the average number of non-Dahdi channels in the conference? |
| 13:58.49 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d24:8246:d470:a9a7) |
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| 14:06.04 | Smiley_Polecat | [TK]D-Fender: thankyou for the time you've spared. i can't express how nice it is to finally know which way is forwards. thanks from dave, hailey and myself :) |
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| 14:11.15 | TSM2 | [TK]D-Fender: is that no its not an issue or no they do not have it planned? |
| 14:12.34 | [TK]D-Fender | tmsits jsut how attended transfers work, and AFAIK there is no plan to add this. there is a question of device support and standards to make it possible |
| 14:15.57 | TSM2 | thats a bit poo, i thought most major manufs supported it, its a major functionality that most PBX systems have and shame that asterisk cant support it, dont know if there is a way to bodge it |
| 14:16.13 | *** part/#asterisk fiddur (n=fiddur@192.121.104.118) |
| 14:16.46 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) |
| 14:17.30 | [TK]D-Fender | TSM2: Who said the blame is all *'s? |
| 14:17.46 | *** join/#asterisk muh-die-kuh (n=hco@muh-die-kuh.de) |
| 14:17.47 | [TK]D-Fender | TSM2: the ENDPOINT has to support such a thing <- 8 talks how many differnt protocols? |
| 14:17.54 | [TK]D-Fender | * |
| 14:21.34 | *** join/#asterisk CrazyTux[w] (n=Administ@216-110-94-230.static.twtelecom.net) |
| 14:22.43 | CrazyTux[w] | Hey guys, currently using SayDigits() to read back a phone number that is "calling", however I want this to be interuptable when they user hits a digit, is this possible with SayDigits, or must I do some kind of loop / playback the audio files manually, each time with Read() ? |
| 14:24.05 | [TK]D-Fender | CrazyTux[w]: latter |
| 14:24.24 | *** join/#asterisk lwh (n=lwh@66.212.183.122.tor.pathcom.com) |
| 14:24.27 | CrazyTux[w] | [TK]D-Fender: ok, thought so :( |
| 14:32.45 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 14:34.18 | TSM2 | i need to prefix all my speed dials so they work with the outbound dialplans i have, we need to press 9 for any outgoing calls, but i need to keep the numbers in the speed dial without the 9 so they can be used for lookup when thoes people call in |
| 14:35.17 | TSM2 | how can i concatinate "Set(SPEEDDIALNUMBER=${DB(sysspeeddials/${ARG1})})" and put a 9 just after the = but every time i do it does not seem to work |
| 14:48.41 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
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| 14:52.52 | *** join/#asterisk moy (n=moy@74.12.134.3) |
| 14:53.06 | [TK]D-Fender | TSM2: and we don't see enough do do anything that'd qualify as "debugging" |
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| 14:57.47 | Katty | http://42ndhealthstreet.blogspot.com/2009/10/nutrient-recommendations-how-what-i-ate.html |
| 14:58.53 | Katty | ^- results from experiment |
| 14:59.24 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:59.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:00.53 | Zeeek | {{{{{{{Katty}}}}}}} |
| 15:01.00 | Zeeek | You weren't at Astricon? |
| 15:07.13 | Katty | no surprise there. |
| 15:09.22 | *** join/#asterisk Get_The_Fish (n=IceChat7@24.8.50.199) |
| 15:09.47 | *** join/#asterisk luca`gervasi (n=ashura@host40-162-dynamic.52-79-r.retail.telecomitalia.it) |
| 15:09.50 | leifmadsen | Katty: omghi! |
| 15:09.52 | luca`gervasi | hello |
| 15:10.08 | luca`gervasi | is anybody using skype4asterisk from digium? |
| 15:10.17 | leifmadsen | I will be later today |
| 15:10.24 | leifmadsen | I just installed my license, and used the beta |
| 15:10.40 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 15:10.42 | Get_The_Fish | does anyone have any more documentation on the new cdr adaptive odbc? Links, etc? Having a hard time finding out more about it... |
| 15:10.51 | leifmadsen | Get_The_Fish: what do you need to know? |
| 15:10.58 | leifmadsen | it's pretty straight forward -- add new column, use new column |
| 15:11.03 | luca`gervasi | i'm unable to let my user login... :'( |
| 15:11.20 | leifmadsen | luca`gervasi: are you using a business account? you need to use the business panel, or whatever they call it |
| 15:11.21 | *** part/#asterisk pta200 (n=paolo@goose.specialai.com) |
| 15:11.42 | luca`gervasi | yes, i created a new account from the business panel and used it |
| 15:11.51 | luca`gervasi | i'm 100% sure is a business account |
| 15:11.59 | leifmadsen | interesting -- not too sure then -- have you restarted asterisk? I think that fixed it for me the time I tried on teh beta |
| 15:12.06 | Katty | Salmon, Tuna, Nuts, Seeds, Spinach, Whole Grains, Cheese, Chicken, Milk, Bananas, Avocado, Rice, Beans, Citrus Fruits, Broccoli, Lean Beef, Peas, Cherries. |
| 15:12.11 | Katty | ^- what i should be eating. |
| 15:12.17 | luca`gervasi | i restarted it many times... |
| 15:12.20 | leifmadsen | Katty: what most of us should be eating |
| 15:12.30 | Get_The_Fish | lief, I was wondering about the filter that I see in the sample conf, what that does exactly. |
| 15:12.30 | Katty | leifmadsen: did you see my post? |
| 15:12.33 | leifmadsen | Katty: I am currently sipping a cup of green tea, and should grab my multivitamin |
| 15:12.35 | leifmadsen | Katty: I did! |
| 15:12.38 | leifmadsen | i didn't read it though |
| 15:12.39 | Katty | leifmadsen: it makes me sad. |
| 15:12.59 | *** join/#asterisk chazzm (n=chazz@173-24-217-9.client.mchsi.com) |
| 15:13.06 | Katty | leifmadsen: 2% of my day's vitamin 3, 21% of vitamin d, no b1 or b2 or b6, 47% of b9 :/ |
| 15:13.16 | Get_The_Fish | populating the corresponding |
| 15:13.16 | Get_The_Fish | ; CDR variables in the dialplan |
| 15:13.18 | Get_The_Fish | oops |
| 15:13.29 | Katty | leifmadsen: not enough b5, or iron, or magnesium, half of the potassium i'm supposed to have. |
| 15:13.30 | Get_The_Fish | populating the corresponding |
| 15:13.30 | Get_The_Fish | ; CDR variables in the dialplan |
| 15:13.44 | Katty | sighs. |
| 15:13.47 | Katty | no wonder i feel awful a lot. |
| 15:14.05 | Katty | needs to get herself some gummy flinstones ;) |
| 15:14.17 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 15:14.23 | leifmadsen | Katty: heh |
| 15:14.29 | *** join/#asterisk DavidR2008 (n=chatzill@208.34.240.2) |
| 15:14.30 | leifmadsen | Katty: I like the Platinum products |
| 15:15.06 | leifmadsen | just took his greens+, fish oil, and multivitamin |
| 15:15.06 | CrazyTux[w] | [TK]D-Fender: ok just implemented a loop to handle it, but it seems "too slow" |
| 15:15.15 | CrazyTux[w] | [TK]D-Fender: is there anyway I can get to progress with in milliseconds and not seconds? |
| 15:15.21 | Zeeek | Katty, no answer? What have I done? |
| 15:15.31 | Katty | Zeeek: i said no surprise there (= |
| 15:15.41 | Katty | leifmadsen: i think i'd rather try to eat my vitamins than take suppliments if at all possible. |
| 15:15.56 | Katty | leifmadsen: it's a sad sad day when you can no longer buy groceries with nutrients, and must consume a pill to get them |
| 15:16.02 | DavidR2008 | had anyone used the ExternalIVR app in ivr:// mode? |
| 15:16.03 | Zeeek | but you didn't acknowledge me which lowered my self sesteem |
| 15:16.05 | leifmadsen | Katty: true, but sometimes its not possible to eat all the things you should in one day :) |
| 15:16.10 | Katty | Zeeek: :< |
| 15:16.13 | Katty | hugs Zeeek |
| 15:16.18 | Zeeek | I HUGGED you, Katty and you were cold to me :( |
| 15:16.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:16.24 | Zeeek | bigger hug! |
| 15:16.27 | Katty | Zeeek: i am very sorry i did not awknowlege you. i did not even see the hug :< |
| 15:16.31 | leifmadsen | Katty: you can buy the groceries to get your nutrients, you just have to buy the right things |
| 15:16.34 | Katty | hugs Zeeek, just for good measure. |
| 15:16.38 | Zeeek | ty |
| 15:16.49 | Zeeek | hugs leifmadsen |
| 15:16.50 | Katty | Zeeek: no, i did not go to astricon. |
| 15:17.08 | Katty | Zeeek: we don't actually sell many asterisk based pbxes anymore. ever. |
| 15:17.15 | Zeeek | so that wasn't you I was hitting on drunk at 4AM in the bar? |
| 15:17.22 | Katty | leifmadsen: i agree with not being able to eat everything you need in 1 day. |
| 15:17.22 | Zeeek | good! |
| 15:17.41 | leifmadsen | Katty: the multivitamin isn't a substitute, it's just a helper |
| 15:18.30 | Zeeek | speaking of multivitamins, the Vitamin Users COnference starts in about 30 minutes in wideband at 200901@login.zipdx.com |
| 15:18.41 | Chainsaw | The ChanIsAvail command, is that still available in Asterisk 1.6? |
| 15:18.45 | leifmadsen | Zeeek: how early can I join?! |
| 15:18.55 | Zeeek | in 30 minutes |
| 15:18.56 | leifmadsen | I feel like I still have a vitamin stuck in my esophogus |
| 15:18.58 | Katty | Zeeek: i was asleep last night at 4am. |
| 15:19.02 | leifmadsen | Zeeek: OK! |
| 15:19.16 | Katty | Zeeek: next to a particularly warm male. |
| 15:19.23 | Katty | as i do every night. |
| 15:19.23 | [TK]D-Fender | Chainsaw: Yes |
| 15:19.28 | Katty | for the last 3 years. |
| 15:19.29 | Zeeek | Katty I don't use asterisk and I was at Astricon |
| 15:19.37 | Katty | (= |
| 15:19.50 | Zeeek | Katty no not last night, at Astricon |
| 15:20.03 | Zeeek | last night I celebrated my anniversary with my wife |
| 15:20.10 | CrazyTux[w] | Katty: I sleep next to four particularly warm females hah! |
| 15:20.29 | Get_The_Fish | sorry, stuck key :) .... I was wondering what exactly was meant by "populating the corresponding CDR variables in the dialplan"- would that be something to the effect of set(cdr(fieldname)=value), where fieldname=variable? |
| 15:20.47 | CrazyTux[w] | (wait wait, that was only in my dream last night) |
| 15:20.49 | CrazyTux[w] | :) |
| 15:21.10 | Get_The_Fish | "populating the corresponding CDR variables in the dialplan" is in the cdr_adaptive_odbc.conf sample file |
| 15:21.28 | Zeeek | I had a hot blonde under me Sunday night at 4AM. Unfortunately she was about 32,000 feet under me as we flew over Miami |
| 15:21.38 | CrazyTux[w] | LMFAO |
| 15:21.42 | CrazyTux[w] | haha |
| 15:21.53 | Zeeek | she had ther nerve to ask "oooh, that was you?" |
| 15:22.02 | leifmadsen | Get_The_Fish: yes, except CDR() needs to be uppercase because it is a function |
| 15:22.11 | leifmadsen | Get_The_Fish: that's exactly it though |
| 15:22.27 | CrazyTux[w] | really sleeps next to the GF + Son every night in which kicks me in the face |
| 15:22.43 | Zeeek | leifmadsen: please stop flooding the channel with appropriate serious asterisk talk while I'm trying to warm up my comic chops |
| 15:22.48 | leifmadsen | Get_The_Fish: additionally, if you don't need to document a certain field, you can just leave it out of the database schema, and asterisk should automaticalyl adapt to it being missing |
| 15:22.53 | CrazyTux[w] | never did like his crib :) |
| 15:23.03 | leifmadsen | Zeeek: you're not funny, so I need something else to distract me |
| 15:23.13 | Get_The_Fish | ah ok, well sweet... extra super sweet cause that is what I have been wishing asterisk would do for quite a while. |
| 15:23.15 | Zeeek | there goes my self esteem again |
| 15:23.25 | leifmadsen | Zeeek: don't worry, you're hot |
| 15:23.44 | Zeeek | looking for the suicide tablets |
| 15:23.48 | leifmadsen | Zeeek: and by hot, I mean temperature wise |
| 15:24.01 | Chainsaw | [TK]D-Fender: Reason I ask is that testing SIP/41726 seems to succeed while I can't dial there. |
| 15:24.04 | Chainsaw | [Oct 23 16:13:08] VERBOSE[31181] pbx.c: -- Executing [726@from-client:500] GotoIf("SIP/45017-44016948", "ChanIsAvail(SIP/41726)?501:510") in new stack |
| 15:24.04 | Chainsaw | [Oct 23 16:13:08] VERBOSE[31181] pbx.c: -- Goto (from-client,726,501) |
| 15:24.04 | Zeeek | no it's freezing here now. Big change from the 100° of Phoenix |
| 15:24.09 | Get_The_Fish | what can I do to document that? just voip-info.com? (I LOVE the new site btw, the /docs with the applications and functions dropdown is awesome) |
| 15:24.11 | *** join/#asterisk gardo (n=gardo@110.55.225.235) |
| 15:24.17 | leifmadsen | Zeeek: I miss PHX |
| 15:24.25 | Zeeek | I do too, except for the food |
| 15:24.27 | [TK]D-Fender | Chainsaw: APPLICATION, not FUNCTION |
| 15:24.35 | leifmadsen | Zeeek: aye, I can't get good food in the US |
| 15:24.48 | Zeeek | no there's little to eat |
| 15:24.52 | Chainsaw | [TK]D-Fender: Right. Is there a function equivalent? |
| 15:24.58 | leifmadsen | Chainsaw: nope |
| 15:25.48 | Zeeek | applications vs functions |
| 15:25.48 | leifmadsen | fight! fight! figth! |
| 15:25.48 | Zeeek | core show apple |
| 15:25.48 | Zeeek | ore show magnetic |
| 15:25.48 | Zeeek | core show marine |
| 15:25.54 | Chainsaw | Zeeek, we're not your Asterisk CLI. That's overthere *points* |
| 15:26.03 | Zeeek | su |
| 15:26.08 | Zeeek | nh8uhjuyu767YY |
| 15:26.09 | Zeeek | $ |
| 15:26.13 | Zeeek | oh daln |
| 15:26.14 | Zeeek | ` |
| 15:26.27 | Zeeek | well, you'll never know the IP hahaha$ |
| 15:27.01 | CrazyTux[w] | lmfao |
| 15:27.32 | Zeeek | you will be able to LYFAO even more on this channel: #voip-users-conference |
| 15:28.18 | CrazyTux[w] | Zeeek: :) |
| 15:28.21 | Zeeek | We will be discussing Google Wave (as related to Asterisk and VoIP), Jabber as related to Google Wave, Astricon, Google Voice as related to SIP |
| 15:28.40 | Zeeek | Girls are allowed, but they can't speak on the channel |
| 15:29.03 | CrazyTux[w] | lmfao |
| 15:29.34 | CrazyTux[w] | Zeeek: just ugly girls right? what about blonde bombshells :) |
| 15:29.52 | CrazyTux[w] | Zeeek: with school girl outfit and teacher glasses :) |
| 15:29.55 | Zeeek | we don't discriminate by looks, only by sex. |
| 15:29.59 | CrazyTux[w] | lol |
| 15:31.04 | Katty | this tea is gross. |
| 15:31.09 | Katty | :< |
| 15:32.16 | Zeeek | green or regular? |
| 15:33.16 | ChannelZ | Tea makes me constipated. |
| 15:33.20 | ChannelZ | Just thought you should know. |
| 15:33.41 | ChannelZ | Happy Friday everyone! |
| 15:35.11 | Zeeek | no Katty , I was referring to the sex? |
| 15:35.36 | Katty | Zeeek: it was regular. |
| 15:35.44 | Katty | Zeeek: but i threw it out. it was icky |
| 15:36.01 | Zeeek | the sex? |
| 15:36.09 | Katty | huh? |
| 15:36.15 | Katty | you lost me. what does tea have to do with sex. |
| 15:36.25 | Zeeek | Glad you asked! |
| 15:36.31 | Katty | k |
| 15:36.35 | Katty | educate me. |
| 15:36.51 | Zeeek | I would but I need to go raise the conference bridges |
| 15:36.55 | Katty | k |
| 15:37.41 | Zeeek | come by and say hi: #voip-users-conference or call in: http://VUC.me or POTS (567) 252-2286 |
| 15:37.57 | Zeeek | bye all, have a great Leif |
| 15:38.27 | leifmadsen | :) |
| 15:38.55 | Zeeek | Talkshoe not working again! |
| 15:38.59 | Katty | :< |
| 15:39.18 | Katty | leifmadsen: how busy are you |
| 15:39.26 | leifmadsen | Katty: reasonably busy, why? |
| 15:39.31 | Katty | leifmadsen: mkay. |
| 15:39.38 | Katty | leifmadsen: i was going to inlist your help about groceries. |
| 15:39.45 | leifmadsen | Katty: I see -- ask away |
| 15:39.46 | Katty | leifmadsen: but that's okay. |
| 15:39.59 | leifmadsen | Katty: I should be able to respond, although latency may be higher than normal |
| 15:40.16 | *** join/#asterisk robl^laptop (n=robl@208.54.83.68) |
| 15:40.30 | leifmadsen | "We are experiencing higher than normal call volume. Your call has been placed in priority sequence, and will be answered by a service support agent as soon as possible." |
| 15:40.40 | Katty | ha |
| 15:40.53 | Katty | well let me formulate a basic plan, then i will pester you with questions. |
| 15:41.57 | Zeeek | Talkshoe up and waiting for me to call from ZipDX |
| 15:42.21 | Katty | bacon has no nutritional value :< |
| 15:42.27 | Katty | just sodium :< |
| 15:42.33 | leifmadsen | heh :) |
| 15:42.35 | Zeeek | mmmmmm bacon |
| 15:42.41 | Zeeek | I eat that once a year |
| 15:42.43 | Katty | and 1.6mg of niacin |
| 15:42.48 | Katty | pouts |
| 15:43.01 | Zeeek | you are what you heat |
| 15:43.10 | Zeeek | ok, I'll leave you alone |
| 15:43.13 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:44.52 | Katty | leifmadsen: what variety(s) of lettuce would you recommend? |
| 15:45.05 | Katty | leifmadsen: i'm guessing iceburg is useless. |
| 15:45.24 | leifmadsen | Katty: they are all good, but I prefer the visual look of baby romain and spinach |
| 15:45.36 | leifmadsen | almost all lettuce is just water and fiber |
| 15:45.52 | Katty | but surely raw spinach has more nutritional value than iceburg |
| 15:46.12 | leifmadsen | Katty: yes, spinach probably has some additional nutrients over most kinds of lettuce |
| 15:46.34 | leifmadsen | I'd go with the baby romain instead, since it has some colour, and I find iceburg lettuce just looks "cheap" |
| 15:47.00 | leifmadsen | although I like it for doing lettuce wraps (instead of using soft taco wraps, which tend to have a lot of oils) |
| 15:47.24 | Katty | digs through google for a comparrison chart of lettuces. |
| 15:47.32 | titter | so the tour around the digium building was nice ... |
| 15:49.57 | Katty | leifmadsen: romaine is much much better for you than iceburg. |
| 15:50.23 | *** join/#asterisk qxork (n=fred-tea@76-206-40-250.lightspeed.livnmi.sbcglobal.net) |
| 15:50.24 | Naikrovek | yes |
| 15:50.42 | leifmadsen | Katty: nice to know -- I figure the darker the green, the better :) |
| 15:50.50 | Katty | especially in the potassium and vitamin c department. |
| 15:51.06 | Katty | iceberg = 1.5mg vitamin c, romain 11.3, and 13.3 vs 48.2 |
| 15:51.29 | Katty | AHA HA! |
| 15:51.35 | Katty | lettuce chart: http://aggie-horticulture.tamu.edu/plantanswers/recipes/spinach/spinach&lettucenutrition.html |
| 15:52.27 | Katty | it's become clear spinach is probably the best choice here. |
| 15:52.46 | Katty | across the board it just has more of everything |
| 15:52.54 | Naikrovek | best for you, tastes the worst, and i don't think that's a coincidence |
| 15:52.56 | Katty | except water content. |
| 15:53.02 | Katty | i like spinach. |
| 15:53.11 | Katty | a whole lot more than iceberg |
| 15:53.16 | Naikrovek | i like it too, but it tastes terrible in comparison to other lettuce |
| 15:53.23 | Naikrovek | put spinach on a cheeseburger and tell me you like it more |
| 15:53.50 | Katty | okay. i admit that as a added item ona bigger item, spinach may not be suitable |
| 15:53.58 | Katty | but for salads, i still prefer the taste of spianch |
| 15:54.05 | Katty | spinach, fetta cheese, and strawberries |
| 15:54.11 | Naikrovek | interesting |
| 15:54.33 | Katty | Naikrovek: second best choice looks like romaine |
| 15:54.37 | *** join/#asterisk JoeMoretti (n=jmoretti@76.164.171.81) |
| 15:54.42 | Naikrovek | i need to eat something besides turkey & mayo sandwiches |
| 15:54.49 | Naikrovek | romaine is awesome |
| 15:54.58 | Qwell | waves to JoeMoretti |
| 15:55.08 | Katty | it's second highest in potassium. |
| 15:55.21 | Katty | and vitamin c. |
| 15:55.38 | Katty | needs serious help with her potassium levels. |
| 15:55.39 | *** join/#asterisk wcselby (n=wcselby@216-110-88-194.static.twtelecom.net) |
| 15:55.46 | Qwell | throws a banana at kaii |
| 15:55.49 | Qwell | Katty too |
| 15:55.58 | Katty | Qwell: :> those are on my grocery list |
| 15:56.01 | Qwell | kaii: sorry, you were a necessary casualty |
| 15:56.16 | wcselby | Katty - he's saving you some time |
| 15:56.24 | *** join/#asterisk Sajam (n=sajam@beta.intelligile.com) |
| 15:57.14 | Katty | leifmadsen: do you know any citrus fruits that don't make a mess when you eat them? |
| 15:57.44 | leifmadsen | Katty: raspberries? |
| 15:57.49 | leifmadsen | cherries? |
| 15:57.49 | Naikrovek | all fruit make a mess when you eat them |
| 15:58.08 | leifmadsen | not sure how much citrus are in those though |
| 15:58.21 | Katty | Naikrovek: i don't like sticky hands :< |
| 15:58.31 | *** join/#asterisk sun28 (n=light@188.19.27.162) |
| 15:58.36 | Naikrovek | i used to eat rhubarb for vitamin C |
| 15:58.40 | Naikrovek | we had a patch of it in my back yard |
| 15:58.43 | sun28 | morning |
| 15:58.47 | JoeMoretti | waves to Qwell |
| 15:58.49 | Naikrovek | "don't eat the red, or you'll be dead" |
| 15:58.51 | afink | Katty: nectarines aren't too bad |
| 15:58.58 | leifmadsen | Katty: I always slice around my orange and pull off all the skins, then break it all up, then wash my hands before I eat it, because it bugs me too |
| 16:00.12 | Katty | leifmadsen: that might work. |
| 16:00.23 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:00.25 | Katty | leifmadsen: what's your favorite? |
| 16:00.57 | leifmadsen | Katty: I like those larger oranges because I only like to have to open one at a time :) |
| 16:01.11 | Naikrovek | watches dj hero mixes on youtube. WANT |
| 16:01.22 | Katty | k |
| 16:01.49 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-68-31.client.mchsi.com) |
| 16:02.02 | leifmadsen | dj hero, lol |
| 16:02.09 | *** join/#asterisk jtodd (i=w47utxta@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:02.09 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:02.11 | [TK]D-Fender | Naikrovek: http://www.southparkstudios.com/episodes/127947 |
| 16:02.31 | Naikrovek | lol |
| 16:03.15 | Naikrovek | I lol'd at DJ hero until i listened to it. now i want it |
| 16:03.28 | leifmadsen | Naikrovek: nice :) |
| 16:03.42 | leifmadsen | Naikrovek: I'm working on gather enough hardware to do actual digital DJing |
| 16:03.48 | leifmadsen | gathering* |
| 16:03.52 | Katty | leifmadsen: do you think fresh peas would be better than frozen? |
| 16:03.58 | Naikrovek | i used to have several turntables and all of that |
| 16:04.07 | leifmadsen | Katty: if they are flash frozen, it doesn't actually matter |
| 16:04.11 | Naikrovek | but i can't scratch an itch on my own back, much less records |
| 16:04.29 | leifmadsen | Katty: the way you cook them is more important -- just steam them, and never boil |
| 16:04.36 | Katty | hmm. |
| 16:04.48 | Katty | i don't think i have any way to steam them |
| 16:04.51 | leifmadsen | frozen veggies are actually a good way of not having to buy and waste veggies |
| 16:05.04 | Naikrovek | yeah freezing veg doesn't harm them |
| 16:05.19 | leifmadsen | Katty: get one of those steamers, which are those metal things that fold in on themselves, and then just stick it in a pot with some water underneath, and boil it |
| 16:05.26 | Naikrovek | mmm |
| 16:05.31 | Katty | ^_- |
| 16:05.34 | Katty | goes to find an image |
| 16:05.54 | Katty | oh a little metal basket? |
| 16:06.14 | leifmadsen | Katty: http://comps.fotosearch.com/comp/phd/PHD407/broccoli-metal-steamer_~OS49014.jpg |
| 16:06.47 | Naikrovek | 503 |
| 16:06.56 | leifmadsen | interesting |
| 16:07.00 | Naikrovek | 500 i should say |
| 16:07.00 | Katty | i found a picture |
| 16:07.01 | leifmadsen | oh ya... |
| 16:07.07 | leifmadsen | weird :) |
| 16:07.18 | leifmadsen | Katty: but ya, that's what you need -- cost like $5 probably |
| 16:07.20 | Katty | http://fantes.com/images/8332steamers.jpg |
| 16:07.27 | Naikrovek | they look like little foldable satellite dishes |
| 16:07.28 | Katty | is that it? |
| 16:07.33 | Katty | hrmm |
| 16:07.39 | Naikrovek | that's a high-end version |
| 16:07.46 | Naikrovek | but it'll do |
| 16:08.04 | Katty | http://www.johnlewis.com/jl_assets/product/230398163.jpg <- that? |
| 16:08.07 | Naikrovek | Katty: http://www.amazon.com/gp/product/B00062B0K6 |
| 16:08.14 | Naikrovek | Katty: that's it |
| 16:08.14 | *** join/#asterisk sjb_gt (n=sachajbe@71-12-73-207.dhcp.gnvl.sc.charter.com) |
| 16:08.25 | Katty | what do you ummm, do with it |
| 16:08.38 | Naikrovek | put veg inside, set it on top of a pot of boiling water |
| 16:08.44 | Naikrovek | wait, then, steamed veg |
| 16:08.51 | Katty | oooh |
| 16:08.55 | Katty | so it doesn't actually go IN the pan |
| 16:09.00 | Naikrovek | nope, on top |
| 16:09.06 | Katty | i think i'd rather have a steaming basket |
| 16:09.16 | Naikrovek | yeah that's what this is |
| 16:11.27 | eppigy | hello |
| 16:11.29 | eppigy | i am dave |
| 16:11.37 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 16:16.42 | *** join/#asterisk fainsys (n=fainsys@c-98-242-73-30.hsd1.ga.comcast.net) |
| 16:16.51 | *** join/#asterisk fainsys (n=fainsys@c-98-242-73-30.hsd1.ga.comcast.net) |
| 16:18.02 | p3nguin_ | katty: Just get a bag of Bird's Eye Steamfresh. You'll have nice veggies right from the microwave and you don't have to worry about finding a steamer basket nor anything else. |
| 16:19.01 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:24.47 | Katty | p3nguin_: something tells me it's lacking severely in nutrients. |
| 16:27.36 | p3nguin_ | katty: It'll be fine. They are flash-frozen just like all other reasonable frozen vegetables, and it comes with its own steamer. |
| 16:28.40 | ryduh | Did the DTMF Debug get removed? |
| 16:29.22 | ryduh | nvm |
| 16:29.51 | p3nguin_ | Did it? |
| 16:30.23 | Chainsaw | It got stashed in a dusty corner. But it's alright, he's found it. |
| 16:30.47 | ryduh | When I press TAB in * 1.4.26.1, I don't see dtmf show up. I did add it to the logger.conf though |
| 16:33.56 | [TK]D-Fender | RyAnd in what version did it show up? |
| 16:34.01 | [TK]D-Fender | ryduh: And in what version did it show up? |
| 16:34.42 | *** join/#asterisk tdi (n=tdi@seth.coloco.pl) |
| 16:35.06 | ryduh | [TK]D-Fender: no clue. I found this: https://issues.asterisk.org/view.php?id=339 and hoped it was still in. |
| 16:35.23 | tdi | hi, anybody knows good external 4 SIM gsm->sip gateway ? (for EU) |
| 16:35.49 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 16:35.52 | tdi | i was considering portech |
| 16:36.05 | [TK]D-Fender | ryduh: Resolution open <--- Submitted in *** 2003 *** |
| 16:36.13 | [TK]D-Fender | ryduh: I think you should ignore that. |
| 16:36.21 | [TK]D-Fender | RyThis is pre-1.0 |
| 16:36.51 | Katty | p3nguin_: microwaving destroys heat sensitive vitamins. |
| 16:37.07 | Katty | p3nguin_: of course, so does a lot of other cooking. |
| 16:37.34 | ryduh | Katty: I was just about to say.. wouldn't that be true for regular cooking as well |
| 16:37.50 | ryduh | [TK]D-Fender: I saw that, thats why I had 'hoped' |
| 16:37.53 | Katty | foods cooked in water lose more nutrients because it leaves the food and enters the water. |
| 16:38.01 | Katty | and i know those packs of food contain water in them |
| 16:38.06 | Katty | so steaming them would probably be better of |
| 16:38.28 | [TK]D-Fender | ryduh: core debug shows DTMF |
| 16:38.47 | ryduh | [TK]D-Fender: thanks |
| 16:39.46 | tdi | anybody with DMR->SIP including asterisk experience, im also interested in any devices for connecting ptt radio to sip networks, can be external |
| 16:40.57 | *** join/#asterisk Micc_ (n=Micc@c-71-231-123-28.hsd1.wa.comcast.net) |
| 16:41.36 | Micc_ | I'm having horrible time with aastra phones and the new firmware. |
| 16:41.47 | *** join/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 16:42.22 | torrancew | got a few MOH/call parking questions, can anybody guide me? won't be too long or involved |
| 16:42.23 | Micc_ | All of a sudden this morning a bunch of our customers with 6731i and 6730i's couldn't answer calls by picking up the handset, the audio didn't click over so a workaround for some is to press line 2 then line 1. |
| 16:44.02 | [TK]D-Fender | ~ask |
| 16:44.02 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:44.53 | ryduh | against your will? Who is in here because they have to be? |
| 16:45.21 | tdi | ryduh: infobot for example :) |
| 16:45.30 | ryduh | lol |
| 16:45.52 | wcselby | so....my shiney new polycom 650 running SIP firmware 3.2.1 supports LLDP, however it doesn't appear the bootrom does....\ |
| 16:46.36 | wcselby | anyone in here know the latest polycom bootrom? |
| 16:46.41 | wcselby | i need to see if I'm running an old version I guess |
| 16:47.02 | [TK]D-Fender | wcselby: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 16:47.10 | Jankooo | Hi! Is there anybody who has a little time for helping a newbie? just about the basics... |
| 16:47.15 | wcselby | yeah that's where I got the sip firmware |
| 16:47.24 | wcselby | [TK]D-Fender - sort of just talking to myself in here |
| 16:47.39 | [TK]D-Fender | wcselby: then do it more quietly... we can still hear you :p |
| 16:47.49 | [TK]D-Fender | Jankooo: ... |
| 16:47.51 | [TK]D-Fender | ~ask |
| 16:47.52 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:47.53 | [TK]D-Fender | ^^^^6 |
| 16:48.36 | torrancew | well, i want to know where I can adjust the parked calls setting, specifically the one that has it call back the line that put it on hold after x seconds |
| 16:48.54 | [TK]D-Fender | torrancew: features.conf |
| 16:49.10 | torrancew | thanks |
| 16:49.42 | torrancew | also, our ip phones aren't showing caller id, they're showing everything from "asterisk", where can i change that? |
| 16:50.32 | Katty | leifmadsen: http://42ndhealthstreet.blogspot.com/2009/10/my-new-grocery-list.html |
| 16:50.38 | Jankooo | I have installed the asterisknow under vmware (with NAT) and i can reach the freepbx admin pages (and i added a user in the extensions tab) but when i try to connect with a voip client to the same ip address (where i can reach the admin page) no response |
| 16:50.56 | Katty | leifmadsen: what do you think? |
| 16:51.00 | Jankooo | and i can see with wireshark, the answer is port not reachable for 5060 |
| 16:51.08 | Jankooo | for the register sip message |
| 16:51.52 | Katty | leifmadsen: any recommendations? |
| 16:52.33 | [TK]D-Fender | #freepbx |
| 16:52.40 | [TK]D-Fender | Jankooo: ^^^^ |
| 16:52.45 | [TK]D-Fender | ~freepbx |
| 16:52.46 | 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 |
| 16:53.09 | [TK]D-Fender | torrancew: Where is the call coming from? |
| 16:53.15 | wcselby | wow |
| 16:53.20 | wcselby | this ip650 keeps rebooting itself |
| 16:53.48 | torrancew | [TK]D-Fender: coming from pstn trunk, through our asterisk server, into our linksys spa-942 phones |
| 16:53.57 | [TK]D-Fender | torrancew: Where is the call coming from? <----- |
| 16:54.41 | torrancew | [TK]D-Fender: can you clarify? so far, any calls that aren't internal (from one of our sip accounts), all show caller id of asterisk |
| 16:55.16 | wcselby | and it's not uploading an -app.log file either |
| 16:55.20 | wcselby | even though it says it is |
| 16:55.34 | [TK]D-Fender | torrancew: precisely how are you interfacing with the PSTN? |
| 16:55.52 | torrancew | Digium 402EF card |
| 16:56.27 | [TK]D-Fender | torrancew: No such thing. Try again. |
| 16:57.02 | torrancew | [TK]D-Fender: Digium Wildcard, 4 port card, has 2 active FXO modules |
| 16:57.03 | [TK]D-Fender | torrancew: And the SIGNALLING used on the card is important too. |
| 16:58.15 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 16:59.49 | [TK]D-Fender | torrancew: in zapata.conf / chan_dahdi.conf : usecallerid=yes callerid=asreceived |
| 17:00.01 | [TK]D-Fender | torrancew: and immediate=no |
| 17:00.12 | torrancew | [TK]D-Fender: thanks, what modules need to be reloaded? both for this and features.conf? |
| 17:00.35 | [TK]D-Fender | torrancew: chan_zap.so / chan_dahdi.so |
| 17:01.01 | torrancew | [TK]D-Fender: i meant asterisk side, but thank you |
| 17:01.11 | wcselby | Naikrovek - ping |
| 17:01.13 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:01.18 | Naikrovek | wcselby: pong |
| 17:01.24 | [TK]D-Fender | torrancew: that WAS Asterisk-side |
| 17:01.31 | wcselby | Naikrovek - do you have any ip650 phones in your network? |
| 17:01.41 | torrancew | [TK]D-Fender: *face-palm* |
| 17:01.51 | Naikrovek | wcselby: don't. 320,330, and 6000s only |
| 17:01.55 | wcselby | ahh |
| 17:02.06 | wcselby | Naikrovek - have any contacts for polycom support? |
| 17:02.27 | Naikrovek | wcselby: no. what problem are you having |
| 17:03.06 | wcselby | Naikrovek - two problems...the 4.1.1 bootrom on this ip650 doesn't seem to support lldp, although the sip firmware 3.2.1 that's loaded on here does. and the phone is randomly rebooting itself |
| 17:03.09 | torrancew | [TK]D-Fender: the usecallerid=yes and callerid=asreceived, are they global or per channel? |
| 17:03.22 | wcselby | also, the -app.log file is 0 bytes |
| 17:03.27 | [TK]D-Fender | torrancew: Everything in zapata.conf is per channel |
| 17:03.37 | Naikrovek | wcselby: poe? |
| 17:03.45 | wcselby | Naikrovek - yes |
| 17:04.02 | Naikrovek | wcselby: does it still reboot when not using poe? |
| 17:05.33 | wcselby | Naikrovek - I don't know, haven't tested that yet |
| 17:05.44 | wcselby | Naikrovek - it started doing it on the port configured with multiple vlans |
| 17:05.57 | wcselby | and lldp vlan tagging |
| 17:06.11 | wcselby | i'm moving it back to the port just setup with the phone vlan and testing it there to see if it reboots itself |
| 17:06.21 | Naikrovek | k |
| 17:06.25 | wcselby | also going over the release notes of bootrom 4.2.0, which I think is latest? |
| 17:06.56 | Naikrovek | ivaguely recall cisco devices providing poe can cause problems for polycom phones if they're not configured in a certain way, under certain circumstances |
| 17:07.16 | wcselby | the switches are all juniper, hence the need for lldp support |
| 17:07.18 | Naikrovek | fix the reboot issue then work on the others. I don't know anything about LLDP |
| 17:07.23 | Naikrovek | ah |
| 17:07.27 | wcselby | bootrom 4.2.0 supports lldp |
| 17:07.41 | wcselby | need to upgrade to latest bootrom |
| 17:07.58 | Naikrovek | yeah |
| 17:08.26 | wcselby | i'll try that and see if everything decides to start working |
| 17:08.30 | wcselby | and get back to you |
| 17:08.32 | wcselby | thanks for the ear :) |
| 17:08.45 | *** join/#asterisk d5s (i=c9522e39@gateway/web/freenode/x-wyjicytdzbhqvzmg) |
| 17:09.06 | d5s | Hi, is it possible to run an AGI script when asterisk comes up? |
| 17:09.11 | ryduh | If I'm using Asterisk to originate a call through a voip provider and then to a PTSN line, is RFC2833 better or is INFO better for DTMF tones? I can't seem to get my old panasonic system here to pick up DTMF tones from asterisk |
| 17:11.53 | Naikrovek | ryduh: you're on a phone connected to asterisk connected to a voip provider? |
| 17:12.36 | ryduh | Naikrovek: I originate a call with * via a call file and then try to send DTMF tones |
| 17:12.54 | Naikrovek | ryduh: where does the panasonic system come into play |
| 17:13.17 | torrancew | can anyone recommend a good web-based interface? |
| 17:13.24 | ryduh | Naikrovek: It is the destination I am trying to get to accept the DTMF tones |
| 17:13.41 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 17:13.55 | torrancew | i've got no problem using the console, but the end users need a different way to interact/see what's going on |
| 17:14.27 | [TK]D-Fender | torrancew: depends on the complexity of the system |
| 17:15.01 | p3nguin_ | Quite honestly, I wouldn't want end-users playing around with my Asterisk configs. |
| 17:15.09 | torrancew | [TK]D-Fender: nothing too complex, 2 lines (soon to be 4) coming from a pstn (our only trunks), and a few end-users |
| 17:15.46 | torrancew | p3nguin_: i'm more looking for something that can let them see parked calls, call logs, and maybe VoiceMail integration, not so much config editing |
| 17:15.52 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 17:15.55 | [TK]D-Fender | d5s: make an app that on startup releases a time-delayed ORIGINATE or call-file |
| 17:16.17 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:17.02 | [TK]D-Fender | torrancew: the last 2 hae separate tools, parking may require something more. Perhaps FOP or one of the other receptionist apps could do if you don't want to sign your sould over completely |
| 17:18.11 | torrancew | [TK]D-Fender: what about asterisk-gui? is it more for config stuff? i played with freepbx, but it added way too much into the configs, and made it too hard to manage unless you were committed to using it and it alone, IMO |
| 17:18.39 | [TK]D-Fender | torrancew: Same shit, differnt smell... only less "complete" |
| 17:19.00 | *** join/#asterisk jicksta_ (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 17:19.20 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 17:20.08 | d5s | [TK]D-Fender: The point is that I need to access the script I ran on asterisk startup if a call is received or a call is placed. Will that be possible? |
| 17:20.54 | [TK]D-Fender | d5s: How do you access something that ran in the past? have you perfected time travel? |
| 17:21.06 | [TK]D-Fender | d5s: I think you need to reword your request... |
| 17:21.20 | d5s | [TK]D-Fender: I'll do that... |
| 17:21.24 | *** join/#asterisk KMiLo (n=GeniuS@190.65.75.167) |
| 17:22.56 | ryduh | When I originate a call from * to my cell phone, and then SendDTMF(1w) SendDTMF(0w) SendDTMF(7w) I only hear the first tone. I should be hearing all 3 tones right? |
| 17:23.08 | d5s | [TK]D-Fender: I have this script that is running in eternal loop. It is part of an FSM. It must be executed on system startup. When a call is received I want to send interruptions to that script. Did you get it? |
| 17:23.39 | [TK]D-Fender | d5s: What is this script doing? |
| 17:23.53 | ryduh | The Flying Spaghetti Monster? |
| 17:24.18 | d5s | Waiting for instructions from other scripts like outgoingCall.agi and incomingCall.agi |
| 17:24.47 | [TK]D-Fender | d5s: And d5s And how does it receive instructions? |
| 17:24.56 | [TK]D-Fender | d5s: and how does it send them? |
| 17:25.17 | d5s | interruption signals. |
| 17:26.48 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:27.34 | d5s | [TK]D-Fender: Can you help me running my script on asterisk initialization? |
| 17:28.53 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 17:29.03 | wcselby | why won't my ip650 pick up the new bootrom? |
| 17:29.12 | wcselby | do I need to do more than just place it in my ftp root dir? |
| 17:29.44 | *** join/#asterisk Tim_Toady (n=moi@adsl41-129.kln.forthnet.gr) |
| 17:30.01 | jameswf | Im retarded whats the vuc room |
| 17:30.56 | ryduh | If I send three SendDTMF()'s I should hear 3 tones correct? |
| 17:37.39 | ryduh | Would anyone know why, when trying to SendDTMF, * changes format to SLIN and then back to ULAW? I have dtfmmode=rfc2833 |
| 17:37.44 | ryduh | http://pastebin.com/d5b480d72 |
| 17:37.46 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 17:39.09 | ryduh | scratch that, I didn't reload my sip.conf file. I'm still only hearing the tone when SendDTMF(1) is sent. I don't hear anything for 0 and 7 |
| 17:39.12 | afink | Hello everyone, I am trying to get realtime sip peers working. When I start up asterisk and the first dynamic host loads it chan_sip.c gets stuck in a loop. |
| 17:40.36 | ryduh | Hrmm. Now that I know I'm in rfc2833 mode, after 3 SendDTMF's, debug shows this: [Oct 23 10:39:42] DEBUG[13361]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw |
| 17:41.30 | *** join/#asterisk d5s (i=c9522e39@gateway/web/freenode/x-lsbexxccuobionrp) |
| 17:42.09 | afink | whats happening for me is as soon as a phone registers I get this: http://pastebin.com/m537716cd in the cli |
| 17:42.30 | d5s | [TK]D-Fender: Hi, I'm not sure you've answered my question since I got disconnected. If you did, could you resend it? |
| 17:42.31 | afink | I wonder if I just set qualify to no if it will work |
| 17:42.55 | ryduh | d5s: he did not answer |
| 17:43.09 | d5s | thanks ryduh. |
| 17:44.00 | p3nguin_ | afink: wtf |
| 17:44.33 | p3nguin_ | afink: If that is accurate, maybe some network troubleshooting is in order. |
| 17:44.53 | [TK]D-Fender | [13:15]<[TK]D-Fender>d5s: make an app that on startup releases a time-delayed ORIGINATE or call-file |
| 17:44.59 | [TK]D-Fender | d5s: I answered you at the very start |
| 17:45.15 | wcselby | helps if I upload the right bootrom to the ftp root |
| 17:45.18 | wcselby | bbiab, lunch |
| 17:47.42 | afink | p3nguin_: it works perfectly with the flat file |
| 17:47.48 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 17:48.44 | afink | Is there something I need to put in the sip.conf? I don't see any mention of it here: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 17:49.45 | d5s | sorry [TK]D-Fender, could you please elaborate a little bit more on that? I'm not familiar to ORIGINATE and don't know how to release a call-file |
| 17:50.17 | [TK]D-Fender | d5s: its all in the BOOK. |
| 17:50.21 | [TK]D-Fender | ~book |
| 17:50.22 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:51.01 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 17:51.08 | d5s | [TK]D-Fender: I have one of those... I'll take a look. Thanks for your time. |
| 17:51.54 | ryduh | Anyone have an idea on how I can further debug my SendDTMF problems? |
| 17:52.16 | afink | anyone here had trouble with realtime sip? |
| 17:54.44 | ryduh | Can I Dial() and then somehow send a DTMF tone from asterisk? |
| 17:54.58 | *** join/#asterisk niekie (i=quasselc@78.129.140.218) |
| 17:54.58 | d5s | [TK]D-Fender: I was only able to find how to call an AGI script from dial plan. Could you give an extra hand on that? |
| 17:55.48 | [TK]D-Fender | d5s: AGI = dialplan. |
| 17:55.58 | ryduh | d5s: You could create some other script, outside of asterisk, to generate a .call file and call your own extension which would then run your script |
| 17:56.02 | afink | then this...: http://pastebin.com/m137e3bbc |
| 17:56.14 | [TK]D-Fender | d5s: You still haven't made it clear why you want DIALPLAN to get called as some sort of "idle process" |
| 17:57.05 | CcRnp | Is it posssible to use CEL in asterisk version 1.2 or 1.4 ? |
| 17:57.53 | d5s | That is the problem [TK]D-Fender: I don't want to call my script from dialplan. I don't know how to start it when asterisk gets up. |
| 17:57.59 | d5s | Let me try to explain: |
| 17:59.19 | [TK]D-Fender | d5s: Explanation of what you actually wat to do would be helpful. See you said "call an agi". that IS DIALPLAN. |
| 18:00.43 | d5s | When I receive a call my outgoingCall.agi script is launched. It has to do a lot of processing. And it is getting more that 4 seconds to be processed. So I want start another script on system initialization (or asterisk initialization) so I can load part of I need to do on this script which will make my outgoingCall.agi run fasten... |
| 18:01.36 | [TK]D-Fender | d5s: and what does that background script do? |
| 18:01.40 | ryduh | d5s: how will it make your outgoingCall.agi faster? |
| 18:02.47 | d5s | [TK]D-Fender: the background script wait for commands from my outgoingCall.agi script. Thats why my script has to be executed by asterisk. They have to be in the same context. |
| 18:03.38 | d5s | ryduh: It is simple. It will save me from importing python libraries. Each of them gets about 1 second to be loaded |
| 18:03.51 | [TK]D-Fender | d5s: WTF |
| 18:04.03 | afink | should sip show peers show realtime registrations? |
| 18:04.12 | [TK]D-Fender | d5s: What the hell is your background script DOING that involes the concept of CONTEX? |
| 18:04.57 | d5s | [TK]D-Fender: ok... my background script will SET and GET variables from asterisk context |
| 18:05.38 | ryduh | Is there anyway to execute commands after a call has been Dial()'d and answered? |
| 18:05.43 | [TK]D-Fender | d5s: No. First Contexts don't have variables. CHANNELS have variables. Second your scrip is not part of that other channel. |
| 18:07.15 | [TK]D-Fender | d5s: And you still haven't specified how your "on call" script will signal to the background script, and vice-versa |
| 18:07.30 | d5s | [TK]D-Fender: The set and get I was talking about are executed with: sys.stdin.read('SET VARIABLE.....) |
| 18:07.43 | d5s | on python |
| 18:07.48 | [TK]D-Fender | d5s: NO. |
| 18:08.04 | [TK]D-Fender | d5s: those are per CHANNEL. And AGI is for the scope of the call that its in. |
| 18:08.14 | [TK]D-Fender | d5s: it does not target another channel <- |
| 18:10.20 | Katty | omnomnomnoms chicken casserole |
| 18:10.28 | d5s | ok. Lets say I have to send and EXEC Dial on my outgoingCall.agi script, ok? |
| 18:10.40 | d5s | thats what I'm used to do: |
| 18:10.42 | afink | anyone know why my realtime registrations won't show up with sip show peers? |
| 18:10.54 | d5s | sys.stdout.write(EXEC Dial....) |
| 18:11.25 | d5s | does it make any sense to you [TK]D-Fender ? |
| 18:12.42 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 18:13.54 | nny | this is odd |
| 18:14.09 | nny | I have an ARG variable getting "lost" in a macro heh |
| 18:14.09 | Katty | is it prime, too? |
| 18:14.22 | TJNII | [TK]D-Fender: I was in Barnes and Noble yesterday and say a Asterisk for dummies book. Made me think of you. |
| 18:14.26 | ryduh | lol. i was just about to make a remark like that |
| 18:14.30 | nny | Katty: heh |
| 18:14.31 | TJNII | s/say/saw/ |
| 18:14.33 | nny | http://pastebin.com/m9fab78b |
| 18:14.39 | niekie | What the hell? [Oct 23 18:14:21] ERROR[1839]: chan_sip.c:11640 register_verify: Peer 'htc' is trying to register, but not configured as host=dynamic |
| 18:14.44 | niekie | It *IS* configured as host=dynamic |
| 18:14.58 | niekie | What am I doing wrong? :S |
| 18:15.22 | TJNII | niekie: Have you reloaded the config? |
| 18:15.23 | niekie | Oh wait. |
| 18:15.32 | niekie | TJNII: just thought of that :D |
| 18:15.47 | Katty | oh boy :< |
| 18:15.54 | niekie | Ok, that works. |
| 18:16.00 | niekie | Sorry to have disturbed. I'm an idiot >_< |
| 18:16.09 | TJNII | Simplest problems can be the hardest to find. |
| 18:16.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:16.53 | jblack | often, they are the hardest to find. |
| 18:17.10 | TJNII | You'll look right at it and not see it. |
| 18:17.10 | Katty | often, they could be all together avoided. |
| 18:17.46 | jblack | never. |
| 18:17.46 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 18:17.46 | nny | so i have Macro(foo,ARG1,ARG2) and [macro-foo] Something(arg1@default,u) under what conditions would ARG1 be nulled out? |
| 18:17.50 | jblack | problems = life. No problems, no life. |
| 18:17.51 | Katty | i'm feeling snippity today |
| 18:18.07 | jblack | so make some side cash as a barber. |
| 18:18.10 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:18.16 | Katty | k |
| 18:18.17 | jblack | grins |
| 18:18.27 | jblack | what's wrong? |
| 18:18.34 | Katty | idk |
| 18:18.38 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 18:18.42 | niekie | And that concludes the setup of SIP on my new shiny Android. \o/ |
| 18:18.52 | Katty | i'm a girl. being snippity comes normal. |
| 18:19.03 | nny | niekie: which app you using? |
| 18:19.20 | jblack | hrmm. Well, can't you solve it the normal way? i.e. torture your boyfriend with subtle, conficting hints? |
| 18:19.25 | niekie | SipDroid |
| 18:19.30 | Katty | why would i do that? |
| 18:19.34 | Katty | he's not the cause of my snippity. |
| 18:19.43 | jblack | isn't that what girls always do? |
| 18:19.49 | Katty | hmm. |
| 18:19.52 | Katty | maybe. |
| 18:19.54 | nny | niekie: did they fix the issue where it wouldn't work with asterisk without losing the auth? |
| 18:19.56 | TJNII | jblack: +1 |
| 18:19.56 | Katty | but then boyfriend becomes snippity. |
| 18:20.00 | Nugget | http://vimeo.com/7151435 <-- I love it too. |
| 18:20.01 | Katty | and i don't care for that. |
| 18:20.26 | jblack | Rule #1 on the boy side of relationships is "Take good care of how she feels, because if she's not happy, she'll make you unhappy. |
| 18:21.07 | ecrane | What's rule #1 on the girl side of relationships? |
| 18:21.16 | jblack | You know. imply that he gives too little much attention and also imply that he gives too much attention. At the same time |
| 18:21.33 | Katty | rule #1 if boyfriend seems cranky, abandon room for 1 hour. return and check status of boyfriend. repeat as needed. |
| 18:21.39 | niekie | nny: no idea. I'm filling in client details on it now. |
| 18:21.50 | jblack | That's a good rule. |
| 18:21.58 | Katty | yesh. |
| 18:22.03 | ecrane | yeah; good rule. |
| 18:22.13 | nny | niekie: there was an issue where it wouldn't dial out or something similar cause the sip header wasn't right on it, that was like 3 months ago |
| 18:22.19 | nny | so |
| 18:22.20 | nny | http://pastebin.com/m9fab78b |
| 18:22.24 | jblack | How about the two of you go out to see a funny movie? There's some good stuff out |
| 18:22.26 | nny | anyone see why I am losing ARG1 there? |
| 18:22.39 | [TK]D-Fender | d5s: Your entire understanding of AGI's needs to be rethought. |
| 18:23.04 | Katty | my snippity isn't a problem that requires fixing. |
| 18:23.07 | jblack | Yeah, You need to reread the section of macros. |
| 18:23.09 | Katty | it will resolve itself on its own. |
| 18:23.18 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 18:23.23 | Katty | namely, as soon as i go home and get away from these monkeys (= |
| 18:23.26 | nny | jblack: ? |
| 18:23.27 | jblack | Katty: Yes ma'am |
| 18:23.41 | d5s | [TK]D-Fender: Hello, I'm d5s co-worker, currently helping with this script development, let me explain to you the real issue |
| 18:23.47 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:23.57 | jblack | nny: You're treating macros like a full context. They're much more limited than that. You need to reread the book/documentation on macros. |
| 18:24.01 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 18:24.12 | Katty | TimRiker: where'd you leave william? |
| 18:24.18 | nny | jblack: exten => 3003,1,Macro(stdexten,3003,SIP/3003&SIP/4003) should pass "3003" as ARG1 and "SIP/3003&SIP/4003" as ARG2 only to the macro no? |
| 18:24.30 | jblack | nny: Go. Read. The. Book. |
| 18:24.32 | nny | jblack: ARG2 is being passed |
| 18:24.37 | nny | jblack: oh ffs I have |
| 18:24.41 | [TK]D-Fender | nny: You are showing us 2 completely separate calls <- |
| 18:24.41 | jblack | AGAIN! |
| 18:24.48 | [TK]D-Fender | nny: 1 SIP, 1 DAHDI |
| 18:24.59 | [TK]D-Fender | nny: -- Executing [3006@sip:1] Macro("SIP/3003-b7c35890", "stdexten|3006|SIP/3006&SIP/4006") in new stack |
| 18:24.59 | Katty | nny: try putting the book on your head. |
| 18:24.59 | nny | [TK]D-Fender: yeah there is some noise in there, I can re capture the info |
| 18:25.02 | TimRiker | Katty: I think he's seducing some woman back on the last planet we visited. |
| 18:25.07 | nny | Katty: can I hit people with it? |
| 18:25.09 | d5s | [TK]D-Fender: I understand that when a script is called via AGI, Asterisk automatically estabilishes a pipe between the script and itself, correct? So, I wanted to use the same principle, but for a script that does not is called by AGI |
| 18:25.11 | Katty | nny: sure! |
| 18:25.14 | [TK]D-Fender | nny: I fail to see... the failure :) |
| 18:25.15 | nny | sweet |
| 18:25.17 | Katty | TimRiker: probably :> |
| 18:25.21 | Katty | TimRiker: <3 |
| 18:25.31 | nny | [TK]D-Fender: -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("DAHDI/4-1", "@default|u") in new stack |
| 18:25.38 | Katty | nny: i often put the book on my head. |
| 18:25.41 | [TK]D-Fender | d5s: A script is not called by an AGI. An AGI *IS* a script |
| 18:25.42 | Katty | nny: somehow, it helps me think. |
| 18:25.50 | jblack | I need to grocery shop, and have some food. |
| 18:25.55 | nny | but in the macro it's exten => s-NOANSWER,1,Voicemail(${ARG1}@default,u) |
| 18:25.55 | Katty | jblack: yesh. |
| 18:25.58 | jblack | Katty: here's to hoping your mood improves. |
| 18:26.00 | Katty | jblack: that sounds divine. |
| 18:26.05 | Katty | jblack: it already has! |
| 18:26.13 | jblack | I'm that good? |
| 18:26.17 | Katty | jblack: but now i have to pee :< |
| 18:26.17 | TimRiker | It's the riker influence that does it. |
| 18:26.17 | [TK]D-Fender | d5s: And AGI continues a CALL's dialplan execution within that script and then dumps back to the dialplan when the script finishes or the call dies |
| 18:26.19 | tuxcrafter | !book |
| 18:26.24 | [TK]D-Fender | d5s: it is NOT a background process |
| 18:26.31 | d5s | [TK]D-Fender: Yes, I know |
| 18:26.32 | jblack | I hear that I cause that problem all the time. :) |
| 18:26.38 | Katty | ;P |
| 18:26.51 | jblack | have fun all. |
| 18:26.54 | Katty | nny: you could always try finding a consultant. |
| 18:26.56 | jblack | [TK]D-Fender: You should beat him, imho |
| 18:27.02 | Katty | nny: someone who will look over your shoulder and Point. |
| 18:27.17 | tuxcrafter | is searching for that online asterisk book |
| 18:27.17 | nny | Katty: cute |
| 18:27.22 | nny | ~book |
| 18:27.22 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:27.31 | tuxcrafter | nny: thanks |
| 18:27.37 | Katty | nny: i'm just saying. |
| 18:27.45 | nny | Katty: what are you saying? |
| 18:27.47 | ryduh | ~buybook |
| 18:27.48 | infobot | [~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 18:27.52 | Katty | :< |
| 18:27.55 | d5s | [TK]D-Fender: My real question is, is there a way to estabilish such a connection before dialplan execution? I mean, make asterisk and another program share the same stdin/stdout for communication? |
| 18:28.12 | Katty | the snippity came back. |
| 18:28.26 | [TK]D-Fender | d5s: Youa re describing AGI. AGI is not a background process. Youkeep trying to ask how to use a screwdriver like a hammer. |
| 18:29.12 | nny | [TK]D-Fender: i fail to see why this macro is improper, i have used it in every build, and it's actually the "macro-stdexten" standard recipe shown in numerous places |
| 18:29.25 | tuxcrafter | is trying to make it possible that extention 406 can call 407 but the logs show me that [Oct 23 20:24:15] NOTICE[2846]: chan_sip.c:13885 handle_request_invite: Call from '406' to extension '407' rejected because extension not found. |
| 18:29.35 | d5s | [TK]D-Fender: Not like that, I'm not trying to use AGI as background script, it's the other way around, use a background script that behaves like AGI, when it comes to communicating with asterisk |
| 18:29.36 | [TK]D-Fender | nny: I never said the macro is improper. I'm saying your little pastebin of CLI output didn't match and didn't show the error |
| 18:29.40 | tuxcrafter | outgoing call are working fine btw |
| 18:29.59 | [TK]D-Fender | d5s: You are not making any sense |
| 18:30.01 | nny | [TK]D-Fender: -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("DAHDI/4-1", "@default|u") in new stack [Oct 23 14:05:19] WARNING[29478]: app_voicemail.c:4011 leave_voicemail: No entry in voicemail config file for ' |
| 18:30.21 | nny | [TK]D-Fender: it's somehow not passing ARG1 , even though it passes ARG2 on the dial statement |
| 18:30.36 | tuxcrafter | http://debian.pastebin.com/m29c1b070 |
| 18:30.37 | [TK]D-Fender | nny: I don't see the MACRO getting called by that channel |
| 18:30.43 | tuxcrafter | see for my configuration of the extentions |
| 18:30.43 | [TK]D-Fender | nyyour CALL DATAT did not match <- |
| 18:30.48 | d5s | [TK]D-Fender: How does an AGI script exchange info with asterisk? Via stdin/stdout correct? |
| 18:31.06 | [TK]D-Fender | tuxcrafter: those are not EXTENSIONS |
| 18:31.22 | [TK]D-Fender | d5s: Yes |
| 18:31.50 | tuxcrafter | hides behind his book and starts reading again |
| 18:32.38 | d5s | [TK]D-Fender: So, I'm asking if you could inform me, or teach, how such a connection is estabilished in the first place. I want to do something like this with another script, one which can't be invoked in the dialplan execution |
| 18:32.40 | Katty | ^_- |
| 18:32.47 | Katty | i walked down the hallway and the sales reps were singing |
| 18:32.50 | Katty | loudly. |
| 18:33.04 | [TK]D-Fender | d5s: You are not providing a coherent enough description to advise ANYTHING. |
| 18:33.16 | nny | [TK]D-Fender: gonna try to recapture the error, but this is a system in use, can't isolate the noise |
| 18:33.20 | [TK]D-Fender | d5s: You are in "theory-land" with the little pieces you are talking about. |
| 18:33.20 | Katty | I advise you to eat a mango. |
| 18:33.28 | nny | [TK]D-Fender: but as far as I can tell, the pastebin I posted has the full call |
| 18:33.36 | nny | <PROTECTED> |
| 18:33.39 | Katty | Protect your electrons! Eat a blueberry! |
| 18:33.45 | nny | theres the macro call |
| 18:33.46 | [TK]D-Fender | d5s: Advise requires a consistent implementation to be defined |
| 18:34.08 | [TK]D-Fender | nny: that is not the DAHDI channel issuing that, so why do i care aboutit? |
| 18:34.11 | d5s | [TK]D-Fender: Not in theory land, IMO the question was pretty clear, I want to share asterisk stdin/stdout with another script, like it is done with AGI |
| 18:34.16 | [TK]D-Fender | [14:29]<nny>[TK]D-Fender: -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("DAHDI/4-1", "@default|u") in new stack [Oct 23 14:05:19] WARNING[29478]: app_voicemail.c:4011 leave_voicemail: No entry in voicemail config file for ' |
| 18:34.18 | Katty | I advise you to implement more happiness into your day |
| 18:34.20 | [TK]D-Fender | nny: not the same channel! |
| 18:34.57 | [TK]D-Fender | d5s: there isn't a single in/out. each call is a SPAWNED PROCEss. not GLOBAL |
| 18:35.04 | [TK]D-Fender | d5s: there is nthing global to hook into |
| 18:35.24 | ian6 | HAPPY INTERNATIONAL CAPS LOCK DAY. |
| 18:35.39 | Qwell | ian6: yeah, that was yesterday |
| 18:35.40 | [TK]D-Fender | ian6: FUCK YEAH! |
| 18:35.57 | ChannelZ | DON'T MAKE ME BUST A CAPS LOCK IN YOUR ASS |
| 18:36.09 | d5s | [TK]D-Fender: Oh, I see it. So I'd have to hook to each call, not asterisk itself |
| 18:36.14 | Katty | oh dear. |
| 18:36.31 | [TK]D-Fender | d5s: Each call is 100% separate from another. |
| 18:36.34 | ian6 | Qwell: Hmm, so it was. Do you really think just one day is enough though? |
| 18:36.46 | Qwell | ian6: That is why there are 2 every year. |
| 18:36.55 | d5s | [TK]D-Fender: Thanks for the info =) |
| 18:37.00 | ian6 | ... also, fuck caps lock. There needs to be an international holds-shift-down-like-a-real-man day. |
| 18:37.06 | d5s | Helped a lot! |
| 18:37.52 | ChannelZ | Or Unnecessarily Initial Cap Each Word Day |
| 18:38.38 | Nugget | http://bash.org/?105199 |
| 18:41.42 | nny | [TK]D-Fender: http://pastebin.com/ |
| 18:41.47 | nny | [TK]D-Fender:er shit |
| 18:41.56 | [TK]D-Fender | ian6: I do.. thats why I lose my caps in the middle all too often :) |
| 18:41.56 | nny | http://pastebin.com/m56e04466 |
| 18:42.04 | nny | [TK]D-Fender: more than you probably needed, but... |
| 18:42.44 | nny | [TK]D-Fender: the Unable to create channel of type 'SIP' is for extensions related to their mobile sip clients, didn't know of a more elegant way of having it dial both, even when one is offline |
| 18:44.20 | [TK]D-Fender | nny: nny What line is your DAHDI channel failing to get the right arg? |
| 18:44.34 | nny | [TK]D-Fender: dude, this is the whole call |
| 18:44.40 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:44.40 | nny | [TK]D-Fender: i dunno why it doesn't say DAHDI on it |
| 18:44.44 | [TK]D-Fender | nny: WHAT LINE? |
| 18:44.45 | *** join/#asterisk baijum (n=baiju@122.166.148.94) |
| 18:45.04 | [TK]D-Fender | nny: The fact you're not looking at this is a bad sign... |
| 18:45.07 | nny | [TK]D-Fender: 350 |
| 18:45.36 | nny | [TK]D-Fender: Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("DAHDI/4-1", "@default|u") in new stack |
| 18:45.43 | [TK]D-Fender | nny: From a zombie channel... perhaps it lost all its vars <- |
| 18:46.00 | nny | [TK]D-Fender: what would cause that? |
| 18:46.09 | [TK]D-Fender | hrm... hold that idea... |
| 18:48.21 | leifmadsen | anyone around who could test a SIP URI for me? |
| 18:48.45 | *** join/#asterisk hugorebelo (n=hugorebe@200.171.132.124) |
| 18:49.49 | [TK]D-Fender | nny: == Channel 'DAHDI/4-1' jumping out of macro 'xfer2exten' |
| 18:50.07 | [TK]D-Fender | nny: I'm wondering if the way args get stacked, and your use of hard jumps is affecting this.. |
| 18:50.41 | *** join/#asterisk RobH (n=RobH@216.38.133.254) |
| 18:51.44 | tuxcrafter | http://debian.pastebin.com/m7f31c202 |
| 18:51.58 | tuxcrafter | phone rings now and i can pick it up, but no audio in or out. i can call from both phones to external numbers with audio in and out so it should work, is my extention wrong? |
| 18:53.33 | nny | [TK]D-Fender: is this a normal issue? Should I rewrite my dial plan somehow? |
| 18:53.34 | nny | [TK]D-Fender: i can post it, it's a whopper |
| 18:54.02 | *** join/#asterisk Tim_Toady (n=moi@77.49.235.108) |
| 19:01.08 | [TK]D-Fender | nny: Not in a position to do a large-scale debug on this |
| 19:01.31 | nny | [TK]D-Fender: i hear ya, i'll figure out a work around until I can review the jumps and see how to combine them |
| 19:01.55 | tuxcrafter | forgot the [internal] |
| 19:02.28 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 19:02.28 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:07.26 | tuxcrafter | still now sound in and out |
| 19:09.02 | Katty | blargh |
| 19:09.06 | Katty | i still have no vitamin d |
| 19:09.45 | TJNII | So go lay in the sun. |
| 19:10.11 | Katty | something tells me that's not going to happen |
| 19:10.31 | TJNII | Hey, it fits with your nick. |
| 19:10.51 | Katty | hmm. |
| 19:10.52 | Katty | k |
| 19:12.05 | dustybin | hugs Katty |
| 19:12.21 | tuxcrafter | http://debian.pastebin.com/m79e29d1d |
| 19:12.41 | tuxcrafter | does somebody see why i can't call extention 406 with sound? |
| 19:13.24 | TJNII | tuxcrafter: You need to pastebin a sip debug to fix audio issues. |
| 19:13.33 | TJNII | tuxcrafter: Any NATs involved? |
| 19:14.46 | *** join/#asterisk Skeeter- (i=Skeeter-@190-141.cgocable.ca) |
| 19:15.19 | tuxcrafter | TJNII: yes there are nats |
| 19:15.26 | TJNII | ~sipnay |
| 19:15.28 | tuxcrafter | but i can call to external numbers and then it works |
| 19:15.30 | TJNII | ~sipnat |
| 19:15.31 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:15.32 | Skeeter- | i have a spectralink SIP phones, it can call ext or receive call, it CANT dial voicemail or queue |
| 19:15.52 | TJNII | tuxcrafter: Sip calls out work but sip calls in don't? |
| 19:16.09 | TJNII | Or, should I say, phones connecting in don't? |
| 19:17.14 | TJNII | Skeeter-: Your dtmfmode is probably wrong |
| 19:17.22 | Skeeter- | i must edit soemthing in [from-sip] i think |
| 19:17.52 | tuxcrafter | TJNII: well i always used the phones to only call out to external numbers, but the 406 has also an other sip settings that is now disabled but is used to call out, so i know the portforwardings are good |
| 19:19.05 | Skeeter- | TJNII: dtmfmode=rfc2833 |
| 19:20.27 | TJNII | tuxcrafter: Well, pastebin a sip debug and maybe we can figure it out. |
| 19:21.35 | TJNII | Skeeter-: I set this on my phones so I may be wrong, but try auto or rfc2833,inband,info (The second one may be formatted wrong, I assume [TK]D-Fender will jump in and correct me if it is) |
| 19:24.14 | Skeeter- | dtmfmode=auto ?? |
| 19:24.20 | Skeeter- | i have a spectralink 8030 |
| 19:27.40 | [TK]D-Fender | tuxYou need a LOT more than jsut port forwarding. Read the guide. |
| 19:28.07 | Skeeter- | in my dialplan [from-sip] do i need to had,: exten => 123,3,Voicemail(44) |
| 19:28.35 | tuxcrafter | TJNII: http://debian.pastebin.com/m4ebe27b5 < the sip debug output |
| 19:31.06 | nny | [TK]D-Fender: hmm odd, removed all the macro nonsense on a test line and still happens |
| 19:31.48 | nny | http://pastebin.com/m73d07f18 |
| 19:32.14 | *** join/#asterisk QaDeS (n=mklaus@p4FC72A6F.dip0.t-ipconnect.de) |
| 19:32.51 | nny | [TK]D-Fender: er nm |
| 19:33.02 | Skeeter- | sip over NAT, this should not exist |
| 19:33.06 | nny | [TK]D-Fender: failed to set something back properly :\ |
| 19:35.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:36.11 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:39.27 | Skeeter- | i need to make a whole dialplan for the [from-sip] |
| 19:40.34 | CcRnp | Is there anyway to track transfer call without using Channel Event Logging ? |
| 19:40.40 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1096762451.dsl.bell.ca) |
| 19:41.28 | dlynes | Are there any known issues navigating t.38 from a 1.6.1.1 client to a 1.4.22 server, over sip? |
| 19:42.30 | tuxcrafter | could somebody make a test call to <sip:406@an4705.voipgate.nl> |
| 19:43.28 | *** join/#asterisk baijum (n=baiju@122.166.148.94) |
| 19:44.28 | CcRnp | Guys anyone anyone have some ideas on tracking transfer calls wihtout using Channel Event Logging , please help me out !! |
| 19:45.06 | wcselby | Naikrovek - just FYI, I got everything to work by updating to the latest bootrom version for my IP650 |
| 19:45.13 | beek | tuxcrafter: I can do that for you. |
| 19:45.16 | Naikrovek | wcselby: awesome |
| 19:45.22 | Naikrovek | i upgraded mine as well |
| 19:46.18 | tuxcrafter | beek: thanks :D |
| 19:46.21 | tuxcrafter | seems to work perfect |
| 19:46.22 | beek | np |
| 19:46.25 | beek | Sounds great |
| 19:52.57 | nny | [TK]D-Fender: yeah removed some of the nested macros and it worked fine. In the future I'll stop using them as often and also need to start using gosub anyways heh |
| 19:53.23 | tuxcrafter | ok got a meetme confrence room up and running |
| 19:58.45 | luca`gervasi | is anybody using skype4asterisk from digium? |
| 19:59.10 | qxork | I am |
| 19:59.40 | luca`gervasi | what do you do to make your account login? i tryied to configure it using the administration manual |
| 19:59.47 | luca`gervasi | following it line by line |
| 20:00.00 | luca`gervasi | but my (business) account doesn't login |
| 20:00.06 | *** join/#asterisk knctrnl (n=aembrey@nat/digium/x-rzmwmwwrlxttrorn) |
| 20:00.11 | luca`gervasi | i enabled all the debug |
| 20:00.36 | luca`gervasi | but all i've got is some too deep code... like "send X", "receive Y" |
| 20:00.59 | qxork | I added the user to chan_skype.conf |
| 20:01.02 | luca`gervasi | i tryied to logon manually with skype logon user blablabla (or something like that) |
| 20:01.24 | luca`gervasi | may i ask your (purged) chan_skype.conf ? |
| 20:01.39 | qxork | k |
| 20:01.53 | luca`gervasi | thanks :D |
| 20:02.07 | luca`gervasi | may i have your relevant extensions.conf ? |
| 20:03.29 | qxork | nothing is needed in extensions for the login / logout. |
| 20:04.01 | qxork | http://pastebin.com/m2887faf0 |
| 20:04.14 | qxork | just replace the REDACTED with your username / password :) |
| 20:04.26 | luca`gervasi | Thanks |
| 20:04.35 | qxork | this one is for 2 users. Mostly the default conf |
| 20:04.36 | luca`gervasi | i'm looking for differences :D |
| 20:04.47 | nny | well |
| 20:05.00 | nny | this is my biggest extensions.conf yet ha |
| 20:05.02 | nny | 23k |
| 20:05.08 | luca`gervasi | why don't you use g729? |
| 20:05.10 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 20:05.10 | nny | should start a contest |
| 20:05.12 | nny | who can have the most bloated dialplan |
| 20:05.34 | [TK]D-Fender | cheats and Installs FreePBX. |
| 20:05.42 | [TK]D-Fender | nny: I WIN! |
| 20:05.49 | nny | lol |
| 20:06.02 | nny | nice one |
| 20:06.05 | qxork | luca`gervasi: because I don't like g729 |
| 20:06.15 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 20:07.12 | luca`gervasi | qxork, may i ask why? (damn... i have the same chan_skype.conf!!!!!) |
| 20:07.33 | *** join/#asterisk Badrobot- (n=badrobot@76.173.229.89) |
| 20:07.33 | qxork | member:luca%60gervasi: I don't like g729 for pretty much any call.... just hate the sound. I'd rather go hidef than lodef |
| 20:08.16 | luca`gervasi | ok |
| 20:08.50 | *** join/#asterisk andres833 (n=andres83@201.244.125.6) |
| 20:10.28 | qxork | just a personal preference. If it's low bandwidth I'll use a diff codec. But I try to keep ulaw or g722 |
| 20:10.58 | luca`gervasi | :( i'm still logout :( |
| 20:11.02 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 20:12.51 | qxork | is it a business account? |
| 20:13.27 | qxork | (you need a special account for skype for asterisk) |
| 20:13.30 | *** join/#asterisk JimDickenson (n=dickenso@c-98-232-185-121.hsd1.or.comcast.net) |
| 20:13.51 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:14.08 | JimDickenson | can someone tell me where issues.digium.com has gone to? |
| 20:14.39 | qxork | https://issues.asterisk.org/main_page.php |
| 20:14.49 | JimDickenson | thanks |
| 20:14.58 | qxork | np. just added a bug :) |
| 20:16.20 | asterwiki | sighs after one of those days and says TGIF |
| 20:19.20 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 20:21.35 | Katty | stuck on the phone with a windows user who has somehow managed to loose the local admin account password on this workstation :< |
| 20:22.01 | Naikrovek | why would the user need the local admin passwd |
| 20:22.08 | Katty | because it was on a domain |
| 20:22.10 | Katty | and it was taken off |
| 20:22.11 | loather-work | to install software |
| 20:22.20 | Katty | with no other users added to the local machine |
| 20:22.32 | Naikrovek | loather-work: so make the domain account a member of the local admin grou |
| 20:22.35 | Naikrovek | group* |
| 20:22.39 | luca`gervasi | i have a two channel license for g729...how can i test it ? :D |
| 20:22.40 | Katty | it is NOT on a domain |
| 20:22.43 | Naikrovek | ah |
| 20:22.45 | Naikrovek | okay |
| 20:22.47 | Naikrovek | why not |
| 20:22.55 | Katty | because it was taken off the domain. |
| 20:23.10 | Naikrovek | none of my machines leave the domain |
| 20:23.11 | Katty | obviously. |
| 20:23.20 | Katty | well this isn't one of your machines, now is it :P |
| 20:23.22 | Naikrovek | yeah but why was it taken off |
| 20:23.30 | Naikrovek | l0phtcrack can find your password, btw |
| 20:23.32 | Katty | because it's going in a back Shipping office which has no network |
| 20:23.33 | loather-work | Naikrovek: bad idea, generally. you don't want users with admin privileges running amok on the system. That's one of the biggest security problems with windows. |
| 20:23.56 | Katty | loather-work: we do local admin on occasion |
| 20:23.57 | loather-work | if i ever have to admin windows domains, IT does all the software installs. No exceptions. |
| 20:24.04 | Katty | loather-work: but even i'm not a domain admin on my account |
| 20:24.12 | Naikrovek | loather-work: thanks for twisting my point into a knife and stabbing me with it. you're not understanding me |
| 20:24.13 | loather-work | :D |
| 20:24.34 | Katty | is using a lil linux boot disk to reset the stuff |
| 20:24.53 | Katty | if they want to take it off the domain, that is their choice. |
| 20:24.56 | Katty | whatever pays the bills. |
| 20:25.06 | Katty | i will be happy to fix it for a fee. |
| 20:26.19 | Naikrovek | disjoining a computer from the domain that you're responsible for is a bad idea and should be avoided. and no one should be an admin with their regular day-to-day account |
| 20:26.28 | Naikrovek | and no end user should know the admin password of any machine ever |
| 20:26.41 | Qwell | # whoami |
| 20:26.42 | Qwell | root |
| 20:27.06 | Katty | Naikrovek: where they are moving it does not have a network drop. |
| 20:27.15 | Katty | Naikrovek: therefor, attaching it to a domain will not be possible. |
| 20:27.22 | Katty | Naikrovek: it is their computer, they can do wahtever they want with it |
| 20:27.33 | Katty | Naikrovek: and in an economy like this you thank your lucky starts there is cash flow (= |
| 20:27.42 | Naikrovek | Katty: then make sure you're not responsible for it, they can dig out their own password |
| 20:27.53 | Katty | Naikrovek: it is my job to Help them. |
| 20:28.00 | Katty | Naikrovek: not to shove them away and say sorry, better luck next time |
| 20:28.05 | Naikrovek | then make sure you can help them. |
| 20:28.05 | Katty | Naikrovek: that equals no cash flow :P |
| 20:29.17 | Naikrovek | let me tell you something learned from hard experiences. working for idiots, no matter how much cash it brings in, is never rewarding nor worth it. |
| 20:29.49 | Naikrovek | whatever |
| 20:29.52 | Katty | i'm not working for idiots. |
| 20:29.57 | Katty | this is a client, and i am helping them. |
| 20:29.59 | Katty | as i do all my clients. |
| 20:30.07 | Naikrovek | koay |
| 20:30.09 | Naikrovek | okay |
| 20:30.18 | Katty | and they love me for it (= |
| 20:30.36 | Naikrovek | i would not do your job for $10,000,000 a year. |
| 20:30.40 | Naikrovek | and yes i mean that |
| 20:30.43 | Katty | that's you (= |
| 20:30.53 | Katty | likes helping people. |
| 20:31.09 | loather-work | at that rate i'd do it for a year and then retire and live off the interest. :) |
| 20:31.33 | loather-work | you can live quite comfortably on around $2,000,000. |
| 20:31.59 | TJNII | loather-work: In this economy, you can't even do that. 0.01% interest rates! (Or whatever stupid low value it is.) |
| 20:32.37 | loather-work | TJNII: you can always find someone to give you 5-6% if you know what to look for. |
| 20:32.45 | luca`gervasi | ..wow... skype for asterisk actually works :D |
| 20:33.24 | loather-work | heh, that's my next project |
| 20:33.47 | luca`gervasi | uhm... i'm offline in my non-business account |
| 20:33.52 | luca`gervasi | that's odd |
| 20:34.23 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 20:37.11 | *** part/#asterisk nny (n=scott@64.203.239.83) |
| 20:40.30 | ChannelZ | yes, yes it does |
| 20:41.16 | eppigy | TRABAJO |
| 20:46.43 | torrancew | has anyone here configured Linksys/Sipura SPA942 IP phones with asterisk? |
| 20:48.53 | *** join/#asterisk st91ang (n=powellju@nat/digium/x-tcyuujjdimzxweyc) |
| 20:50.25 | titter | hi |
| 20:51.41 | luca`gervasi | good night! |
| 20:52.11 | *** join/#asterisk knctrnl (n=aembrey@nat/digium/x-vlznwsdungqudyrd) |
| 20:55.04 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d24:8246:d470:a9a7) |
| 20:57.18 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 20:57.18 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:58.14 | bkruse | needs a CACERT for two domains |
| 21:00.59 | sun28 | bkruse: rapidsslonline.com $18 |
| 21:01.08 | ecrane | ~book |
| 21:01.09 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:01.24 | ecrane | torrancew: There is a section in the book for that phone |
| 21:05.30 | bkruse | sun28: Ty, I need to get a couple by Monday, that should work? |
| 21:06.18 | sun28 | yep |
| 21:06.32 | bkruse | sun28: Great, ty sir |
| 21:06.37 | sun28 | ^__^ |
| 21:07.02 | bkruse | Wow, I was about to jbot: sun28++ but I guess I've been out of the loop too long :) |
| 21:11.20 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 21:12.19 | *** part/#asterisk st91ang (n=powellju@nat/digium/x-tcyuujjdimzxweyc) |
| 21:14.19 | *** part/#asterisk esaym153 (n=esaym153@cpe-24-174-176-203.satx.res.rr.com) |
| 21:15.38 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 21:16.34 | Chodorenko | Hello All |
| 21:17.07 | Chodorenko | please consult me howe to i can setup asterisk for support SRTP ? |
| 21:17.37 | torrancew | ecrane: my exact issue isn't covered there - i think it's hijacking caller id and not displaying what asterisk broadcasts |
| 21:18.17 | torrancew | ecrane: for example, the linphone softphone will show real numbers that dial in, but anything coming from my trunk displays as "asterisk" on the spa942s |
| 21:18.18 | loather-work | is there any known workaround for the "stays in-use" problem when a Local/1234@context/n added as a dynamic queue member transfers a call? |
| 21:20.17 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:23.23 | Katty | looks in |
| 21:26.35 | Chodorenko | loather-work: i not understand howe to you can add Local/1234@context/n as Dynamic member ? |
| 21:26.50 | Deeewayne | pats Katty on the head |
| 21:27.31 | loather-work | Chodorenko: it's easy, add it dynamically with AddQueueMember(queuename,Local/1234@context/n) |
| 21:27.45 | manxpower | ~answers |
| 21:27.46 | infobot | well, 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 |
| 21:28.39 | Katty | Deeewayne: :> |
| 21:29.48 | Chodorenko | loather-work: mey be it`s not true path for add user in queue |
| 21:30.28 | Chodorenko | loather-work: add as SIP IAX or other protocol and open bug on bugs.digium.com |
| 21:30.39 | loather-work | it's already a known issue. |
| 21:30.45 | loather-work | just wondered if anyone knew a workaround. |
| 21:32.28 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 21:33.03 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 21:34.08 | Chodorenko | loather-work: some is understand why adding in queue local exten , by that in the local contest can be multiple users such as dial (sip/100 & IAX2/2000) |
| 21:34.19 | Chodorenko | *not understand |
| 21:35.20 | loather-work | I have multiple agents per desk, and agents aren't tied to a specific desk, either. |
| 21:35.43 | loather-work | so I need to track agents by their agent ID instead of an extension |
| 21:35.51 | loather-work | otherwise my metrics get all screwed up |
| 21:36.42 | Chodorenko | use agent mechanism , agent add queue by press dial pad key |
| 21:36.58 | loather-work | The agent mechanism is horribly broken. |
| 21:37.24 | jblack | you could write your own as an agi. I've done that a couple times; it's not that hard |
| 21:37.28 | Chodorenko | loather-work: ? |
| 21:37.55 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 21:37.55 | loather-work | no, it's not hard ... i've got it almost fully implemented in dialplan logic. |
| 21:38.48 | loather-work | The deficient queueing system is one of my bigger gripes with asterisk; working around it is annoying. oh well. |
| 21:39.26 | Chodorenko | loather-work: you may be use dialplan variables as ${CHANNEL} for add member in queue |
| 21:41.30 | Chodorenko | loather-work: for example exten =>1111,1,AddQueueMember(queuename,${$CHANNEL}) |
| 21:42.08 | Chodorenko | loather-work: for example exten =>1111,1,AddQueueMember(queuename,${CHANNEL}) |
| 21:43.46 | jpcansa | hi, what do i have to configure if my SIP clients are outside behind nat? i mean on the phone configuration, what are the main settings? |
| 21:43.56 | loather-work | Chodorenko: doesn't work the way i need it to. |
| 21:44.04 | Chodorenko | loather-work: member add in queue from phone dialed number |
| 21:44.39 | loather-work | Chodorenko: take a look at http://chainsaw.drjays.com/~khudson/callcenter-context.conf , callcenter-extensions.conf , callcenter-macros.conf and that's what I'm doing. |
| 21:44.55 | loather-work | username speak / password friend |
| 21:45.15 | Chodorenko | jpcansa: STUN , nat setting in sip.conf |
| 21:45.17 | torrancew | ah |
| 21:45.54 | *** join/#asterisk jadl_ (n=jadl@89.130.82.210) |
| 21:45.56 | torrancew | if you call multiple extensions in one Dial () [Dial(${EXT1}&${EXT2},20)], would that affect inbound cid? |
| 21:46.28 | loather-work | not that i know of. I do that to the phone on my desk at both offices |
| 21:46.34 | loather-work | CID passes through just fine |
| 21:47.08 | torrancew | my only other guess would be that our provider borked our CID |
| 21:47.23 | torrancew | it once worked fine, but now everything shows as coming from "asterisk" |
| 21:47.45 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:47.47 | torrancew | and the only major change was that our pstn provider came and added more lines and tweaked our hunt group today |
| 21:48.00 | loather-work | they probably broke something. |
| 21:48.14 | loather-work | ISDN or analog? |
| 21:48.18 | torrancew | i've been banging my head against the wall all day trying to get to the bottom of it |
| 21:48.20 | jpcansa | Chodorenko: i have 2 nics configured on my box, 1st with public ip, and 2nd with LAN ip, is tehre something i have to configure in that scenario, SIP clients on the outside cannot communicate to SIP clients on the inside. |
| 21:48.20 | torrancew | analog |
| 21:48.35 | Chodorenko | jpcansa: localnet in sip.conf |
| 21:48.37 | loather-work | check and see if the provider changed the CID signalling type on the lines. |
| 21:48.49 | torrancew | loather-work: how might i do that? |
| 21:48.56 | loather-work | call them and ask :) |
| 21:49.00 | torrancew | kk |
| 21:49.17 | loather-work | you could also add some debug statements in your incoming dialplan |
| 21:49.23 | torrancew | such as? |
| 21:49.27 | jadl_ | I can not configure asterisk for Yacom voip sip and x-lite, I need help please |
| 21:49.27 | loather-work | check it for caller ID information and see what your call flow looks like |
| 21:49.35 | torrancew | syntax? |
| 21:50.25 | Chodorenko | jpcansa: and "canreinvite=no" in dialplan |
| 21:50.26 | loather-work | exten => 5595,1,NoOp("Got call from ${CALLERID(name)} at ${CALLERID(num)}") |
| 21:50.27 | jadl_ | |
| 21:50.27 | jadl_ | I have read, but can not find the problem |
| 21:50.27 | jpcansa | Chodorenko, localnet only, no externip right? |
| 21:51.05 | Chodorenko | jpcansa: external IP setup to external interface |
| 21:52.01 | jpcansa | Chodorenko: ok thx, let try that |
| 21:53.35 | Chodorenko | jpcansa: "canreinvite=no" disable communicate beetwen phone directly , all media transfer with server |
| 21:53.50 | torrancew | loather-work: no info in the debug |
| 21:53.55 | torrancew | blank name, blank num |
| 21:54.19 | loather-work | ok, then that means that your channel driver doesn't have the information when the call is originated |
| 21:54.46 | torrancew | loather-work: how could i go about seeing if that's my problem or theirs? |
| 21:54.47 | loather-work | so either the CID isn't being sent by the telco, or the hardware isn't configured to properly read it |
| 21:54.56 | torrancew | esp if it was working last week |
| 21:55.04 | Chodorenko | loather-work: i not fully understand you dialplan, in my view its not good way |
| 21:55.21 | jadl_ | took several days to enter the channel and it always helps, but I can not solve the problem, please anyone can help me and wants? |
| 21:55.46 | loather-work | well, if they installed lines and messed with your hunt group the CID configuration could have been wiped out at the switch. I'd be willing to be that's what the problem is. |
| 21:55.47 | torrancew | loather-work: chan_dahdi is set to use cid, as well as use what's provided |
| 21:55.48 | jadl_ | I don't speak English, I use Google sometimes |
| 21:55.55 | torrancew | thanks so muc |
| 21:55.57 | torrancew | much* |
| 21:56.00 | loather-work | welcome :) |
| 21:56.03 | torrancew | my money's with you |
| 21:56.21 | Chodorenko | jadl_: i talk by Russian :) |
| 21:56.36 | jadl_ | xd |
| 21:56.47 | *** part/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 21:56.58 | *** join/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 22:01.01 | jpcansa | Chodorenko: so do i configure canreinvite=no in sip.conf for my external sip clients? |
| 22:01.42 | Chodorenko | jpcansa: in GENERAL section |
| 22:01.57 | Chodorenko | jpcansa: for all |
| 22:02.18 | jadl_ | must be easy for any of you really need help |
| 22:03.12 | [TK]D-Fender | jadl_: And in the past few days you haven't SHOWN us the problem. |
| 22:03.15 | [TK]D-Fender | ~pb |
| 22:03.15 | 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 |
| 22:03.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 22:04.36 | jadl_ | is the same problem but not solved |
| 22:05.24 | jadl_ | I do not know that show |
| 22:06.31 | [TK]D-Fender | jadl_: You are showing no debug and no description of the problem |
| 22:06.40 | jadl_ | |
| 22:06.40 | torrancew | loather-work: yep, they broke it |
| 22:06.40 | jadl_ | when I make a call using x-lite, the operator tells me, now your call does not go through with the prefix dialing |
| 22:06.53 | [TK]D-Fender | jadl_: what "operator"? |
| 22:07.06 | jadl_ | yacom |
| 22:07.11 | sun28 | Chodorenko: me too :) |
| 22:07.12 | kfife | any ideas as to why alwaysauthreject=yes, but still sending back 'Wrong password' on 1.4.24.1 instead of '401 unauthorized'? |
| 22:07.16 | [TK]D-Fender | jadl_: And what is "prefix dialing"? |
| 22:07.50 | jadl_ | "prefijo marcado" |
| 22:08.12 | jadl_ | I don't know |
| 22:08.14 | [TK]D-Fender | jadl_: Doesn't help.... the term doesn't have a fixed meaning |
| 22:08.36 | [TK]D-Fender | jadl_: Again, WHAT is giving this message? |
| 22:08.39 | Chodorenko | sun28: Russian ? why not present in #asteriskru |
| 22:08.47 | sun28 | hmm |
| 22:09.34 | jadl_ | as show debug and description information? |
| 22:10.03 | jadl_ | hago: core set (debug , verbose) 9 |
| 22:10.09 | Chodorenko | [TK]D-Fender: Howe to setup asterisk for support SRTP without TLS ? |
| 22:10.13 | jadl_ | I'm sorry |
| 22:10.29 | loather-work | torrancew: :D figured :) |
| 22:10.40 | torrancew | they borked the hunt as well |
| 22:11.17 | jadl_ | do or make (I don't know): core set (debug , verbose) 9 |
| 22:11.30 | *** join/#asterisk gilevy (n=gil@c-24-10-28-163.hsd1.ca.comcast.net) |
| 22:11.41 | jadl_ | and what else? |
| 22:11.48 | [TK]D-Fender | jadl_: "sip set debug", "core set verbose 10" |
| 22:12.07 | gilevy | hi i've never used asterisk before and i was wondering how i can set up my gizmo5 number with asterisk |
| 22:12.07 | jadl_ | ok |
| 22:12.11 | jadl_ | and what else? |
| 22:13.37 | [TK]D-Fender | ~book |
| 22:13.38 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:13.43 | [TK]D-Fender | gilevy: ^^^ |
| 22:14.01 | jadl_ | sip set debug |
| 22:14.01 | jadl_ | No such command 'sip set debug' (type 'help sip set debug' for other possible commands) |
| 22:14.44 | [TK]D-Fender | jadl_: "sip debug on" |
| 22:14.51 | gilevy | thank you [TK]D-Fender |
| 22:15.05 | jadl_ | sip debug on |
| 22:15.05 | jadl_ | No such command 'sip debug on' (type 'help sip debug on' for other possible commands) |
| 22:15.27 | [TK]D-Fender | jadl_: "sip set debug on" |
| 22:15.45 | jadl_ | ok |
| 22:15.51 | jpcansa | Chodorenko: i have everything set up, but i can hear only one way |
| 22:16.15 | [TK]D-Fender | jpcansa: pastebin your sip.conf and the SIP DEBUG of a failed call. |
| 22:16.24 | tuxcrafter | hmm wierd i still cant get my nat settings right for ekiga |
| 22:16.38 | tuxcrafter | i got two voip phones here |
| 22:16.46 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:17.22 | tuxcrafter | so i changed the ekiga settigs to listen_port = 7060 |
| 22:17.23 | tuxcrafter | upd_port_range = 7000:7100 |
| 22:17.34 | tuxcrafter | and also made the forwarding in iptables |
| 22:17.42 | tuxcrafter | i can call the phone |
| 22:17.45 | [TK]D-Fender | tuxYou don't need forwarding.... |
| 22:17.51 | [TK]D-Fender | tuxcrafter: You don't need forwarding.... |
| 22:18.21 | Chodorenko | jpcansa: for all clients outside you network set "nat=yes" in peer setup in sip.conf , setup "localnet" and "externip" in in sip.conf |
| 22:18.26 | jadl_ | now what? |
| 22:18.41 | tuxcrafter | [TK]D-Fender: http://debian.pastebin.com/m5555962d |
| 22:18.47 | tuxcrafter | i meand prerouting |
| 22:19.05 | jadl_ | I ask because I do not know |
| 22:19.08 | tuxcrafter | 406@an4705.voipgate.nl |
| 22:19.12 | tuxcrafter | i can call the number |
| 22:19.14 | tuxcrafter | it rings |
| 22:19.20 | tuxcrafter | and i can answer |
| 22:19.24 | tuxcrafter | but there is no audio |
| 22:19.44 | tuxcrafter | when i call from teh 406 extention i do get audio |
| 22:19.58 | jadl_ | do not want to be rude |
| 22:20.18 | tuxcrafter | i tried the extentions on a single sip phone and there it did work :) |
| 22:20.29 | tuxcrafter | but know i am trying to get both systems working |
| 22:21.57 | [TK]D-Fender | tuxYou do NOT need to port forward for a phone. |
| 22:22.04 | *** part/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 22:22.17 | [TK]D-Fender | jadl_: Pastebin a call attempt |
| 22:22.18 | [TK]D-Fender | ~pb |
| 22:22.19 | 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 |
| 22:22.26 | jadl_ | ok |
| 22:22.44 | *** join/#asterisk moos3 (n=rgenthne@cpe-76-179-253-210.maine.res.rr.com) |
| 22:22.59 | moos3 | is there away from the cli to see current calls in session? |
| 22:23.16 | jpcansa | [TK]D-Fender: http://pastebin.com/mcdac4b this time it worked, but for some reason it doesnt sometimes, do you see something wrong?? |
| 22:23.18 | ChannelZ | core show channels |
| 22:23.19 | [TK]D-Fender | moos3: "core show channels" |
| 22:23.45 | moos3 | sweet thanks |
| 22:24.12 | kfife | any ideas as to why alwaysauthreject=yes, but still sending back 'Wrong password' on 1.4.24.1 instead of '401 unauthorized'? |
| 22:24.17 | jadl_ | nothing |
| 22:24.33 | [TK]D-Fender | jpcansa: that is not a complete call. |
| 22:24.39 | *** part/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 22:24.40 | moos3 | is there a way to show that in a web page? |
| 22:24.49 | [TK]D-Fender | jadl_: if You see nothing then packets aren't getting from your phone to * |
| 22:25.08 | [TK]D-Fender | jadl_: So you have either a firewall/routing problem, or your client is misconfigured |
| 22:25.26 | [TK]D-Fender | moos3: Yes. Go write a script for a web server |
| 22:25.41 | Chodorenko | kfife: jpcansa: in : http://pastebin.com/mcdac4b i not can view external IP |
| 22:25.56 | jadl_ | because I respode an operator? |
| 22:26.02 | tuxcrafter | [TK]D-Fender: what is needed then to make incommming calls with audio work? when there is no forwarding needed? |
| 22:26.03 | Chodorenko | jpcansa: jpcansa: in http://pastebin.com/mcdac4b i not can view external IP |
| 22:26.19 | ChannelZ | moos3: http://www.asternic.org/ |
| 22:26.37 | [TK]D-Fender | tuxcrafter: CLIENTS don't need forwarding, only * if it is behind NAT |
| 22:26.41 | jpcansa | Chodorenko: do i need that even if my box is not behind nat? |
| 22:26.44 | moos3 | cool thanks |
| 22:27.12 | tuxcrafter | [TK]D-Fender: i am on a NAT |
| 22:27.22 | [TK]D-Fender | tux.... |
| 22:27.32 | [TK]D-Fender | tuxcrafter: ... WTF is "I"? |
| 22:27.46 | tuxcrafter | wel the my systems are :) |
| 22:27.50 | [TK]D-Fender | ........ |
| 22:28.09 | *** part/#asterisk moos3 (n=rgenthne@cpe-76-179-253-210.maine.res.rr.com) |
| 22:28.16 | [TK]D-Fender | tuxcrafter: Got any more non-descript bits to share? |
| 22:28.37 | tuxcrafter | internet | debian iptablefirewal with NAT | phone01, desktop with ekiga |
| 22:29.01 | tuxcrafter | the asterisk server is not on my network |
| 22:29.15 | [TK]D-Fender | tuxWhere the hell is it on that line? |
| 22:29.52 | Chodorenko | jpcansa: external IP use for clients marked as NAT |
| 22:29.53 | tuxcrafter | asterisk is on an other server/netwerk with a fixed ip |
| 22:30.11 | [TK]D-Fender | tuxdraw a network map that actually includes both ends. |
| 22:30.45 | [TK]D-Fender | [18:26]<jpcansa>Chodorenko: do i need that even if my box is not behind nat? <- no |
| 22:31.05 | [TK]D-Fender | jpcansa: Regrettably over half the info you've been given is incorrect |
| 22:32.26 | jpcansa | [TK]D-Fender: what do i have wrong? |
| 22:32.45 | tuxcrafter | askterisk server [an4705.voipgate.nl] | internet | debian server[84.245.3.195] with iptable and nat | fully working sip phone01 [192.168.1.62], debian desktop with ekiga that can be called but does not have audio when called [192.168.1.236] |
| 22:32.51 | tuxcrafter | [TK]D-Fender: ^ |
| 22:33.12 | tuxcrafter | and ekiga listens to 7070 and phone01 listens to 5060 |
| 22:33.19 | [TK]D-Fender | tuxcrafter: Whats at that 2nd IP? |
| 22:33.35 | [TK]D-Fender | tuxand WHERE is that 2nd IP? |
| 22:33.51 | tuxcrafter | em how do you mean? |
| 22:33.51 | Chodorenko | [TK]D-Fender: if you need give packet from external IP you need setup NAT option in peer ... no ? |
| 22:34.13 | Chodorenko | [TK]D-Fender: else server answer you from internal Ip |
| 22:34.27 | [TK]D-Fender | Chodorenko: Not in his case. |
| 22:34.34 | jadl_ | thanks, but I do not know that to do |
| 22:34.56 | jadl_ | I'm thinking |
| 22:35.01 | [TK]D-Fender | jadl_: Check your firewalls |
| 22:35.03 | jadl_ | but... |
| 22:35.07 | jadl_ | how |
| 22:35.09 | jadl_ | ? |
| 22:35.13 | *** join/#asterisk errotan (n=errotan@a1719.adsl.pool.eol.hu) |
| 22:35.19 | [TK]D-Fender | jadl_: "iptables --list" <- from OS CLI |
| 22:35.39 | jpcansa | [TK]D-Fender: what is that i have incorrect? |
| 22:36.11 | [TK]D-Fender | jpcansa: pastebin your sip.conf and the call attempt. Also, NO FORWARDING on the remote side. |
| 22:36.33 | p3nguin_ | jadl_: iptables -L -nv |
| 22:38.54 | tuxcrafter | [TK]D-Fender: i think i am going to buy an aditional hardware sip phone, i dont know what ekiga is doing or not doing |
| 22:39.27 | p3nguin_ | tuxcrafter: Doesn't sip debug provide any useful information? |
| 22:39.48 | tuxcrafter | p3nguin_: well it provides an awfull log of data |
| 22:39.51 | [TK]D-Fender | tuxcrafter: I think you're jumping to conclusions, questioning advise, providing piecemeal information, and litte to no debug. |
| 22:39.55 | tuxcrafter | and i dont know what to look for |
| 22:40.01 | [TK]D-Fender | tuxcrafter: But feel free to buy whatever you want. |
| 22:40.09 | jadl_ | http://paste.debian.net/49799/ |
| 22:40.28 | jadl_ | all well, no? |
| 22:40.31 | [TK]D-Fender | jadl_: Where is *, and where is your client? |
| 22:41.11 | jadl_ | I do not understand |
| 22:41.45 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 22:41.45 | [TK]D-Fender | jadl_: Describe the networking between * and your client |
| 22:41.45 | p3nguin_ | jadl_: Is your phone on that computer (a softphone)? |
| 22:41.48 | tuxcrafter | [TK]D-Fender: http://debian.pastebin.com/d626c812 |
| 22:42.34 | jadl_ | yes, is x-lite |
| 22:43.09 | tuxcrafter | that is the sip debug when i make a call from phone01[sip:31107079913@sip.tweakphone.nl] to the ekiga softphone [sip:406@an4705.voipgate.nl] |
| 22:43.31 | jpcansa | [TK]D-Fender: the complete call is not showed in console?? |
| 22:43.48 | tuxcrafter | it rings at the ekiga software and i can pick it up but then there is no audio |
| 22:43.59 | ecrane | what is the 1.6.1 equivalent of 'dialplan reload'? |
| 22:44.00 | tuxcrafter | the other way ekiga to phone01 it work perfectly |
| 22:45.18 | p3nguin_ | jadl_: Where's Asterisk in this scenario? |
| 22:45.29 | p3nguin_ | ecrane: dialplan reload |
| 22:45.46 | tuxcrafter | what does this mean: # |
| 22:45.47 | tuxcrafter | Audio is at 84.243.247.215 port 11834 |
| 22:46.13 | p3nguin_ | It means the RTP is using port 11834 on the IP address 84.243.247.215 |
| 22:47.29 | tuxcrafter | Peer audio RTP is at port 84.245.3.195:7010 |
| 22:48.59 | [TK]D-Fender | [18:41]<[TK]D-Fender>jadl_: Describe the networking between * and your client |
| 22:49.14 | jadl_ | how? |
| 22:50.49 | jpcansa | [TK]D-Fender: is this complete? http://pastebin.com/d2352514 |
| 22:51.37 | [TK]D-Fender | jpcansa: No. |
| 22:52.19 | [TK]D-Fender | jadl_: I'm sorry. The language barrier is clearly to big. Perhaps someone else can assist you. |
| 22:52.25 | jpcansa | [TK]D-Fender: dunno how to get it then, sorry |
| 22:52.57 | [TK]D-Fender | jpI don't se the call come IN. You show as of the first dialplan line being executed. Where the hell is the NEGOTIATION leading up to that? |
| 22:53.11 | jadl_ | I'm sorry |
| 22:53.17 | jadl_ | thanks |
| 22:53.38 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 22:56.11 | jadl_ | bye |
| 22:56.23 | jpcansa | [TK]D-Fender: i´m calling out to a dahdi channel, let me make it between extensions |
| 22:56.47 | [TK]D-Fender | jpcansa: Stop changing circumstances involved in the test |
| 22:57.03 | [TK]D-Fender | jpcansa: How are you supposed to pin down the probelm when you keep moving the damn target? |
| 22:57.47 | tuxcrafter | hmm i cant find anything wrong yet, having two sip phone with incommming calls seems a bit hard |
| 22:57.48 | jpcansa | [TK]D-Fender: is happening the same in both scenarios |
| 22:58.18 | [TK]D-Fender | tuxcrafter: have you disabled the forwarding? |
| 22:58.56 | Chodorenko | To All ____if you have manual by setup STRP without TLS in Asterisk please give link ____________ |
| 22:59.03 | jpcansa | [TK]D-Fender: i just dont know why i cannot get the complete call to show it, i copied everything since the moment i dialed to the moment i hanged up |
| 22:59.43 | [TK]D-Fender | jpcansa: You haven't enabled GLOBAL SIP debug or you aren't paying attention |
| 23:02.07 | tuxcrafter | [TK]D-Fender: jups all forwardings for the the ekiga client are disabled |
| 23:02.40 | tuxcrafter | i will continue my quest for having two working sip phones on the same network an other time :D |
| 23:02.46 | tuxcrafter | it is getting very late now |
| 23:04.07 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 23:07.11 | jpcansa | [TK]D-Fender: http://pastebin.com/m2dcde3d2 |
| 23:09.59 | p3nguin_ | Can't have two operable SIP phones on the same network??? |
| 23:10.42 | Chodorenko | jpcansa: where external IP Option ? |
| 23:11.22 | *** join/#asterisk sier (n=sier@unaffiliated/sier) |
| 23:11.24 | sier | Hi! |
| 23:16.01 | jpcansa | Chodorenko, is not true i have to set up external ip only if my box is behind NAT with a private IP? |
| 23:16.30 | dlynes | Does anyone know where I might be able to get Indianapolis dids of a decent voice quality? |
| 23:16.53 | dlynes | Preferably from the downtown area, or Hamilton County? |
| 23:16.53 | p3nguin_ | dlynes: Check VoIP.ms and Flowroute. |
| 23:16.59 | dlynes | p3nguin_: thanks |
| 23:17.21 | jpcansa | flowroute is good |
| 23:17.21 | p3nguin_ | dlynes: They provide good quality services, but I can't say if they have DIDs available for your area. |
| 23:18.00 | Chodorenko | jpcansa: http://pastebin.com/m2dcde3d2 i can not see this option |
| 23:18.43 | Chodorenko | external IP use for communicate to all clients not present in localnet |
| 23:18.57 | jpcansa | Chodorenko: i havent set it up because my box is outside NAT |
| 23:19.03 | jpcansa | am i wrong? |
| 23:19.38 | Chodorenko | jpcansa: external IP use for communicate to all clients not present in localnet , you client in localnet ? |
| 23:19.43 | Chodorenko | no |
| 23:19.52 | Chodorenko | please setup external IP |
| 23:20.09 | p3nguin_ | jpcansa: If Asterisk has a public IP address, there's no reason to use NAT settings on it. |
| 23:20.12 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 23:20.36 | p3nguin_ | jpcansa: You might want to use NAT settings for clients/peers/users if they are behind NAT, though. |
| 23:22.23 | hardwire | https://issues.asterisk.org/view.php?id=7403 <- anybody familiar with this |
| 23:22.26 | jpcansa | p3nguin_: not in asterisk but in SIP clients from the outside |
| 23:22.48 | hardwire | was a patch committed? I certainly don't see committed changes |
| 23:22.52 | jpcansa | p3nguin_: mean nat settins |
| 23:23.25 | p3nguin_ | jpcansa: I do not understand what you are saying. |
| 23:24.26 | Chodorenko | p3nguin_: +1 |
| 23:24.51 | jpcansa | <p3nguin_> jpcansa: If Asterisk has a public IP address, there's no reason to use NAT settings on it. |
| 23:26.07 | jpcansa | p3nguin_: no reason to use NAT settings on asterisk but is necesary on sip clients on the outside |
| 23:26.07 | Chodorenko | strange people, tell them what to do, and they stopped by, do not understand why? After all, many are long gone, these adventures |
| 23:27.00 | Chodorenko | I go Sleep , All bye |
| 23:27.17 | jpcansa | thanks for your help Chodorenko, night |
| 23:29.16 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:29.22 | scalex000 | Hello guy, good evening. I need to know how to record a call on call center. Step by Step (I setup static queue) |
| 23:34.27 | Katty | has dinner!!! |
| 23:34.38 | p3nguin_ | scalex000: Something like AgentMonitorOutgoing? |
| 23:35.12 | scalex000 | p3guin: I found MixMonitor() |
| 23:35.21 | Katty | where did mister madsen go :< |
| 23:35.30 | Katty | now i have no one to share my healthy dinner with |
| 23:35.43 | Katty | is saddened |
| 23:35.44 | p3nguin_ | scalex000: That's what I use for incoming calls, but are you looking to record all calls or only certain ones, and are you wanting to record incoming or outgoing? |
| 23:37.07 | scalex000 | p3nguin, I don't I can record outgoing, that interesting news for me but now I need incoming |
| 23:38.08 | p3nguin_ | scalex000: For all incoming calls, you can easily use MixMonitor(). |
| 23:38.11 | Katty | note to self: do NOT breathe apple juice |
| 23:39.10 | *** join/#asterisk moos3 (n=rgenthne@cpe-76-179-253-210.maine.res.rr.com) |
| 23:39.32 | moos3 | can someone help me figure out why my sip isn't working but my t1 is |
| 23:39.49 | scalex000 | p3nguin: Ok, but this is use in dialplan or queue. I need to specify the name of file. or its use for default |
| 23:39.49 | p3nguin_ | scalex000: http://pastebin.ca/1640932 |
| 23:40.46 | scalex000 | p3nguin: thanks |
| 23:41.00 | moos3 | can anyone help with this http://pastebin.ca/1640936 |
| 23:41.12 | moos3 | my offices SIP phones are down |
| 23:41.12 | p3nguin_ | scalex000: That creates file names such as the follwing: 20091023-8886114466.wav |
| 23:42.00 | manxpower | moos3: pastebin the output of "sip show peers" |
| 23:42.29 | scalex000 | p3nguin: wav this type of file occupy too much. |
| 23:42.51 | p3nguin_ | scalex000: Then change it to something else! |
| 23:43.04 | scalex000 | p3nguin: ok |
| 23:43.08 | p3nguin_ | scalex000: All you have to do is change the file type in the dialplan. |
| 23:43.12 | moos3 | k |
| 23:43.33 | p3nguin_ | scalex000: Change the extension in the MixMonitor command and it changes the file type. |
| 23:43.42 | moos3 | http://pastebin.ca/1640939 |
| 23:43.52 | scalex000 | p3nguin: I will do it |
| 23:43.58 | moos3 | we copied all the configs over from the old asterisk box to the new one |
| 23:44.03 | scalex000 | p3nguin: thanks for create an example. |
| 23:44.11 | manxpower | moos3: most of your phones are not registered (as shown by unspecified as the ip) |
| 23:44.17 | manxpower | Therefore you cannot call those phones. |
| 23:44.42 | moos3 | is there a way to force them to register from asterisk |
| 23:44.53 | p3nguin_ | You can call 8101 or 8060, though. |
| 23:44.56 | manxpower | moos3: no. |
| 23:45.06 | moos3 | k |
| 23:45.22 | manxpower | moos3: the entire POINT of registering is to tell Asterisk what the IP of the device is. |
| 23:45.43 | moos3 | what about it not using my sip trunk |
| 23:46.10 | manxpower | moos3: that's not in the pastebin |
| 23:46.23 | moos3 | k |
| 23:51.24 | Katty | omnomnomnomnoms on grilled cheese |
| 23:52.09 | moos3 | manxpower: thanks for the help everything is up and working |
| 23:52.30 | *** join/#asterisk TSM (n=the_soft@87.194.32.212) |
| 23:57.00 | gilevy | i'm having trouble connecting gizmo with asterisk for incoming has anybody done this? |
| 23:57.30 | p3nguin_ | manxpower: Good job! |
| 23:58.41 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 23:59.36 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.131.152.dynamic.adsl.gvt.net.br) |
