| 00:01.28 | *** part/#asterisk tclark (~TC@S01060080ad113f15.gv.shawcable.net) |
| 00:02.26 | *** join/#asterisk Darwin35 (~Darwin35@69-160-8-12.bflony.adelphia.net) |
| 00:02.48 | Darwin35 | here come the storms |
| 00:02.57 | *** join/#asterisk smmerf (~smmerf@7206-1-ric-64-83-28-82-serial.t1.cavtel.net) |
| 00:03.21 | Legend` | Jesster: my 7940s run 7.1 just fine |
| 00:03.57 | Jesster | really? all of ours won't |
| 00:04.04 | Jesster | and i've seen reports about it not working too |
| 00:04.13 | Gushi | Hey guys, what's the command I should be using on pure sip calls to listen for a user to put in a response? I'm using Background and it seems to be ignoring my tones. |
| 00:04.35 | Jesse- | do you have the dtmf type set right in sip.conf |
| 00:04.43 | Jesster | Legend`: we get a "Universal Aplication Loader ... Protocol Application Invalid" msg at bootup |
| 00:04.53 | Jesster | Legend`: and that's for 7.0 and 7.1 sip and mgcp |
| 00:04.59 | Legend` | Jesster: update your SIPDefault.cnf |
| 00:05.14 | Jesster | Legend`: for the image_version ? |
| 00:05.33 | Jesse- | gushi |
| 00:05.36 | *** join/#asterisk FuzzyCat (~ScaredyCa@e67171.upc-e.chello.nl) |
| 00:05.50 | Legend` | marconi:/var/tftp# cat OS79XX.TXT |
| 00:05.50 | Legend` | P003-07-1-00 |
| 00:05.50 | Legend` | marconi:/var/tftp# |
| 00:06.02 | Legend` | marconi:/var/tftp# cat SIPDefault.cnf |grep image |
| 00:06.03 | Legend` | image_version: "P0S3-07-1-00" |
| 00:06.03 | Legend` | marconi:/var/tftp# |
| 00:06.18 | Jesster | Legend`: is your tftp public ? |
| 00:06.22 | Legend` | nopes |
| 00:06.26 | Gushi | Yes Jesse? |
| 00:06.44 | Gushi | I just set it for inband and got a WHOLE LOT of Jun 9 20:07:29 WARNING[-1231733840]: dsp.c:1478 ast_dsp_process: Unable to process inband DTMF on 256 frames |
| 00:06.48 | Jesster | Legend`: the only difference I see is i had it without quotes, just like I do for 6.3 |
| 00:06.54 | *** join/#asterisk Basketci_Kizz (Present@dup-148-221-196-69.prodigy.net.mx) |
| 00:06.55 | Gushi | What's the default? |
| 00:07.01 | Jesse- | um |
| 00:07.03 | Gushi | I didn't have anything set in my main sip.conf |
| 00:07.16 | Jesse- | do sip show channel and it'll say |
| 00:07.22 | Gushi | although some of my sip phones had it set one way or the other. |
| 00:07.41 | Jesse- | if you call the voicemail app does that accept your tones? |
| 00:07.46 | Gushi | this is for calls handed off to me via broadvox, which doesn't seem to fall into a default one. |
| 00:08.21 | *** part/#asterisk nicolasg (~nicolas@ip-189.houseware.com.ar) |
| 00:09.07 | *** join/#asterisk zotzz (~zotzz@24.231.36.159) |
| 00:09.43 | *** join/#asterisk {{{SEVDA}}} (~FISTIK__@host96-74.pool8291.interbusiness.it) |
| 00:09.49 | *** join/#asterisk tclark (~TC@S01060080ad113f15.gv.shawcable.net) |
| 00:10.27 | *** join/#asterisk jmb-ct (~trillian@moose.staff.chagres.net) |
| 00:10.43 | jmb-ct | any ENUM folks around |
| 00:12.31 | Gushi | Well, I've tried all three modes. Two just don't work, one spews insane amounts of errors. |
| 00:12.51 | Gushi | From my internal sip phones voicemail works, but this is on a call coming from outside via Broadvox. |
| 00:13.04 | Gushi | And I'm not sure how they're passing dtmf, if at all. |
| 00:13.40 | Jesse- | gushi just do sip show channel whatever |
| 00:13.46 | Jesse- | and see if the dtmf mode matches up to your phone |
| 00:14.19 | Gushi | This isn't my phone. |
| 00:14.24 | Gushi | It's a call from the outside. |
| 00:18.10 | Gushi | Odd. It only fails with g729 |
| 00:18.56 | Gushi | I'll have to contact broadvox |
| 00:19.55 | Jesse- | well whatever its supposed to match up to |
| 00:19.57 | Jesse- | phone or non phone |
| 00:22.41 | *** join/#asterisk BabyAnGell^^^ (Baby__16@212.110.83.91) |
| 00:24.48 | *** join/#asterisk Girl^^14^^ (~aYCa_@modem-357.gorilla.dialup.pol.co.uk) |
| 00:24.49 | *** part/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 00:26.33 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 00:28.19 | *** join/#asterisk panny-chineese (panny-hotl@217.14.141.7) |
| 00:29.29 | *** join/#asterisk bobman (~bobman@68-169-194-223.agstme.adelphia.net) |
| 00:30.29 | file | bandwidth... where did you go |
| 00:30.37 | *** join/#asterisk Nokia (aSLan@host209-133.pool8251.interbusiness.it) |
| 00:30.43 | Darwin35 | Hummity justjumped up |
| 00:30.52 | *** join/#asterisk xXxGirL^ (~[She-uLe]@62.162.17.66) |
| 00:30.53 | Darwin35 | thumnderstorm must be close... |
| 00:30.55 | *** join/#asterisk MaCaKizi^20^ (~manosh007@219.95.153.56) |
| 00:31.28 | hardkey | exit |
| 00:31.44 | hardkey | estoy perdiendo la puta cabeza |
| 00:31.48 | hardkey | adiós |
| 00:31.58 | *** part/#asterisk hardkey (~pepe@213.149.232.172) |
| 00:41.46 | *** join/#asterisk Movies (Bios@66-42-121-140.lsan.dial.netzero.com) |
| 00:41.49 | PilotPTK-Home | It's raining like a BITCH here. |
| 00:44.26 | Connor | My SIP provider has freaking VAD turned on... Dang it.. What can I do about that? |
| 00:44.34 | Darwin35 | ok it just goit bitching cool breeze |
| 00:44.44 | Darwin35 | its going to downpour any min |
| 00:44.57 | Darwin35 | Pilot where are you |
| 00:45.10 | PilotPTK-Home | South East Michigan |
| 00:45.25 | Darwin35 | ok then we are going to get what our having |
| 00:45.33 | Darwin35 | I am in Wuffalo |
| 00:45.45 | PilotPTK-Home | yep..tomorrow morning..look out. |
| 00:46.01 | Gushi | Okay. |
| 00:46.05 | Darwin35 | no I think its comming now i just saw a flash |
| 00:46.16 | PilotPTK-Home | I always fly right over buffalo on the way to NyNy |
| 00:46.16 | Darwin35 | and now thunder |
| 00:46.28 | Darwin35 | lol yeah |
| 00:46.56 | Darwin35 | wow its pitch black and its not 9:30 |
| 00:47.10 | Darwin35 | ok this storms going to be hard and fast |
| 00:48.47 | *** join/#asterisk MicrofonDa_Mc (Syntrilliu@host33-172.pool8251.interbusiness.it) |
| 00:51.46 | *** join/#asterisk nhuisman (~intranete@hilo-library.IfA.Hawaii.Edu) |
| 00:51.57 | nhuisman | has anyone ever had problems with voicemail regarding phones rebooting? |
| 00:52.17 | nhuisman | If i beging to press random buttons after a bit the phone will stop responding and reboot |
| 00:52.29 | nhuisman | while in the voicemail menus that is |
| 00:53.06 | MustDie | nhuisman: you can't ask questions without specifiying model of phone at least. |
| 00:53.26 | nhuisman | i'm sorry bout that, its a ciso 7960 using SIP 6.3 |
| 00:53.34 | nhuisman | running CVS head from yesterday |
| 00:53.53 | MustDie | nhuisman: current version is 7.1 read cisco site |
| 00:54.04 | PilotPTK-Home | i've never seen the issue... but I don't often push random buttons on my 7960s for no good reason... |
| 00:54.05 | MustDie | they've discover some stuff related to rebooting, if you press buttons too fast |
| 00:54.13 | nhuisman | i had the problem there too |
| 00:54.16 | nhuisman | but i'll check again |
| 00:54.33 | MustDie | i think most of people here are using 7960 and they are pretty happy with it |
| 00:54.33 | nhuisman | even in 7.1 it reboots with buttons too fast? |
| 00:54.36 | MustDie | no |
| 00:54.42 | nhuisman | ok i'll upgrade and see |
| 00:54.50 | PilotPTK-Home | I Love my Polycom's more than my Cisco's ANY DAY. |
| 00:55.24 | PilotPTK-Home | although i do wish Polycom would get off their ass and release a new firmware with features they've been promising. |
| 00:55.54 | nhuisman | we are going to buy some of the wiresless cisco phones |
| 00:57.56 | nhuisman | its strange though because i had the same problem with rebooting i 7.1 |
| 00:57.57 | nhuisman | in |
| 00:58.41 | file | paper cuts for all! |
| 00:59.08 | nhuisman | k 7.1 loaded, now to press random buttons :) |
| 00:59.43 | nhuisman | yep |
| 00:59.48 | nhuisman | still wierding out |
| 00:59.57 | nhuisman | that time all i pressed was a single 4 |
| 01:00.22 | *** join/#asterisk smoke-one (~HeLLaS@62.162.235.142) |
| 01:02.22 | nhuisman | here is what is happening with 7.1 |
| 01:02.32 | *** join/#asterisk greg_work (greg@dsl-205-210-52-151.kingston.net) |
| 01:02.38 | nhuisman | i dial into voicemail, it starts listing options, if press any key it reboots the phone |
| 01:02.43 | rootbe | could someone try to call SIP:0700102030@voice.telephonix.be ? |
| 01:02.48 | rootbe | just installed asterisk |
| 01:02.53 | rootbe | and want to know if it works :) |
| 01:03.48 | rootbe | still haven't fine-tuned the codecs |
| 01:03.52 | rootbe | to might not work... |
| 01:04.18 | rootbe | s/to/so/ |
| 01:04.31 | nhuisman | sigh, this sucks |
| 01:05.16 | rootbe | nhuisman ? |
| 01:05.25 | nhuisman | will 7.1 load up on 7910's ? |
| 01:05.39 | Connor | nhuisman: No. |
| 01:05.49 | nhuisman | rootbe, Having problems with 7960 phones rebooting while in voicemail menus |
| 01:06.11 | *** join/#asterisk {Yalniz-f} (~fiGenn@ACC60B4A.ipt.aol.com) |
| 01:06.36 | nhuisman | using 7.1 they simply reboot as soon as a single number is pressed, in 6.3 it takes about 10-20 numbers |
| 01:06.49 | nhuisman | using cvs HEAD from yesterday |
| 01:06.53 | nhuisman | also happend with stable |
| 01:08.25 | nhuisman | havn't been able to find any info on it anywhere |
| 01:08.52 | *** join/#asterisk SuZannA__ (EMiLYa@c211-30-187-134.rivrw1.nsw.optusnet.com.au) |
| 01:08.55 | *** join/#asterisk ScaredyCat (~ScaredyCa@e67171.upc-e.chello.nl) |
| 01:11.26 | Gushi | Hrmmm, is there a way to, based on the inbound number of a call, just transfer to the top of a different context? |
| 01:11.40 | *** part/#asterisk Bentley (~rbc@dsl-mtryl-209-5-169-44-cgy.nucleus.com) |
| 01:12.08 | Gushi | I guess "goto" would do it. |
| 01:12.42 | ariel_ | rootbe I tryied your sip and just got number busy at ip address 193.109.62.141 |
| 01:12.47 | znoG | is it normal to experience echo from <PSTN> <-> <ISDN> <-> <asterisk> <-> <SIP Phone> ? |
| 01:13.14 | *** join/#asterisk Corydon76-home (gold@pcp08665860pcs.500ash01.tn.comcast.net) |
| 01:13.15 | znoG | the echo is on the SIP Phone side, according to the cisco echo analysis document the problem is on the far end (<PSTN>) but i somehow doubt it. |
| 01:14.51 | ariel_ | znoG echo is a problem even on PSTN to PSTN. There are many different ways to help with the echo. |
| 01:14.57 | file | 2 days to go... just 2 days |
| 01:15.05 | *** join/#asterisk ScaredyCat (~ScaredyCa@e67171.upc-e.chello.nl) |
| 01:15.10 | ariel_ | 2 days till what? |
| 01:16.13 | znoG | ariel_: what are some of those ways to help with the echo? |
| 01:17.33 | nhuisman | I have cisco 7960 phones using SIP 6.3, when in the voicemail menus after a dozen button presses, phone reboots. I have had these problems on asterisk 1.0 stable, and i am currently on latest dev sources. |
| 01:18.02 | Gushi | Hey guys, I'm getting this error: Registration from 'z100 <sip:z100@65.125.237.65>' failed for '65.125.237.243' |
| 01:18.13 | Gushi | But I'm pretty sure I have my sipura box configged right. |
| 01:18.44 | Umaro | Guys, i have a call queue setup, and the strategy is set to random, but whenever a call comes in, it goes to the first agent specified in the agents.conf for that group |
| 01:18.52 | ariel_ | znoG first try echocancel=yes there there is echocancelwhenbridged=yes |
| 01:18.54 | *** join/#asterisk FoXi (~kardelenn@209-193-18-38-cdsl-rb1.jnu.acsalaska.net) |
| 01:19.06 | Gushi | Nevermind I'm an idiot. |
| 01:19.24 | Umaro | I've got "member => Agent/:1" in my queues.conf for the queue, but it doesn't send calls to any other agents |
| 01:19.33 | znoG | ariel_: i tried echocancel=yes in capi.conf, that didn't help, there was no "echocancelwhenbridged" in the file.. i don't even know which options are accepted |
| 01:19.44 | nhuisman | Anyone ever had problems with cisco phones rebooting while in the asterisk voicemail menus? |
| 01:19.48 | znoG | i thought it was a zapata.conf thing |
| 01:20.40 | ariel_ | znoG yes it's a zapata.conf but since I have never used capi maybe it is also supported there. |
| 01:21.08 | smmerf | is this possible to add somethink to DIAL(..... | ??) if i prefer restrict caller id ? |
| 01:21.12 | ariel_ | nhuisman I have 2 7960 and there working just fine on head 4 days go. 6.3. |
| 01:21.19 | smmerf | something |
| 01:21.35 | nhuisman | try this, go into voicemail then press random buttons for about 2 mins |
| 01:21.38 | *** join/#asterisk ^-^Samantha^-^ (~sHe_17_@219.95.100.70) |
| 01:22.20 | ariel_ | nhuisman I have a boss that plays with one of the phones almost all day long. And he is aways in voicemail. |
| 01:22.29 | nhuisman | hmm |
| 01:22.47 | nhuisman | if i'm 7.1 it crashes after a single button press in the voicemail |
| 01:23.08 | *** join/#asterisk suma (~suma@81-86-221-193.dsl.pipex.com) |
| 01:23.26 | ariel_ | smmerf make a dial plan that sets the callerID = privete or what ever. If your PRI line acepts you setting up the caller id it will send that info out. |
| 01:24.41 | ariel_ | nhuisman I used it to call voice mail and just press every button even 2 or 3 at the same time very fast and it works.. Dam thing hates me now! I think I hit it too hard. |
| 01:24.42 | smmerf | i can add to sip.conf restrictcid=yes ... but for me is not good. i prefer into extension, and DIAL (something like |hidecid) :) |
| 01:25.36 | Gushi | Is there a way to just have a comment echoed out in the CLI, so that for example I can know what context I'm entering/leaving? |
| 01:25.58 | ariel_ | smmerf in the exten => _XXXX.,1,SetCAllerID(XXXXXX) |
| 01:26.24 | Gushi | noOp |
| 01:26.49 | ariel_ | Gushi a line in exten => _XXXXX,1,NoOp |
| 01:27.17 | smmerf | yes yes, i know but something is wrong, becouse my other phone says strange number :) |
| 01:27.17 | smmerf | aaaaa |
| 01:27.18 | smmerf | moment |
| 01:28.18 | hez | gaak, anyone here have exp with an iaxy? I plugged everything in (in the correct order) but it doesnt appear to be making a dhcp request |
| 01:29.34 | ariel_ | hez I think you need to plug it into a 10baseT 1/2 duplex first and then let it upgrade the firmware... I read something like that earlier. Mine worked out fine out of box. |
| 01:29.40 | smmerf | ariel_ as i said , other phone where i try make call 'says' number : 1 |
| 01:29.54 | hez | ariel_: ahhhhhhhh! |
| 01:30.04 | hez | ariel_: i dont think i have a 10mb hub ;( |
| 01:30.31 | ariel_ | hez ask kram mine worked out of the box. I have a netgear 10/100/1000 switch. |
| 01:30.33 | hez | ariel_: will it automaticly get the new firmware? |
| 01:30.53 | smmerf | ariel_ SetCallerID( ) says 1; SetCallerID( 123123123) says and working good = display 123123123 |
| 01:30.54 | ariel_ | if it finds one yes |
| 01:30.54 | hez | kram: ideas? |
| 01:30.57 | oelewapperke | how can one change the context the console is in ? |
| 01:31.24 | ariel_ | smmerf you need to set something up. not blank. |
| 01:31.25 | hez | ariel_: meaning i have to set up a tftpd and stick the firmware on it |
| 01:31.26 | hez | ? |
| 01:31.27 | kram | hez: about what? |
| 01:31.45 | hez | kram: plugged iaxy in but no dhcp request sent |
| 01:31.57 | ariel_ | Evening kram |
| 01:32.20 | *** join/#asterisk ScaredyCat (~ScaredyCa@e67171.upc-e.chello.nl) |
| 01:32.33 | *** join/#asterisk draconius (~j00@12-218-60-36.client.mchsi.com) |
| 01:32.36 | kram | evening ariel |
| 01:32.42 | kram | hez: are you getting link? |
| 01:32.57 | ariel_ | I need to go and see if I can find the key to the A/C box to turn the dam thing on. It's hot here at work at night... |
| 01:33.01 | hez | yeah =) (the cord works i was using it for a laptop) |
| 01:36.02 | kram | hez: then i have no explanation, unless it was programmed with a static ip |
| 01:36.20 | hez | kram: just got it in the mail from digium this afternoon |
| 01:36.30 | kram | well try another switch and setup then |
| 01:36.35 | kram | because every unit goes through test here |
| 01:36.39 | kram | also |
| 01:36.42 | kram | tcpdump -ni eth0 port 9999 |
| 01:36.46 | kram | you should see it spit out a message |
| 01:37.03 | *** join/#asterisk _ALEMKIZ_away (~metali_cc@62.162.235.118) |
| 01:37.38 | hez | it will try port 9999? (on which machine the dhcp server?) |
| 01:38.36 | kram | just on anything |
| 01:38.42 | hez | ack |
| 01:38.44 | kram | it's sent to the broadcast address |
| 01:38.46 | kram | if you don't see it |
| 01:38.50 | kram | then you have some link layer issue |
| 01:38.54 | kram | try a different hub or switch |
| 01:39.00 | kram | every single iaxy, without exception, goes through test |
| 01:39.01 | hez | it just didnt like the switch in my room, the other one works fine ;( |
| 01:39.07 | kram | which includes a functional test |
| 01:39.19 | kram | okay then it doesn't like the switch in your room |
| 01:39.35 | kram | so it works on one switch but not the one in your room? |
| 01:39.46 | hez | correct |
| 01:39.52 | file | kram is the man! |
| 01:39.57 | file | kram: muffin! |
| 01:40.00 | drumkilla | kram is my hero |
| 01:40.00 | kram | *nods* that could have been helpful to know at the beginning of the conversation |
| 01:40.08 | kram | well, use the other switch then |
| 01:40.11 | hez | kram: just found it out |
| 01:40.16 | kram | okay |
| 01:40.20 | hez | kram: well for know thats doable |
| 01:40.25 | file | mmm... IAXy |
| 01:40.26 | ariel_ | hez told you they worked out of the box... |
| 01:40.29 | hez | its not a very good long term solution though |
| 01:40.39 | drumkilla | same cable? |
| 01:40.42 | kram | what kind of switch ies it? |
| 01:40.43 | hez | ariel_: i trust you =) it just wasnt liking me |
| 01:40.45 | hez | drumkilla: yes |
| 01:41.05 | ariel_ | what switch do you have in the other room. |
| 01:41.16 | hez | kram: working one is a netgear, the other one is a surecom |
| 01:41.34 | drumkilla | can anything else on the surecom contact your dhcp server? |
| 01:41.40 | ariel_ | hez how many deep is it. |
| 01:41.44 | kram | okay, well it doesn't like it |
| 01:41.45 | file | kram: who is behind you? |
| 01:41.48 | kram | remember the iaxy is 10mbs only |
| 01:41.58 | kram | if you have something that doesn't like 10mbps that's a definite problem |
| 01:42.03 | kram | file: that's wil |
| 01:42.06 | hez | drumkilla: yeah, netgear pcmcia network card, grandstream BT100, and a intel eepro100 |
| 01:42.24 | hez | kram: BT100 is 10mb |
| 01:42.35 | evilbuny | kram: any plans to update chan_sip.c to turn srvlookup = 1 instead of 0? |
| 01:42.38 | hez | ariel_: ummm two i think |
| 01:42.51 | hez | ariel_: both switchs |
| 01:43.06 | *** join/#asterisk ScaredyCat (~ScaredyCa@e67171.upc-e.chello.nl) |
| 01:43.14 | hez | ariel_: no wait.... switch -> switch -> hub -> dhcp server |
| 01:43.19 | Darwin35 | ok kids I need photos of what you think the new GS 102d shold look like |
| 01:43.25 | Darwin35 | we need ideas |
| 01:43.25 | kram | we changed the default config |
| 01:43.27 | kram | but that's all |
| 01:43.28 | file | kram: btw, your chair is really ... odd |
| 01:43.32 | Darwin35 | nothing to fancy |
| 01:43.43 | evilbuny | kram: i noticed :) |
| 01:43.46 | kram | yah |
| 01:43.50 | Darwin35 | but needs to look dif then the bt 101/102 |
| 01:43.51 | znoG | has anyone implemented Call Forwarding as per http://www.junghanns.net/asterisk/page6.html ? |
| 01:43.55 | ariel_ | I guess you all have access to an online cam. |
| 01:43.56 | drumkilla | that chair is awesome |
| 01:44.01 | evilbuny | but i was asking if the default was going to be flicked over as well if the option isn't specified |
| 01:44.04 | file | I've never seen one like it |
| 01:45.00 | Darwin35 | znog your way behind thatas old |
| 01:45.07 | znoG | Darwin35: whats used now? :) |
| 01:45.16 | ariel_ | hez switch - switch - hub... wow problem hub... |
| 01:45.17 | *** join/#asterisk gage_man (~root@h00095b093ebb.ne.client2.attbi.com) |
| 01:45.22 | Darwin35 | so if you find a basic design you like point me to it |
| 01:45.58 | hez | ariel_: nope, its working plugged in to the second switch (eg switch -> hub -> dhcp server) |
| 01:46.00 | ariel_ | Darwin35 you are working on designing the phone? |
| 01:46.22 | ariel_ | hez no what I mean is a hub is a problem in general. |
| 01:46.39 | hez | ariel_: how so? |
| 01:46.53 | gage_man | anyone know this error |
| 01:46.56 | gage_man | *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Goto("Zap/1-1", "home|1") in new stack -- Goto (default,home,1) -- Executing NoOp("Zap/1-1", "") in new stack -- Executing Macro("Zap/1-1", "record-on|XXXXXXXXXXX|9787711755") in new stack -- Executing AGI("Zap/1-1", "set-timestamp.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi Failed to execute '/var/lib/asterisk |
| 01:47.17 | Darwin35 | yes |
| 01:47.25 | gage_man | how to fix it |
| 01:47.27 | ariel_ | hubs see all traffic. can't do QoS if you start an FTP it will take full band witdth at first. |
| 01:47.29 | Darwin35 | I am helpiing with it |
| 01:47.43 | gage_man | why do I get a NOOP |
| 01:47.48 | gage_man | help |
| 01:47.48 | *** join/#asterisk essobi___ (kstone@75.137.26.216.host.teledvance.com) |
| 01:47.52 | Darwin35 | znog did you get the paste |
| 01:47.59 | essobi___ | Anyone using SCCP? |
| 01:48.00 | oelewapperke | how do you transfer a call to a certain peer ? |
| 01:48.05 | oelewapperke | in extensions.conf ... |
| 01:48.08 | gage_man | I use to use that crap |
| 01:48.28 | essobi___ | gage_man Are you talking to me? |
| 01:48.40 | gage_man | no to the people who invented SCCP |
| 01:48.45 | gage_man | no names |
| 01:48.50 | gage_man | pls |
| 01:48.58 | oelewapperke | ie how do I get to include a sip phone in a dial plan ? |
| 01:49.12 | ariel_ | Darwin35 numbers are way to big. Get them to 1/2 size and it might look better. |
| 01:49.15 | hez | ariel_: true.... though eating up 100mb down to the hub would be hard i think... |
| 01:49.15 | essobi___ | gage_man the functionality of SCCP far exceed SIP/H323/MGCP or anything else for that matter.. The sccp drivers for * however sucks. |
| 01:49.21 | gage_man | oh ok |
| 01:49.29 | gage_man | sure |
| 01:49.36 | ariel_ | hez not at first startup of an FTP |
| 01:49.46 | essobi___ | 0_o |
| 01:49.53 | Darwin35 | we are looking for a overall change |
| 01:49.57 | gage_man | I happen to test with all of them and I don't find that so maybe with CCM 3.3 |
| 01:49.59 | hez | ariel_: internal transfer could do it.. but putting a switch down there wouldnt help everything is plugged in to the switch |
| 01:50.00 | gage_man | maybe |
| 01:50.13 | gage_man | but it will stiil use other methods for trunking |
| 01:50.14 | gage_man | pls |
| 01:50.24 | hez | ariel_: out bound traffic is qos, it'll start dumping things pretty quickly if you send at 100mb |
| 01:50.30 | essobi___ | gage_man I run a two 3500 user CCM installs. |
| 01:50.43 | *** join/#asterisk ^quEEn^ (~tR@219.92.185.29) |
| 01:50.44 | Darwin35 | and the 102d will have a headset plugin |
| 01:50.46 | gage_man | great |
| 01:50.52 | gage_man | sorry when you have to convert |
| 01:50.55 | gage_man | night man |
| 01:50.57 | essobi___ | YAM |
| 01:51.04 | ariel_ | hez yes I understand but the moment wait you loose voice and can get echo's. |
| 01:51.36 | ariel_ | Darwin35 10/100 switch would be nice dual ports. |
| 01:51.43 | essobi___ | gage_man Frankly I've wrote an CCM to * conversion framework. |
| 01:51.59 | ariel_ | Darwin35 iax2 as well...???? |
| 01:52.00 | hez | ariel_: possibly, im not worried about it really |
| 01:52.01 | gage_man | great for you what a waste f time |
| 01:52.11 | Darwin35 | it will be like the 102/b and have dual net ports / passthrew |
| 01:52.15 | essobi___ | gage_man You're a cocksmoker. you know that? |
| 01:52.24 | gage_man | hey just honest |
| 01:52.35 | gage_man | sorry freeworld |
| 01:52.37 | hez | do i need to specify the ip for the iaxy, ie it wouldnt take a name? ;( |
| 01:52.43 | Darwin35 | iax2 is being discused but they are not sure |
| 01:52.54 | *** join/#asterisk Simon1 (~Simon@dialin113.eagle.ca) |
| 01:52.55 | Darwin35 | they have to review it more |
| 01:53.00 | gage_man | sorry you don't enjoy opinions but truth does f hurt |
| 01:53.09 | ariel_ | Review it... It's needed like yesterday... |
| 01:53.22 | hez | Darwin35: iax2! |
| 01:53.25 | Darwin35 | gage what are you bitchingabout |
| 01:53.29 | essobi___ | Oh, I enjoy shooting down pompous convoluted opinions. |
| 01:53.32 | gage_man | nothing |
| 01:53.35 | gage_man | having a good time |
| 01:53.46 | gage_man | freaking trying to get this X100p working |
| 01:53.57 | *** join/#asterisk TeaDoR (~MeTaLLiCa@219.92.185.33) |
| 01:54.00 | essobi___ | Hahaha. Now wonder you're so pissy. |
| 01:54.12 | Darwin35 | hez they are thinking and workingon the idea but not sure about iax2 |
| 01:54.12 | essobi___ | s/Now/No |
| 01:54.32 | gage_man | essobi have a great night sorry man |
| 01:54.34 | hez | Darwin35: I think its a great idea! sip is broken ;) |
| 01:54.40 | Darwin35 | but they had a hard time with ILBC and its still needs work to fix first |
| 01:54.46 | gage_man | just had bad experiences with gx and CCM |
| 01:54.51 | gage_man | not a big fan |
| 01:54.54 | gage_man | sorry |
| 01:55.11 | hez | Darwin35: embed an iaxy in it ;) |
| 01:55.18 | ariel_ | skip ILBC use GSM and IAX better. |
| 01:55.26 | Darwin35 | but if you find pics of pjones that would not cost a arm and leg to manufacutre point me to photos |
| 01:55.42 | Darwin35 | that would drive the cost up |
| 01:55.57 | gage_man | *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Goto("Zap/1-1", "home|1") in new stack -- Goto (default,home,1) -- Executing NoOp("Zap/1-1", "") in new stack -- Executing Macro("Zap/1-1", "record-on|XXXXXXXXXXX|9787711755") in new stack -- Executing AGI("Zap/1-1", "set-timestamp.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi Failed to execute '/var/lib/asterisk |
| 01:56.07 | Darwin35 | and they want to keep it under 80 bucks |
| 01:56.10 | *** join/#asterisk Mer|_in (~Merlin@c211-30-34-107.mirnd1.nsw.optusnet.com.au) |
| 01:56.11 | hez | Darwin35: yeah but it would do iax =) |
| 01:56.15 | ariel_ | So what there are many people waiting with there month open for a hard iax phone. |
| 01:57.10 | ariel_ | Darwin35 you just say IAX in a hard phone and people start to well you know... |
| 01:57.17 | Darwin35 | if you all want iax then write a letter to support@grandstream.com and say you would like to see the iax2 protocal added to the 101/102 |
| 01:57.20 | *** join/#asterisk Bentley (~rcourtna@dsl-mtryl-209-5-169-44-cgy.nucleus.com) |
| 01:57.21 | *** join/#asterisk PoLiCe33 (~TehlikeLi@pcd180026.netvigator.com) |
| 01:57.34 | hez | iax! |
| 01:57.50 | Darwin35 | send emails it will push them to do it |
| 01:57.57 | ariel_ | you see humps already... |
| 01:58.00 | Darwin35 | as it stands now its in debate |
| 01:58.07 | Bentley | hez - yes, Telus has disconnect supervision - just confirmed |
| 01:58.12 | Darwin35 | and on the middle back shelf of ideas |
| 01:58.25 | hez | Bentley: mmmm so why does it take so long to hangup ;( |
| 01:58.43 | ariel_ | Darwin35 a real flash button would be nice too. |
| 01:58.59 | Darwin35 | cool the wood for my new computer case will be here in the am |
| 01:59.02 | nhuisman | Anyone ever had problems with cisco phones rebooting while in the asterisk voicemail menus? |
| 01:59.03 | ariel_ | Darwin35 or even a backup PSTN port... |
| 01:59.10 | Darwin35 | the flash works fine |
| 01:59.33 | Bentley | hez - not sure, i think it must just be a limitation of the digium card or somethign |
| 01:59.33 | nhuisman | I have cisco 7960 phones using SIP 6.3, when in the voicemail menus after a dozen button presses, phone reboots. I have had these problems on asterisk 1.0 stable, and i am currently on latest dev sources. |
| 02:00.05 | Darwin35 | no that wont happen its a net phone not a pstn phone |
| 02:00.06 | hez | Bentley: could be ;( (it would be nice if the damn voicemail app would detect the short msg and silence and just drop it) |
| 02:00.34 | Darwin35 | they would have to much coding for it to know if pstn vs net |
| 02:00.38 | *** join/#asterisk Ze{R}Da (~Poor@203.131.155.173) |
| 02:00.40 | Bentley | i think CVS-HEAD has that option in voicemail, no? |
| 02:00.44 | Darwin35 | and that would have to be hard coded |
| 02:00.47 | Bentley | ie: delete empty voicemails |
| 02:01.11 | *** join/#asterisk N-A-T-A-S-C-H-A (~CiciKizz@219.93.58.22) |
| 02:01.27 | hez | Bentley: possibly... im only running 1.0 |
| 02:01.41 | ariel_ | Darwin35 you were asking for suggestions so I suggested.. I understand.. But do something about those stupid colors. |
| 02:02.00 | *** join/#asterisk menger (~menger@dsl-40.66.240.220.rns01-dryb-mel.dsl.comindico.com.au) |
| 02:02.03 | Darwin35 | thats why we are asking for case ideas |
| 02:02.07 | gage_man | -- Starting simple switch on 'Zap/1-1' |
| 02:02.07 | gage_man | <PROTECTED> |
| 02:02.07 | gage_man | <PROTECTED> |
| 02:02.08 | gage_man | Jun 9 22:01:34 WARNING[1192437440]: pbx.c:1814 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 'home' in context 'default', but no invalid handler |
| 02:02.08 | gage_man | <PROTECTED> |
| 02:02.12 | gage_man | anyone seen this error |
| 02:02.20 | Darwin35 | point to a nice clean case |
| 02:02.33 | gage_man | trying to install X100p |
| 02:03.01 | hez | kram: thanks for the help eh? ;) (got busy playing and forgot to thank you) |
| 02:03.05 | andrewg | 9/win 32 |
| 02:03.08 | andrewg | -9 |
| 02:03.14 | menger | gage_man: seems okay to me, extension home does not exist in [default] in extensions.conf |
| 02:03.21 | ariel_ | gage_man home? |
| 02:03.31 | menger | gage_man: and you don't have a catch extension to goto for invalid extensions |
| 02:03.58 | gage_man | can you help me with some syntax |
| 02:04.09 | gage_man | I got this working in about 30 min |
| 02:04.18 | gage_man | just need to use this for a test bed |
| 02:04.22 | gage_man | for coding |
| 02:04.28 | ariel_ | gage_man use pastebin to put you zapata.conf section and extensions.conf for us to see. |
| 02:04.39 | gage_man | got X100p trying to dial in and get an operator |
| 02:05.01 | ariel_ | gage_man use the examples at http://www.fnords.org/~eric/asterisk/ |
| 02:05.47 | Umaro | Guys, i have a call queue setup, and the strategy is set to random, but whenever a call comes in, it goes to the first agent specified in the agents.conf for that group |
| 02:05.53 | Umaro | I've got "member => Agent/:1" in my queues.conf for the queue, but it doesn't send calls to any other agents |
| 02:06.13 | Umaro | if that first agent is busy, it just calls them again. |
| 02:06.53 | Umaro | also, how come none of the queue options work? |
| 02:07.10 | Darwin35 | hmm time to case hunt |
| 02:07.12 | menger | Umaro: try roundrobin |
| 02:07.51 | Umaro | menger, roundrobin doesn't seem to work right either |
| 02:08.23 | Umaro | menger, When I set it to roundrobin, it rings one person, if they are busy, it waits for retry time, then rings that same agent again |
| 02:09.14 | *** join/#asterisk Feri[D]e (~ReNGiN_@218.111.176.37) |
| 02:09.20 | menger | Umaro: i have not tried using agents, but I use member => SIP/1101,1 and so on and it worked in roundrobin |
| 02:09.34 | menger | Umaro: and dynamic agents worked |
| 02:09.43 | menger | Umaro: do you have them all set to the same penalty? |
| 02:10.01 | Umaro | menger, yep |
| 02:10.06 | gage_man | can I not have a SIP phone installed to get this X100p just to answer phones |
| 02:10.16 | menger | Umaro: good, cause i could not get different penalties to work at all |
| 02:10.21 | gage_man | I jsut really want auto attendant |
| 02:10.21 | Umaro | menger, hehe |
| 02:10.28 | gage_man | I have everything else |
| 02:10.38 | Umaro | menger, Does app_queue take options work for you? |
| 02:10.58 | menger | gage_man: you don't need any phones to have your x100p answer phones |
| 02:10.58 | Umaro | -work |
| 02:11.12 | menger | Umaro: take option? |
| 02:11.17 | gage_man | CLI> -- Starting simple switch on 'Zap/1-1' |
| 02:11.17 | gage_man | <PROTECTED> |
| 02:11.17 | gage_man | <PROTECTED> |
| 02:11.18 | gage_man | Jun 9 22:11:05 WARNING[1192437440]: pbx.c:1814 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 'home' in context 'default', but no invalid handler |
| 02:11.18 | gage_man | <PROTECTED> |
| 02:11.19 | gage_man | <PROTECTED> |
| 02:11.21 | gage_man | stilll getting this |
| 02:11.23 | gage_man | man |
| 02:11.33 | menger | gage_man: look in your extensions.conf |
| 02:11.40 | menger | gage_man: your [default] section |
| 02:11.40 | gage_man | yup |
| 02:11.47 | menger | do you have an extension home? |
| 02:11.47 | gage_man | there |
| 02:11.57 | *** join/#asterisk su-|lappy (su-|lappy@pcp03415506pcs.kilng01.ct.comcast.net) |
| 02:12.13 | *** join/#asterisk LoneWolf071 (~yousuck@c-24-1-86-83.client.comcast.net) |
| 02:12.28 | *** join/#asterisk CeyDa^^ (~lesbian20@219.95.120.78) |
| 02:13.00 | gage_man | exten => s,1,Goto(home,1) |
| 02:13.04 | LoneWolf071 | i have nortan symantics as my firewall, but it takes up so much system resources, can anyone recommend a firewall that doesn't take up a lot of system resources |
| 02:13.40 | gage_man | exten => s,1,Goto(home,1) |
| 02:13.40 | gage_man | exten => home,1,NoOp |
| 02:13.40 | gage_man | exten => home,2,Macro(record-on|${MYHOMEPHONE}|${CALLERIDNUM}) |
| 02:13.40 | gage_man | exten => home,3,Dial(SIP/11&SIP/12&SIP/13&SIP/14|30|tr) |
| 02:13.41 | file[ashan] | a router with NAT provides a nice natural firewall |
| 02:13.41 | gage_man | exten => home,104,Goto(home|6) |
| 02:13.41 | gage_man | exten => home,4,Voicemail2(u10) |
| 02:13.42 | gage_man | exten => home,5,Wait(2) |
| 02:13.44 | gage_man | exten => home,6,Playback(vm-goodbye) |
| 02:13.46 | gage_man | exten => home,7,Hangup |
| 02:13.48 | gage_man | exten => 999,1,Answer |
| 02:13.50 | gage_man | exten => 999,2,MusicOnHold(default) |
| 02:13.55 | gage_man | SORRY FOTR THE PASTE |
| 02:14.06 | LoneWolf071 | don't need a routre |
| 02:14.07 | LoneWolf071 | er |
| 02:14.14 | LoneWolf071 | this is my family computer |
| 02:14.23 | LoneWolf071 | want firewall becuase of hackers |
| 02:14.30 | Umaro | menger, yeah like timout, or ring instead of moh.. |
| 02:14.34 | Umaro | er timeout |
| 02:14.41 | menger | Umaro: I have timeout working |
| 02:14.46 | Umaro | menger, stable? |
| 02:14.52 | Umaro | I mean, using asterisk stable? |
| 02:15.10 | menger | Umaro: yes, but i had to back port the app_queue patches to get it to work |
| 02:15.22 | menger | Umaro: app_queue has heaps of better stuff in current |
| 02:15.32 | file[ashan] | a router is handy though |
| 02:15.34 | Umaro | menger, can you send me that backport? |
| 02:15.38 | file[ashan] | I've got two. |
| 02:16.04 | menger | Umaro: yep, email? |
| 02:16.28 | *** part/#asterisk LoneWolf071 (~yousuck@c-24-1-86-83.client.comcast.net) |
| 02:16.29 | Umaro | menger, umaro@comcastNOSPAMPLS.net |
| 02:16.37 | *** join/#asterisk juice (1000@user148.net283.mo.sprint-hsd.net) |
| 02:16.44 | ariel_ | gage_man why are you using home instead of s |
| 02:17.45 | ariel_ | gage_man home should be [home] a context |
| 02:18.03 | Umaro | menger, is this a patch, or just a app_queue.c ? |
| 02:19.21 | menger | Umaro: patch |
| 02:19.24 | menger | Umaro: emailed |
| 02:19.29 | Umaro | menger, thanks |
| 02:20.46 | Darwin35 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=41374&item=4193527923&rd=1 |
| 02:20.50 | Darwin35 | Ilike this one |
| 02:21.06 | Darwin35 | it would be a great sip phone |
| 02:21.14 | *** join/#asterisk KoMite (~Sevecen@cras77p34.navix.net) |
| 02:21.24 | *** join/#asterisk su-|lappy (su-|lappy@pcp03415506pcs.kilng01.ct.comcast.net) |
| 02:21.47 | *** join/#asterisk Dj_Fanks (~Melteem@219.95.99.22) |
| 02:22.03 | ariel_ | Darwin35 yes it would but I would like real button better. |
| 02:22.14 | menger | bbl |
| 02:22.16 | Darwin35 | that can be done |
| 02:22.36 | *** join/#asterisk macmx (~mike@201.129.119.8) |
| 02:25.29 | ariel_ | something is not right here! |
| 02:26.53 | Darwin35 | I cant grab the pic |
| 02:26.56 | Darwin35 | this sucks |
| 02:27.02 | *** join/#asterisk gaRip (~BlueMaRY@219.94.64.38) |
| 02:29.07 | Darwin35 | ok got the image |
| 02:30.20 | *** join/#asterisk KoCaeLi (~MeSSaGe@219.95.25.159) |
| 02:32.24 | ariel_ | OK what is NFAS and GR-303? |
| 02:33.25 | file[ashan] | if I had a dollar for everytime that was asked... oh wait, that's a great idea |
| 02:34.23 | ariel_ | dollar that all you want a dollar... Hell I can do that. Oh wait I am broke till mid july...Argh can't do that! |
| 02:34.38 | *** join/#asterisk webmind (~cme2@217-195-236-172.dsl.esined.net) |
| 02:34.46 | file[ashan] | ariel_: rain check! |
| 02:35.01 | Umaro | menger, ok, timeout works.. but the ring instead of moh doesn't.. do you use that? |
| 02:35.40 | SplasPood | bkw_: around? |
| 02:35.42 | ariel_ | ok so let's have it... or get them to change the top posting to what it is... |
| 02:36.03 | nhuisman | I have cisco 7960 phones using SIP 6.3, when in the voicemail menus after a dozen button presses, phone reboots. I have had these problems on asterisk 1.0 stable, and i am currently on latest dev sources. |
| 02:36.07 | nhuisman | Anyone ever had problems with cisco phones rebooting while in the asterisk voicemail menus? |
| 02:36.17 | file[ashan] | http://www.telcordia.com/services/testing/integrated-access/gr/gr303/ |
| 02:37.02 | file[ashan] | NFAS = Non-Facility Associated Signaling |
| 02:37.11 | *** join/#asterisk E|ny (~voip@205.200.64.172) |
| 02:37.18 | file[ashan] | allows a single D channel to control multiple PRIs |
| 02:37.35 | file[ashan] | so let's say you've got 2 PRIs |
| 02:37.46 | ariel_ | oh that. That is still a dream.. |
| 02:37.47 | file[ashan] | 1 D channel on the first, and 23 voice - then on the second you could have 24 voice |
| 02:37.51 | gage_man | anyone give me a quik hand |
| 02:37.55 | file[ashan] | mmmkthx |
| 02:38.02 | gage_man | extensions and X100p |
| 02:38.11 | ariel_ | gage_man yes did you read what I posted? |
| 02:38.22 | gage_man | no send again pls |
| 02:38.57 | ariel_ | home needs to be it's context [home] then start each one with an s not home |
| 02:39.32 | ariel_ | also you need to include=home under default |
| 02:40.00 | *** join/#asterisk DonaTeLLo (~SranLi@218.208.243.85) |
| 02:40.04 | ariel_ | also you need to include=home under default or not! forgot about this one.. |
| 02:40.33 | file[ashan] | feeling blue I'm thinking of you |
| 02:40.37 | file[ashan] | oh I was a fool to let you walk away |
| 02:40.52 | gage_man | never had to do that before but will try |
| 02:41.52 | file[ashan] | I should try to find my keys... |
| 02:42.03 | ariel_ | exten => home is not correct. exten => s,1 |
| 02:42.31 | ariel_ | Hell I should go home been here since 8:30 am it's now 10:42 pm.. Long day now night. |
| 02:43.04 | file[ashan] | ha found them |
| 02:43.08 | file[ashan] | they were sitting under a floppy drive |
| 02:43.27 | ariel_ | floppy drive you still have one of those... |
| 02:43.43 | file[ashan] | in my computers no, but they're piled up |
| 02:43.57 | gage_man | Jun 9 22:43:32 WARNING[1200825920]: pbx.c:1814 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension '10' in context 'default', but no invalid handler |
| 02:43.59 | gage_man | got that |
| 02:44.29 | *** join/#asterisk Saneeem^ (~muGiSh__@219.95.193.225) |
| 02:45.37 | ariel_ | gage_man can you paste your settings into www.pastebin.com so I can see it please |
| 02:45.56 | gage_man | sure' |
| 02:45.58 | gage_man | thanks |
| 02:46.03 | gage_man | by the way |
| 02:46.24 | Darwin35 | ok I found a few images |
| 02:46.30 | Darwin35 | I like alot |
| 02:47.13 | *** join/#asterisk inSerT (~DeepBlue@219.95.233.47) |
| 02:47.53 | gage_man | did you see them |
| 02:52.18 | *** join/#asterisk webmind (~cme2@217-195-236-172.dsl.esined.net) |
| 02:54.06 | *** join/#asterisk brc_ (~Brian@ip68-108-199-177.ph.ph.cox.net) |
| 02:54.10 | file[ashan] | lies! all lies! |
| 02:55.01 | brc_ | file |
| 02:55.11 | brc_ | was it you who was porting asterisk to win? |
| 02:55.18 | wasim | eww |
| 02:55.34 | brc_ | I know |
| 02:55.34 | file[ashan] | brc_: maybe |
| 02:55.36 | file[ashan] | why? |
| 02:55.43 | file[ashan] | it was a "proof of concept/can be done" |
| 02:55.44 | brc_ | couldn't remember who it was |
| 02:55.48 | brc_ | yah I know |
| 02:55.48 | Alric | file[ashan], Think very carefully about your answer :) |
| 02:55.58 | brc_ | wasim why are ya around always now? |
| 02:56.04 | brc_ | you aren't slacking off on farfon are you? |
| 02:56.08 | brc_ | ~thwack wasim |
| 02:56.10 | jbot | ACTION hits wasim with a Holy Bible in the knee |
| 02:56.59 | *** join/#asterisk ^^TuRk4eVa^^girl (WATCH@AC9B6A42.ipt.aol.com) |
| 02:57.30 | wasim | no, my sleep cycle is all busted now, i'm awake at ungodly hours like 0800 |
| 02:57.44 | brc_ | why |
| 02:58.38 | Moc__ | hi all |
| 02:58.44 | hez | ariel_: thank you for the help earlier =) I found a ugly solution... ;) ( iaxy -> old 10 mb hub -> switch (the one in my room that it didnt like) -> switch -> hub -> dhcp server |
| 02:58.44 | Moc__ | hi wasim, how things going ? |
| 02:59.16 | file | hubs are so... icky |
| 02:59.40 | hez | file: back in my day they were all we had |
| 02:59.58 | file | that was then, this is now |
| 03:00.14 | hez | yeah, and? |
| 03:00.24 | file | we have switches and routers! |
| 03:00.24 | *** join/#asterisk doLL-15f (~tR@207.226.41.185) |
| 03:00.34 | hez | file: the iaxy wont talk to that switch ;( |
| 03:02.05 | file | silly foreign languages |
| 03:02.10 | YoYo | man, what's with all the peeps having problems with the iaxy? |
| 03:02.25 | YoYo | does the thing have a retarded nic? |
| 03:02.41 | file | probably *cough* cheap |
| 03:02.46 | Darwin35 | night all |
| 03:03.00 | hez | Darwin35: happy dreams of the iax protocol ;) |
| 03:03.12 | file | the IAX protocol is nice, I poked through the libiax2 source |
| 03:03.45 | *** part/#asterisk Simon1 (~Simon@dialin113.eagle.ca) |
| 03:03.49 | *** part/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 03:05.04 | *** join/#asterisk DenyMe (Mr_Pride@202.155.128.222) |
| 03:05.22 | *** join/#asterisk ^sinem_ (iNDigence@219.95.15.48) |
| 03:05.24 | Moc__ | file, make a RFC ;) |
| 03:05.25 | DenyMe | good morning |
| 03:05.48 | Moc__ | good nice DenyMe ;) |
| 03:05.53 | Moc__ | nice = nite |
| 03:06.18 | DenyMe | need advice on adtran ta 750 |
| 03:06.22 | *** join/#asterisk cowox_cool (sayang@cm182.omega80.maxonline.com.sg) |
| 03:06.25 | DenyMe | :0 |
| 03:06.29 | Moc__ | dont have t1 stuff yet |
| 03:06.54 | gage_man | anyone know what this bug is |
| 03:06.57 | gage_man | <PROTECTED> |
| 03:06.57 | gage_man | <PROTECTED> |
| 03:06.57 | gage_man | <PROTECTED> |
| 03:06.58 | gage_man | <PROTECTED> |
| 03:06.58 | gage_man | Jun 9 23:06:22 WARNING[1200825920]: pbx.c:1814 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension '10' in context 'default', but no invalid handler |
| 03:07.05 | gage_man | or operator error |
| 03:07.13 | *** join/#asterisk MustDie (~voip@ool-18b91fa7.dyn.optonline.net) |
| 03:07.20 | gage_man | flat spin |
| 03:07.22 | gage_man | here |
| 03:07.24 | gage_man | anyone |
| 03:07.38 | *** join/#asterisk MeRwEciq`misil`m (~AnTiSpam@adsl-66-136-62-5.dsl.snantx.swbell.net) |
| 03:07.38 | YoYo | gage: it's pretty clear |
| 03:07.47 | gage_man | YoYo help |
| 03:07.48 | YoYo | gage, you don't have an extension 10 in the [default] context |
| 03:07.56 | gage_man | since I just started this an hour ago |
| 03:08.27 | cowox_cool | hai all, i try to connect x-lite to my asterisk box, but the box said registration failed :< |
| 03:08.41 | YoYo | cowox: www.cisco.com |
| 03:08.45 | YoYo | buy a real SIP phone |
| 03:08.51 | YoYo | =D |
| 03:08.58 | cowox_cool | cannot use x-lite ya |
| 03:09.12 | YoYo | x-lite works... but it's configuration is funky |
| 03:09.19 | YoYo | keep playing with it |
| 03:09.37 | cowox_cool | which configuration ? |
| 03:09.48 | cowox_cool | using windows messanger aso result the same |
| 03:09.53 | cowox_cool | :'( |
| 03:10.40 | *** join/#asterisk AsteriskAR (~mm@lineaAF191.velocom.com.ar) |
| 03:11.25 | cowox_cool | where is the forum for asterisk user ? ^____^ gotta read sum more |
| 03:11.40 | *** join/#asterisk juice (1000@user148.net283.mo.sprint-hsd.net) |
| 03:12.16 | xyzx | http://www.asterisk.org/index.php?menu=support |
| 03:12.49 | hez | cowox_cool: try firefly, its better then x-lite |
| 03:14.54 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 03:15.00 | zamsler | <~~ wishes firefly would work in linux... |
| 03:15.00 | cowox_cool | ok bos !! |
| 03:15.15 | znoG | anyone know how to do a DBPut from the asterisk console? |
| 03:15.21 | cowox_cool | where i can set user for SIP in asterisk ? ^____^ |
| 03:18.37 | *** part/#asterisk gage_man (~root@h00095b093ebb.ne.client2.attbi.com) |
| 03:18.47 | hez | zamsler: i know what you mean! |
| 03:19.06 | hez | zamsler: Adam is looking for someone to port the ui, the rest already compiles =) |
| 03:21.38 | *** join/#asterisk _Phantom (~neil@gateway-nelson.thepacific.net) |
| 03:21.43 | _Phantom | anyone tried EFTPOS via a asterisk server? |
| 03:21.44 | DenyMe | Yoyo : is adtran ta 750 support for 24 fxo? |
| 03:25.12 | zamsler | awesome |
| 03:26.01 | Ro0tSiEgE | cowox_cool: the user is the exten |
| 03:26.02 | *** join/#asterisk draconius (~j00@12-218-60-36.client.mchsi.com) |
| 03:26.09 | Ro0tSiEgE | take a look at sip.conf |
| 03:26.16 | ariel_ | gage_man I am back... for a few |
| 03:28.13 | essobi___ | Mmmm. |
| 03:31.34 | ariel_ | ok so I am sitting here looking at extensions.conf logic and find that it's really fucked. you can do what ever you want with it. go here go there and there is no real logic unless you build it into to you rules. |
| 03:32.35 | ariel_ | I have a customer that we took over there system and they really fucked it up.. I am having to rewrite the complete extensions.conf from scratch... Argh.. |
| 03:33.08 | *** join/#asterisk RoyK (~roy@153.80-202-161.nextgentel.com) |
| 03:34.20 | znoG | how annoying |
| 03:34.27 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 03:34.37 | hez | ariel_: its not possible to send alternate rings to the iaxy eh? ;( you know like on zap/1/555r2 |
| 03:35.10 | ariel_ | hez I have not done that. I think alternate is mainly for zap ports. |
| 03:35.22 | essobi___ | Mmmm. |
| 03:35.34 | hez | damn, it would be kinda nice =) |
| 03:35.55 | ariel_ | ok start page one. All macro's are in there own file.. use include... |
| 03:36.26 | *** join/#asterisk tiziano_18f (~java@62.162.235.73) |
| 03:36.32 | ariel_ | hez have you tried it? |
| 03:36.47 | file | gooooooooooooooodnight |
| 03:36.50 | *** join/#asterisk r0d3nt-600m (~RatMan@wsip-24-234-241-186.lv.lv.cox.net) |
| 03:37.12 | hez | ariel_: thought about it ... couldnt figure how it would look though.... IAX2/hezsiaxyr2 ? ;( dont think so |
| 03:37.14 | ariel_ | good night... wow it's already 11:37pm ... |
| 03:37.15 | hez | night file |
| 03:37.19 | iMediax | whats does "chan_iax2.c:5689 set_config: Ignoring port for now" mean? the port is set and iax->iax is working.. |
| 03:38.05 | ariel_ | iax2 only uses 4569 if you set it up to 5036 it will not use it so back to 4569 |
| 03:38.29 | ariel_ | iMediax it's hard coded.. |
| 03:38.32 | iMediax | ok |
| 03:38.36 | *** join/#asterisk XARiUS (~xarius@wsip-68-224-170-22.sd.sd.cox.net) |
| 03:38.38 | iMediax | so just ignore it |
| 03:38.58 | ariel_ | or change the iax.conf correctly or extensions.conf |
| 03:39.13 | XARiUS | ariel: have any PRI experience? |
| 03:39.14 | iMediax | ok ill try that... thanks |
| 03:39.33 | ariel_ | XARiUS yes but do I have to really say yes... |
| 03:39.39 | XARiUS | lol.. |
| 03:39.57 | XARiUS | Launched * today, everything's awesome, except this one thing I can't seem to figure out |
| 03:40.02 | XARiUS | we get no ringing on outbound pri calls! |
| 03:40.04 | XARiUS | (local ringing) |
| 03:40.07 | ariel_ | my B-channels just reset... every hour on it does that.. |
| 03:41.02 | XARiUS | you'll hear a half a ring, then nothing, just silence until someone answers |
| 03:41.23 | XARiUS | I did a search, but all I see is no ringing on inbound (which is the 'r' option) |
| 03:41.30 | ariel_ | XARiUS I think you have a problem with your carrier. I have not had to do any changes to mine.. I do not use the r in the rules. |
| 03:41.43 | XARiUS | well, it works on the other pbx we have lol |
| 03:41.48 | XARiUS | the one that * replaced today |
| 03:42.37 | ariel_ | XARiUS then for now use the r for outbound calls it's a hack but it some time is needed. Is the PRI set to national (NI2) by the carrier |
| 03:42.48 | *** join/#asterisk cilek-- (DeRYa___@203.106.169.223) |
| 03:42.56 | XARiUS | I did.. still no ringing |
| 03:43.00 | ariel_ | XARiUS are you doing your timing off the PRI? |
| 03:43.15 | XARiUS | yes, it's provided by the pri |
| 03:44.15 | ariel_ | whats your settings for it in zaptel.conf |
| 03:45.02 | ariel_ | span=1,0,0,esf,b8zs is mine... |
| 03:45.40 | XARiUS | yeah it's very basic |
| 03:45.43 | XARiUS | same settings |
| 03:46.12 | XARiUS | span=1,1,0,esf,b8zs |
| 03:46.12 | XARiUS | bchan=1-23 |
| 03:46.12 | XARiUS | dchan=24 |
| 03:46.12 | XARiUS | defaultzone=us |
| 03:46.12 | XARiUS | loadzone=us |
| 03:46.15 | *** join/#asterisk yahozna (rmger@nrez.net) |
| 03:46.45 | ariel_ | yes you got it. I don't use timing on my pri I use it off the T1 |
| 03:46.57 | XARiUS | I guess I could try that |
| 03:47.35 | ariel_ | XARiUS I have 2 pri and 1 T1 inbound/outbound.. |
| 03:47.58 | *** join/#asterisk BreaKDance1 (RIZA@210.186.115.47) |
| 03:48.04 | *** join/#asterisk juice (~juice@user148.net283.mo.sprint-hsd.net) |
| 03:48.41 | XARiUS | yah |
| 03:48.47 | XARiUS | this should be straightforward.. |
| 03:50.18 | ariel_ | Wow I just killed my customers system... HEhehehe. I knew I was going to do that.... Now hope it comes back up..... |
| 03:51.22 | XARiUS | it's gotta be pri related because internal calls have ringing |
| 03:54.19 | ariel_ | has anyone tried faxing via IAX2 and ulaw... I am getting some stranges things via IAX2 trunking.... |
| 03:55.10 | *** join/#asterisk jonk (~jon@203-219-213-186-qld.tpgi.com.au) |
| 03:55.31 | XARiUS | here's a weird one for ya ariel |
| 03:55.32 | _Phantom | anyone tried EFTPOS via a asterisk server? |
| 03:55.35 | *** join/#asterisk r0d3nt_600m (~RatMan@wsip-24-234-241-186.lv.lv.cox.net) |
| 03:55.41 | XARiUS | if I remove "r" from the dialstring I get no ringing at all |
| 03:55.53 | XARiUS | whereas if it's there I get like, half a ring to a full ring |
| 03:56.31 | ariel_ | XARiUS sound like asterisk thinks the other party has already pickup the line. |
| 03:57.04 | ariel_ | _Phantom what is EFTPOS? |
| 03:57.22 | _Phantom | Electronic Funds Tranfer at Point of Sale..... |
| 03:58.04 | ariel_ | _Phantom thanks.. I did not know that. |
| 03:58.33 | XARiUS | hm, maybe.. |
| 03:58.53 | _Phantom | basicly a data connection to the bank for payments at POS |
| 03:59.00 | _Phantom | modem sort of thing |
| 03:59.13 | *** join/#asterisk mundane (~mundane@ip68-99-138-114.ph.ph.cox.net) |
| 03:59.43 | andrewg | win 24 |
| 04:00.10 | ariel_ | _Phantom so you need a modem to dial out then. |
| 04:00.21 | *** join/#asterisk menger (~menger@dsl-40.66.240.220.rns01-dryb-mel.dsl.comindico.com.au) |
| 04:01.39 | ariel_ | _Phantom there is some modem.conf but I have never played with it. Also there are some ras support in Asterisk again something I have never played with. |
| 04:02.27 | jon-k | hello everyone |
| 04:02.39 | menger | hello Jon |
| 04:02.40 | jon-k | what values are acceptable for zttest? |
| 04:02.43 | menger | hello jon-k |
| 04:02.49 | jon-k | hello menger :) |
| 04:02.51 | menger | jon-k: no idea |
| 04:02.53 | ariel_ | OK got my customer system back up. I guess I can finish the rest in the morning for them.. Silly customer trying to do it all them selfs... |
| 04:06.14 | ariel_ | I just got an error on diax.. Run-time error '6': Overflow.. Funny I was not using it. It was just there on the desktop.... |
| 04:10.18 | hez | I noticed the iaxy has what looks like a regular fxs module, has someone ripped it apart and replaced it with a fxo module yet? =) |
| 04:10.33 | jon-k | anyone here running a e100p card? |
| 04:10.35 | _Phantom | the modem to dialout isn't the issue, however I'm concerned about compression and that sort of thing disrupting the data connection, how well does asterisk work with services such as faxes etc over its network? |
| 04:10.51 | hez | _Phantom: ok but you have to use u/alaw |
| 04:10.54 | jon-k | if so, can you show me what the output from zttest looks like on your system? |
| 04:11.05 | hez | _Phantom: and they have to stay in the slower speeds |
| 04:11.29 | _Phantom | cool.... I might have a play with some of this then..... |
| 04:11.33 | _Phantom | cheers |
| 04:11.49 | _Phantom | using a SIPPhone Call-in-One unit with my asterisk server, very cool box indeed |
| 04:11.58 | jon-k | kram? bkw? citats? twisted? |
| 04:13.20 | XARiUS | well I give up, I've flipped every switch there is. |
| 04:13.36 | _Phantom | XARiUS, time to start flipping the undocumented ones ;) |
| 04:15.56 | XARiUS | no kidding Phantom |
| 04:16.32 | _Phantom | heh |
| 04:16.54 | *** join/#asterisk CarefuL (~rock_shee@cm223.omega149.maxonline.com.sg) |
| 04:16.58 | XARiUS | well, I still have that free hour of support from digium from when I bought the te410p =) |
| 04:17.10 | XARiUS | might well take advantage of it in the morning hehe |
| 04:20.00 | essobi___ | howdy kram |
| 04:20.12 | kram | howdy |
| 04:20.13 | XARiUS | ooo I bet kram can help! |
| 04:20.22 | XARiUS | might save me that phone call in the morning lol |
| 04:20.35 | XARiUS | actually, damn, don't you ever sleep Kram? =) |
| 04:20.46 | ariel_ | hello kram question on iax2. If we are using it for trunking two server on an internal IP address how do we get the box to use iax2 on the 2nd nic card? |
| 04:20.50 | kram | apparently it's the only alternative to working |
| 04:21.09 | XARiUS | lol you're in the wrong place then.. thanks ariel, just cut line why don't ya. =) |
| 04:21.34 | ariel_ | XARiUS you just type faster then me. |
| 04:21.45 | essobi___ | kram I feel you. |
| 04:22.37 | XARiUS | can't say much myself, I'm not at work, and I'm working... hehe |
| 04:22.51 | *** join/#asterisk ^derya_ (~fLoweR19f@host-29-122-38-212.azdata.net) |
| 04:23.03 | ariel_ | Damm I am at work and I am going home soon.. Got to get some sleep... |
| 04:23.11 | XARiUS | Kram: can you think of some uber secret switch/setting that would give me ringing on outbound pri calls? =) te410p... |
| 04:23.16 | kram | ariel: sounds like a question for paid support |
| 04:23.17 | jon-k | kram eh |
| 04:23.55 | XARiUS | if I use the 'r' setting on the dial string, I get half a ring, maybe a full ring, then nothing until the callee answers.. :( |
| 04:24.30 | ariel_ | kram really now! So my 6 T410p cards did not come with at least one hour of support? |
| 04:24.59 | *** join/#asterisk r0d3nt-600m (~RatMan@wsip-24-234-241-186.lv.lv.cox.net) |
| 04:25.13 | XARiUS | ariel: sure they did, but not at 12:30am lol |
| 04:25.14 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
| 04:25.16 | kram | ariel: don't know, not my question to answer |
| 04:25.23 | XARiUS | I'll prolly call in the morning myself |
| 04:25.24 | kram | areil: my job is writing asterisk |
| 04:25.29 | XARiUS | if I don't get it nailed tonight |
| 04:25.37 | ariel_ | In any case it's something I will have to do in the morning. I am wasted already. |
| 04:26.02 | kram | ariel: you can try tech support, sometimes if you've bought a lot they'll help you with something not related to the hardware itself |
| 04:26.17 | kram | areil: but if they don't, it's because we have to be careful of feature creep on support |
| 04:26.30 | kram | next thing you know people expect you to configure 100 sip phones becuase the bought an x100p |
| 04:26.33 | ariel_ | kram thanks I will call them in the morning.. |
| 04:26.39 | kram | okay, good luck |
| 04:26.44 | XARiUS | lol I hear that |
| 04:26.46 | kram | if they need me, they'll get me |
| 04:26.53 | kram | but in general, my time is better spent programming than providing tech support |
| 04:26.54 | XARiUS | I do support for my company's software products as well. |
| 04:27.05 | XARiUS | people take advantage of every inch they can milk out of you. |
| 04:27.05 | kram | generally i only get involved when there are segs and such |
| 04:27.30 | *** join/#asterisk ComPuTeR (~EsHRa@218.111.2.202) |
| 04:27.32 | jon-k | kram: what does the output of zttest mean? |
| 04:27.36 | jon-k | obviously 100% is good |
| 04:27.38 | ariel_ | kram yes your right. I was just wondering. Due to the ip address only has one place to bind to in iax.conf. |
| 04:27.44 | jon-k | but what about the 99.9%'s? |
| 04:27.47 | jon-k | are they bad, or acceptable? |
| 04:27.58 | jon-k | is there a rough cut off for when it starts to cause errors? |
| 04:29.39 | kram | jonk: the more 9's the better, 99.99 is pretty good |
| 04:29.56 | kram | jonk: it's a measure of the accuracy of the agreement of your zap timer to your computer's timer |
| 04:30.10 | kram | jonk: when it fails in an important way it's usually very very dramatic |
| 04:30.54 | gafachi | kram: so is GR303 public domain? I thought it needed to be licensed from someone? |
| 04:31.11 | ariel_ | you all have a great night. I need to leave for now. I still need to drive home as it is... |
| 04:31.22 | gafachi | I know I always see GR303 T1s priced more expensively than PRI.... |
| 04:31.43 | *** join/#asterisk macmx (~mike@201.129.119.8) |
| 04:34.56 | jon-k | kram: okay, so 99.9% is pretty good then |
| 04:35.16 | jon-k | thanks for the help |
| 04:37.17 | *** join/#asterisk Landrocker (Landrocker@219-88-205-48.adsl.ihug.co.nz) |
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| 04:38.20 | kram | gafachi: not if you write your own stack it isn't |
| 04:38.28 | kram | I just integrated GR-303 into libpri |
| 04:38.34 | kram | jonk: glad to be of service |
| 04:43.03 | RoyK | wtf is GR-303? |
| 04:43.49 | *** join/#asterisk draconius (~j00@12-218-60-36.client.mchsi.com) |
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| 04:49.06 | XARiUS | oh well off to bed.. night all |
| 04:49.34 | *** join/#asterisk FeRit (~java@207-224-17-113.ptld.qwest.net) |
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| 05:09.10 | atacomm | anyone know how to get an employee's actual pay rate by taking the amount they want after taxes? |
| 05:11.34 | menger | atacomm: couldn't you just multiply by 1.<tax rate here>? |
| 05:12.05 | atacomm | menger: one has to figure out the correct federal rate, fica rate, and state rate.... i was trying to find an online calculator but am unsuccesful.... |
| 05:12.17 | Umaro | What on earth does GR-303 have to do with *? |
| 05:12.27 | menger | atacomm: move to australia, is much simpler lol |
| 05:12.32 | atacomm | lol |
| 05:13.42 | Umaro | atacomm, I don't know, how do you get an employee's actual pay, when you are taking the gross amount of what they get paid? |
| 05:13.50 | *** join/#asterisk Jelibon (~eJder@host-116-115-38-212.azdata.net) |
| 05:13.53 | twisted | atacomm: yeah, TAKE what they want afte taxes, then let them come ask you for that amount back :P |
| 05:14.08 | atacomm | lol |
| 05:15.14 | *** join/#asterisk juice (1000@user148.net283.mo.sprint-hsd.net) |
| 05:15.21 | twisted | atacomm: or just simply ask them ;) |
| 05:15.30 | *** join/#asterisk r0d3nt-lv (RatMan@wsip-24-234-241-184.lv.lv.cox.net) |
| 05:15.56 | atacomm | lol, no, see i've been told how much i need to put in someone's pocket every month..... so now i have to figure out what the real pay rate is before taxes...... |
| 05:16.03 | twisted | ahhhhhh |
| 05:16.10 | twisted | do you know the tax rate there? |
| 05:16.23 | atacomm | nope, heh...... i wish i could just find an online calc |
| 05:16.31 | twisted | wait a minute |
| 05:16.50 | twisted | atacomm: don't you run a business? |
| 05:16.53 | *** join/#asterisk FaNtazi (~TeXT@212.175.61.180) |
| 05:17.18 | atacomm | yeah.....lol, but payroll is outsourced and i need to know what to tell paychex his pay rate is |
| 05:17.27 | twisted | call paychex, and ask them |
| 05:17.38 | twisted | they should be happy to help you |
| 05:19.20 | MustDie | atacomm: yo, it's based on your employee earnings. Ie if i just made 150K in 11 month |
| 05:19.37 | MustDie | and you paying me $5 per hour |
| 05:19.46 | MustDie | my pay rate would be negative after all :P |
| 05:20.14 | MustDie | atacomm: are you there ? |
| 05:21.10 | *** join/#asterisk xxyxxz (~nnnnn@c-24-10-198-127.client.comcast.net) |
| 05:25.38 | *** part/#asterisk tooley (~tooley@ip68-2-55-132.ph.ph.cox.net) |
| 05:27.43 | *** join/#asterisk FuzzyCat (~ScaredyCa@e67171.upc-e.chello.nl) |
| 05:28.19 | atacomm | MustDie: sorry, having an algebra discussion on this with twisted |
| 05:29.57 | bkw_ | lalala |
| 05:30.07 | bkw_ | whats up? |
| 05:30.12 | atacomm | bkw: what do you think of airtunes? |
| 05:30.25 | bkw_ | dont know |
| 05:30.51 | bkw_ | apple needs to re release the newton |
| 05:30.54 | atacomm | lol, yeah |
| 05:31.02 | atacomm | i was pissed with their crappy g5 upgrade |
| 05:31.05 | *** join/#asterisk KaranfiL^^ (ECE_19F_@217.14.100.140) |
| 05:31.19 | bkw_ | now thatwas a pda' |
| 05:32.00 | bkw_ | any way im ircing from my phonw:)' |
| 05:32.08 | atacomm | ah |
| 05:32.33 | _Phantom | heh |
| 05:32.38 | bkw_ | har hqar |
| 05:33.47 | bkw_ | bed time |
| 05:34.00 | Tekati | What is the bug number of the sql config stuff? |
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| 05:38.04 | jon-k | bkw/kram: what debugging information would you guys need from me to try and determine the cause of a crosstalk problem on an e1 line? |
| 05:41.51 | jon-k | bueller? bueller? bueller? |
| 05:41.54 | MustDie | jon: what are you talking about ? |
| 05:42.12 | MustDie | crosstalk means, that on 1 channel, you can hear another ? |
| 05:42.16 | jon-k | yeah |
| 05:42.17 | jon-k | basically |
| 05:42.21 | jon-k | call in progress |
| 05:42.25 | jon-k | someone else dialls up |
| 05:42.39 | jon-k | they can sometimes hear the call in progress on the other line |
| 05:42.42 | jon-k | in between the ring cadences |
| 05:42.55 | jon-k | that is, until the line is answered |
| 05:42.59 | jon-k | and then its all hunky dory again. |
| 05:43.08 | jon-k | problem's been going on for quite a while now |
| 05:43.21 | jon-k | i'm starting to wonder if it has something to do with zero-ing buffers or something |
| 05:43.47 | jon-k | though i'm just wondering if someone has an idea of what i should be looking for |
| 05:43.54 | jon-k | or perhaps if there's something obvious i'm doing wrong |
| 05:43.57 | MustDie | i have aggressvie suppressor enabled, so i can hear huge echo until line is answered |
| 05:44.08 | jon-k | mmm echo |
| 05:44.10 | jon-k | that's another problem i get |
| 05:44.12 | jon-k | though not so bad |
| 05:44.18 | MustDie | jon: try to 1,Answer |
| 05:44.25 | MustDie | then 2, Dial(123213,r |
| 05:44.29 | jon-k | mmm |
| 05:44.31 | jon-k | i'll check that out |
| 05:44.34 | MustDie | try to answer line first |
| 05:44.42 | MustDie | right away |
| 05:44.58 | jon-k | the situation is a little more confused since i'm using iax2 to bridge an e1 line over a network :) |
| 05:45.08 | jon-k | but i'll check to make sure i'm answering everything first |
| 05:45.14 | MustDie | on E1 side |
| 05:45.32 | MustDie | just fix your dial plan, to Answer, right before dialing string |
| 05:45.44 | MustDie | jon: best - is to file a bug report |
| 05:46.09 | MustDie | http://bugs.digium.com/ |
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| 05:50.21 | jon-k | mustdie: will do, if this fixes the problem I'll even try and fix it |
| 05:50.25 | jon-k | (ie, in source) |
| 05:50.54 | jon-k | sorry, still changing dialplan - lots and lots of convoluted redirects and extensions here :D |
| 05:51.06 | jon-k | including breaking out some e1 lines onto a 4 port analogue card :D |
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| 05:56.40 | jon-k | okay |
| 05:56.56 | jon-k | dialplan changed and reloaded |
| 05:56.56 | jon-k | answers for all |
| 05:56.57 | jon-k | we'll run this for a day or two and see how it goes |
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| 06:01.02 | jon-k | what's an overlap call i wonder |