| 00:00.11 | asteriskn00b | so as many so I can have "press 999" for sales dept and so forth |
| 00:01.21 | three55ml | asteriskn00b: I'm not aware of a hard limit on the number of them, my only guess would be programming limits (variable sizes) |
| 00:01.35 | Sedorox | I have... 1 for sales.. 2 for techsupport... 3 to enter exten, 4 to repeat.. or something like that |
| 00:05.06 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 00:05.06 | *** mode/#asterisk [+o twisted] by ChanServ |
| 00:06.55 | *** join/#asterisk Chotaire (chotaire@nyc.us.chotaire.net) |
| 00:06.58 | Chotaire | morning all. |
| 00:07.03 | Chotaire | anyone has a copy of gphone for palmos?= |
| 00:07.22 | Chotaire | that vliusa.com site is somewhat slow, when I found the app, the company is probably already bankrupt. |
| 00:07.22 | shmaltz | if I do ifconfig eth0:1 where does the config for eth0:1 get saved to in slackware? |
| 00:08.05 | Chotaire | shmaltz: /proc/sys/net/etc.. ? or what do you mean? |
| 00:08.16 | Chotaire | doing that will NOT save the configuration. |
| 00:08.48 | shmaltz | Chotaire, the configs for eth0 is saved in /etc/rc.d/rc.inet1.conf where is the configs for eth0:1 saved? |
| 00:08.55 | *** join/#asterisk cybast1 (~cybast1@64.235.221.209) |
| 00:09.35 | Chotaire | so where is eth1 saved then? |
| 00:09.48 | blitzrage | Chotaire: /etc/sysconfig/network-scripts/ |
| 00:10.12 | Chotaire | will slackware read runtime configuration by cron and put them into the sysconfig? |
| 00:10.23 | blitzrage | slackware, no idea :) |
| 00:10.36 | pepzi | shmaltz: add your ifconfig-line yourself to the end of /etc/rc.d/rc.inet1 |
| 00:10.39 | Chotaire | me neither... shmaltz, if you want that config saved, ifconfig will not help. |
| 00:10.41 | shmaltz | in /etc/rc.d/rc.inet1.conf |
| 00:10.45 | Chotaire | you must edit the conf file. |
| 00:11.02 | Chotaire | yes, like pepzi says then ;) |
| 00:11.32 | Hogie | hey twisted, you awake? I want to give you some info that I figured out |
| 00:11.35 | shmaltz | the problem i'm facing is that it is saved already ( or so I think, b/c rc.inet1 stop and then start brings it back up), but I can't find it anywhere |
| 00:11.44 | shmaltz | webmin also reports that it is saved |
| 00:11.49 | shmaltz | but I have no clue where |
| 00:12.08 | Chotaire | grep for eth0:1 ? ;) |
| 00:12.32 | cybast1 | I have a question regarding the extensions.conf file . . . I have it successfully programmed up to route calls which are _91NXXXXXXXXXX out to a voip trunk, calls that are local (_9NXXXXXX) to a local analog trunk. These routes are working great. I then entered in the following to handle 911, 611, and 411 calls _9XXX,Dial(zap/8/${EXTEN:1}) but this doesn't seem to work . . . . the pbx waits for approx 10 secs then issues reo |
| 00:12.32 | cybast1 | reder tone |
| 00:12.36 | cybast1 | any ideas |
| 00:13.01 | Chotaire | shmaltz: try grep -r eth0:1 /etc/rc.d ; grep -r eth0:1 /etc/sysconfig |
| 00:13.32 | Hogie | cybast1: try _9X11, |
| 00:13.32 | Hogie | ? |
| 00:13.42 | cybast1 | ok will do |
| 00:14.00 | Chotaire | ok, let me retry... anyone got gphone for palmos? |
| 00:14.08 | Chotaire | I need that freeware software to test it on a treo 600. |
| 00:14.37 | cybast1 | nope that doesn't work either |
| 00:14.57 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
| 00:14.59 | cybast1 | does it matter the order in which I place this in the extensions file |
| 00:15.03 | Murkyl | <Bump> Anyone have a TDM400P with 4 FXO and Linux 2.6? I'm having problems with the hardware I think. |
| 00:15.51 | Hogie | that's weird, when i try dialing zap/g1/411, it says I must dial a 1, lol |
| 00:15.58 | three55ml | cybast1: Make sure you don't have ignorepat => 9 |
| 00:16.11 | cybast1 | i do have that |
| 00:16.29 | cybast1 | explain please |
| 00:16.29 | shmaltz | OK, thanks guys, Chotaire its in rc.modules |
| 00:16.40 | three55ml | It strips the 9 out |
| 00:16.41 | Chotaire | yup ok, I dont use slackware though ;) |
| 00:16.43 | cybast1 | I want to have 9 to get an outside line and then have to dial 911 |
| 00:16.44 | Chotaire | havce fun. |
| 00:16.52 | three55ml | Ahh |
| 00:17.11 | Sedorox | mine.. I don't have it stripping the 9 off.. I just leave it on |
| 00:17.14 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:17.15 | three55ml | I'm not sure, but I bet you're problem lies somewhere in there. The 9 is getting remvoed by ignorepat |
| 00:17.34 | cybast1 | if I just leave it as dialing 911 then too many acciedents can happen when one dials 91xxxxxxxx for long distance |
| 00:17.51 | three55ml | Yeah |
| 00:17.52 | cybast1 | If someone hits the 1 twice by accident |
| 00:18.05 | shmaltz | cybast1, which number starts with 11? |
| 00:18.10 | three55ml | There's a way around it I believe, but I don't know it off the top of my head. |
| 00:18.22 | cybast1 | I am not actually dialing 911 for obvious reason I am trying 411 |
| 00:18.29 | Hogie | isn't that why you match it again _91N ? |
| 00:18.33 | shmaltz | last time I checked there isn't a number in the north american dialplan that starts with 11 |
| 00:18.37 | *** join/#asterisk mrproper_ (~b@61.95.55.242) |
| 00:18.55 | sivana | 011 to exit north america |
| 00:18.57 | Kumbang | guys, can i make back to back test call mfcr2 with unicall in TE400P? |
| 00:18.58 | shmaltz | Hogie, that is correct |
| 00:19.08 | mrproper_ | any ideas why im getting: [chan_h323.so]Apr 13 09:08:10 WARNING[3632]: loader.c:305 __load_resource: /usr/lib/asterisk/modules/chan_h323.so: undefined symbol: _ZNK20H323_RealTimeChannel17GetRTPPayloadTypeEv |
| 00:19.11 | Sedorox | but you need the 0 |
| 00:19.12 | shmaltz | sivana, exactly but not 911 |
| 00:19.12 | cybast1 | no number starts with 11 but if someone dials 9 to get an outside line followed by 1 for long distance and then by accident hits the 1 again . .voila police are on the way |
| 00:19.31 | Hogie | cybast1: you can call 911 as long as you state what you are doing. I do it to test that stuff when we make changes to our key system here. Just tell them you are testing for the pbx to make sure stuff is allowed out and ask them for the line info and they usually give you the address and name and such |
| 00:19.35 | sivana | ya |
| 00:19.46 | Hogie | cybast1: _91N |
| 00:19.48 | Hogie | wont match 911 |
| 00:19.52 | sivana | that's why we didn't do 9 for outside lines |
| 00:19.54 | Hogie | N = 2-9 |
| 00:20.05 | cybast1 | Iwhat do you use for outside lines then |
| 00:20.17 | sivana | nothing.. just dial it direct |
| 00:20.23 | shmaltz | cybast1, and if they bump into a police man on the streeet and by mistake the police mans gun lands in their pocket when they are on their way to the airport and voila they are in a federa prison |
| 00:20.28 | cybast1 | I have in _9XXX |
| 00:20.57 | want561or772did | true, that's very likely |
| 00:21.17 | cybast1 | it seems like the pbx is waiting for more digits like it didn't make a match |
| 00:21.20 | shmaltz | cybast1, the mistake you describe will never happen to the point that the call is completed |
| 00:21.50 | sivana | ya, certainly before the digit timeout |
| 00:22.05 | sivana | they'd have to pause long enough |
| 00:22.21 | cybast1 | so hen what should I put in then |
| 00:22.24 | cybast1 | then |
| 00:22.33 | cybast1 | _X11 |
| 00:22.48 | cybast1 | and what should I use for an outside line |
| 00:23.06 | sivana | _9NXXXXXX |
| 00:23.15 | shido6 | feeding time |
| 00:23.33 | cybast1 | sivana thats for a local number |
| 00:23.39 | cybast1 | i have that |
| 00:23.47 | three55ml | _9NXXNXXXX |
| 00:23.47 | sivana | _91NXXXXXXXXX |
| 00:23.51 | three55ml | hehe :) |
| 00:24.10 | cybast1 | I have those two entries and they work fine |
| 00:24.27 | cybast1 | it's just the _9XXX that's not working |
| 00:24.43 | sivana | why do you have three Xs? |
| 00:24.45 | cybast1 | does it matter the order I put the entries in the extensions file |
| 00:24.58 | Murkyl | I took a funny way out with 911 calls while using 9 as an outside line. I have 911 mapped to a voice prompt and 9911 dialing direct. |
| 00:24.59 | cybast1 | X = any number correct?? |
| 00:24.59 | sivana | yes, top down it reads |
| 00:25.09 | three55ml | cybast1: Yes |
| 00:25.21 | shmaltz | sivana, no it doesn't read top down |
| 00:25.30 | sivana | yes it does |
| 00:25.33 | cybast1 | so then I should put the _9XXX befor ethe _9NXXXXXXX |
| 00:25.42 | sivana | and first match |
| 00:26.11 | shmaltz | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting |
| 00:26.11 | sivana | why do you need _9XXX? |
| 00:26.12 | PatrickDK | heh, that doesn't matter |
| 00:26.15 | Murkyl | sivana: I think it just so happens to process that way but * does not guarantee. Not 100% sure. Need to check handbook. |
| 00:26.21 | want561or772did | you probably want _9X11 |
| 00:26.25 | shmaltz | sivana look at the link |
| 00:26.25 | want561or772did | for 611, 411, 511, etc |
| 00:26.27 | Murkyl | For 411 as an example |
| 00:26.29 | three55ml | _9NXX |
| 00:26.45 | PatrickDK | _9XXX or _9NXXXXXXXX will match correctly in any order |
| 00:26.46 | three55ml | 111 isn't valid |
| 00:26.47 | want561or772did | or _9N11 |
| 00:26.59 | sivana | silly |
| 00:27.03 | PatrickDK | it depens on how many numbers you type before the timeout |
| 00:27.08 | shmaltz | sivana, whats silly |
| 00:27.11 | Murkyl | three55ml: I would be inclined to add 911 as a valid external dial. |
| 00:27.13 | cybast1 | II'm trying switching the order |
| 00:27.22 | Murkyl | Someone may not know that you need a 9 for an outside line. |
| 00:27.27 | shmaltz | cybast1, take a look: |
| 00:27.29 | shmaltz | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting |
| 00:27.33 | three55ml | Murkyl: I'm saying _9XXX would allow you to dial 9,111 which isn't really needed. |
| 00:27.49 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
| 00:27.52 | *** join/#asterisk FengShui (~ted@gray.impulse.net) |
| 00:28.01 | Murkyl | Well.. 111 is not a valid service now, but it may be later. Or it may also depend on your locality |
| 00:28.08 | three55ml | As far as I know, shmaltz is correct. If you put the _9NXXNXXXX first - 9,911 would be considered a match until the timout occurs. |
| 00:28.18 | three55ml | Putting it first would be a better scenario. |
| 00:28.30 | cybast1 | sivana's right |
| 00:28.30 | tainted- | three55ml how's premierepbx going |
| 00:28.37 | cybast1 | changing the order worked |
| 00:28.39 | PatrickDK | three55ml, order doesn't matter at all |
| 00:28.46 | shmaltz | threee55ml, and everybody else, putting first does *NOT* help |
| 00:28.46 | PatrickDK | asterisk changes the order when it loads the config |
| 00:28.51 | cybast1 | I used _9X11 |
| 00:29.01 | three55ml | tainted-: It's good |
| 00:29.02 | shmaltz | *READ THIS* |
| 00:29.04 | shmaltz | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting |
| 00:29.10 | cybast1 | and placed the entry befor ethe local and long distance entries |
| 00:29.10 | three55ml | tainted-: Testing out DID providers to get someone solid |
| 00:29.15 | P-Chan | What would cause iax2 show peers to have OK on both sides, but NOTICE[26752]: chan_iax2.c:5441 socket_read: Rejected connect attempt from 68.142.66.200? |
| 00:29.15 | want561or772did | use _9N11 if you want to exclude that pattern from catching 911 |
| 00:29.22 | tainted- | three55ml that's a toughie |
| 00:29.30 | three55ml | tainted-: :) |
| 00:29.30 | tainted- | three55ml how much are u going to charge for a license |
| 00:29.34 | cybast1 | good one want |
| 00:29.46 | three55ml | tainted-: I haven't come up pricing yet for licensing it out. |
| 00:29.51 | sivana | cybast1: I think 'show dialplan' will output the order? |
| 00:29.52 | shmaltz | gtg guys |
| 00:30.00 | shmaltz | sivana, it should |
| 00:30.07 | tainted- | three55ml if it's a solid product, i'd like to use it |
| 00:30.13 | Murkyl | shmaltz: Later! |
| 00:30.14 | shmaltz | hoever, don't bet on it next time you load asterisk |
| 00:30.20 | three55ml | tainted-: Have I talked to you about it before? I don't remember. |
| 00:30.23 | tainted- | three55ml i could code it all myself.. but i'm busy doing other stuff |
| 00:30.25 | *** join/#asterisk dizzydiffi (dizzydiffi@adsl-70-240-211-145.dsl.hstntx.swbell.net) |
| 00:30.28 | dizzydiffi | hello |
| 00:30.30 | three55ml | tainted-: I know how that goes. |
| 00:30.34 | sivana | show dialplan show's you the "sorted" plan? |
| 00:30.44 | tainted- | three55ml yea i made fun of your design (37signals) |
| 00:30.55 | three55ml | Oh yeah :) Let me show you the new one...I think you'll like it. |
| 00:31.11 | cybast1 | THANKS GUYS |
| 00:31.13 | three55ml | Sent it to you in a message, don't want it in the channel |
| 00:31.28 | tainted- | yea u've got the touch for good design |
| 00:31.44 | tainted- | banner pics are a bit dark for the rest of the page |
| 00:31.44 | cybast1 | is there a 'more' or 'less' equivilnt at the CLI |
| 00:31.54 | tainted- | not sure if u are going for contrast.. maybe reduce opacity a bit? |
| 00:32.03 | tainted- | otherwise looks very good! |
| 00:32.06 | dizzydiffi | help i need to figure out how to \make SIP to H323 calls through asterisk |
| 00:32.17 | three55ml | tainted-: Yeah, still playing with it some more. Getting there. |
| 00:33.22 | tainted- | u use imageready? |
| 00:33.52 | three55ml | tainted-: For the rolloevers? No. If that's what you were talking about then yeah, I don't like them right now (the plan buttons). |
| 00:33.56 | cybast1 | how do you unblacklist a phone number? |
| 00:34.06 | dizzydiffi | people |
| 00:34.09 | three55ml | tainted-: I had the HTML ImageReady generates. |
| 00:34.39 | cybast1 | help |
| 00:34.42 | christo | three55ml - I'm closer to the casue now. The System() command dies when I call a script with the following line (even if it's the only line in the script) : |
| 00:34.45 | christo | resample -by 5.5125 /var/spool/asterisk/monitor/$1-in.wav /var/spool/asterisk/monitor/$1-in-upped.wav 1>/dev/null |
| 00:35.09 | three55ml | christo: Try > /dev/null 2>&1 at the end |
| 00:35.11 | cybast1 | sorry wrong window I was trying to type help into asterisk |
| 00:35.39 | three55ml | christo: I assume you're trying to run it in the background, right? |
| 00:35.46 | sivana | hehe |
| 00:36.10 | tainted- | three55ml only virtual pbx or going to do pbx boxen too? |
| 00:36.25 | P-Chan | if iax2 show peers both say ok, and when calling out from an asterisk server which trunks to one with a PRI you get the following message: |
| 00:36.27 | P-Chan | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT |
| 00:36.27 | P-Chan | <PROTECTED> |
| 00:36.27 | P-Chan | <PROTECTED> |
| 00:36.31 | P-Chan | What does that mean? |
| 00:36.34 | christo | three55ml - I've tried with > /dev/null 2>&1 - I'm not running in the background. Should I be specifying that when I call System() ? |
| 00:36.50 | three55ml | tainted-: Both. |
| 00:37.02 | three55ml | christo: Try removing the 1>/dev/null completely and see what happens. |
| 00:37.07 | three55ml | Have you tried that? |
| 00:37.11 | christo | yeah |
| 00:37.13 | christo | same thing |
| 00:37.26 | three55ml | Is resample in the global path? |
| 00:37.28 | christo | can I put a '&' at the end to background it I wonder? |
| 00:37.34 | three55ml | Try putting the full path to it |
| 00:37.39 | christo | yes it is |
| 00:37.46 | christo | will try that anyway - just to be sure |
| 00:37.53 | three55ml | > /dev/null 2>&1 & <---- That would make it totally in the background and output nothing |
| 00:38.48 | dizzydiffi | does anyone now how to make H323 phone call a SIP phone |
| 00:38.49 | cybast1 | I hit *60 thinking it was another feature and it added the last incoming caller to my blacklist . .how do I remove this entry |
| 00:38.59 | christo | three55ml - oooh |
| 00:39.11 | christo | pushing to the background seems to help.... |
| 00:39.29 | file[mac] | cybast1: what phone? |
| 00:39.47 | file[mac] | or ATA... |
| 00:39.51 | three55ml | cybast1: help database - if you did it in Asterisk. You can delete it manually, not sure of the command off the top of my head. |
| 00:39.51 | cybast1 | on my analog phone |
| 00:40.08 | cybast1 | connected to a tdm400 |
| 00:40.18 | cybast1 | a call came in the trunk |
| 00:40.28 | file[mac] | ah ic, you must have some dialplan that has some feature codes or whatever... do what three55ml said to find the entry |
| 00:40.31 | cybast1 | I hit *60 and blacklisted it |
| 00:40.42 | christo | three55ml - however, this opens up an interesting connumdrum. the rest of the script assumes that the 'backgrounded' line has finished running (it actually takes a couple seconds) so that means I'm going to have to make the script a helluva lot more complex to ensure that it's done |
| 00:40.47 | cybast1 | ok |
| 00:40.54 | cybast1 | I'll try help database |
| 00:41.24 | cybast1 | yes . . if I do database show it shows the entry |
| 00:42.20 | cybast1 | it requires a family and key to delete the entry |
| 00:42.28 | *** join/#asterisk odie_flocon_ (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 00:42.31 | three55ml | christo: :) So if it's not in the background, it doesn't work at all? I was going to say you could make it say "Please wait..." or similar. |
| 00:42.32 | cybast1 | what is the family and key? |
| 00:42.33 | odie_flocon_ | hello all. |
| 00:42.36 | three55ml | cybast1: It should display that. |
| 00:42.38 | three55ml | Let me look |
| 00:42.51 | odie_flocon_ | hey anybody know the default admin password for a polycom ip600 phone? |
| 00:42.59 | Murkyl | 456 |
| 00:43.07 | Murkyl | odie:_flocon_: Default is 456 I believe. |
| 00:43.36 | odie_flocon_ | ok thanks. |
| 00:43.42 | odie_flocon_ | I'll try it in the morning. |
| 00:43.50 | odie_flocon_ | I just got one in for work. :D |
| 00:44.35 | cybast1 | got it |
| 00:44.40 | three55ml | cybast1: Cool |
| 00:44.48 | cybast1 | thanks again guys |
| 00:44.52 | three55ml | No problem |
| 00:45.05 | Murkyl | Excellent! What was the fix? |
| 00:45.56 | cybast1 | are you talking to me Murkyl? |
| 00:46.07 | Murkyl | cybast1: Yup. |
| 00:46.34 | cybast1 | I did a datashow show whixh returned /blacklist/phonenumber |
| 00:46.53 | cybast1 | so I entered database del blacklist phonenumber |
| 00:47.07 | cybast1 | blacklist was the family and the phone number was the key |
| 00:47.16 | Murkyl | cybast1: Ok. filing it away somewhere in my mind for future reference. :) |
| 00:47.37 | cybast1 | hit *60 it will blacklist the last incoming call |
| 00:47.45 | cybast1 | then you can try it |
| 00:48.07 | christo | three55ml - corrrect. it just doesn't work[tm] if it's not in the background by the looks of things |
| 00:48.15 | cybast1 | speaking of feature activation codes is there any definitive guide to all the feature activation codes in asterisk?? |
| 00:48.52 | AgiNamu | OK, can someone tell me when printf("%x", bytes[i]) can give a value larger than 255? bytes is unsigned char * |
| 00:49.02 | file[laptop] | in asterisk there is none fyi... |
| 00:49.12 | sivana | AgiNamu: maybe try -dev :P |
| 00:49.16 | file[laptop] | chan_zap has some though |
| 00:49.19 | AgiNamu | heh |
| 00:49.32 | file[laptop] | like disabling callerid, do not disturb |
| 00:49.37 | AgiNamu | cybast1, google for "Verticle Service Codes" |
| 00:49.38 | cybast1 | it will give ou up to 0xFF |
| 00:49.41 | AgiNamu | NANPA defines them |
| 00:49.45 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 00:49.50 | cybast1 | thanks AgiNamu |
| 00:49.57 | file[laptop] | there's only a few implemented in chan_zap though |
| 00:49.58 | AgiNamu | cybast1, well, it's printing "fffffff2 1e ffffffc5 ffffff91" (calling it in a loop) |
| 00:50.09 | Mavvie | try %ux |
| 00:50.33 | Mavvie | no |
| 00:50.34 | AgiNamu | now it outputs "4294967282x 30x 4294967237x 4294967185x" |
| 00:50.46 | Mavvie | %hhx |
| 00:50.50 | Mavvie | according to printf(3) |
| 00:50.50 | cybast1 | perhaps somehow you;re getting an address for a pointer and not the contents |
| 00:50.52 | sivana | try %*&#^$ |
| 00:50.54 | Mavvie | <PROTECTED> |
| 00:50.54 | Mavvie | <PROTECTED> |
| 00:50.56 | cybast1 | weird |
| 00:51.14 | AgiNamu | Mavvie, now its printing "fff2 1e ffc5 ff91 ff8c" |
| 00:51.25 | cybast1 | are you increment i and going outside the buffer |
| 00:51.28 | odie_flocon_ | hmmmm interesting. |
| 00:51.57 | cybast1 | you should it's a unsigned char . . . a char will give you weird results like that |
| 00:51.58 | AgiNamu | I even tried by assigning a char tmp |
| 00:52.00 | Murkyl | AgiNamu: is your var definition: unsigned char *var1; ? |
| 00:52.03 | P-Chan | Well, I'm almost out of the woods. Thanks for everyone's help today - especially Sedorox! ;) I'm outta here for today. |
| 00:52.03 | odie_flocon_ | AgiNamu, did you pick that name cuz you are good at AGI? |
| 00:52.04 | cybast1 | once you get over 7F |
| 00:52.08 | Mavvie | AgiNamu: let me try |
| 00:52.12 | *** join/#asterisk NormAst (~NormAst@toronto-HSE-ppp3959569.sympatico.ca) |
| 00:52.13 | AgiNamu | odie_flocon_, nope |
| 00:52.18 | sivana | hey Norm |
| 00:52.25 | NormAst | Hay. |
| 00:52.30 | odie_flocon_ | hey nrom. |
| 00:52.38 | odie_flocon_ | i mean norm |
| 00:53.04 | AgiNamu | Murkyl, yep, it's printbytes(unsigned char *ciphertext, int length) { for... printf("%x", ciphertext[i]); } |
| 00:53.09 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
| 00:53.47 | sudhir492 | a little quiet here today |
| 00:53.51 | NormAst | Very.. |
| 00:54.15 | cybast1 | that should work |
| 00:54.19 | NormAst | Found out today that using the monitor Application creates echo on calls... when doin' TDM to TDM bridge |
| 00:54.34 | Mavvie | AgiNamu: http://pastebin.ca/9452 |
| 00:54.43 | tainted- | NormAst what about voip |
| 00:54.44 | sudhir492 | NormAst: That is very weird |
| 00:55.05 | NormAst | Yea... But i creates about 60ms of delay. |
| 00:55.13 | NormAst | it |
| 00:55.21 | tainted- | over voip? |
| 00:55.24 | tainted- | or tdm |
| 00:55.28 | want561or772did | quiet at voipjet too. they don't answer their email |
| 00:55.32 | NormAst | no... TDM to TDM bridged. |
| 00:55.47 | sudhir492 | How about SIP-TDM calls. Do they also suffer with echo? |
| 00:55.50 | tainted- | NormAst what's your cpu util |
| 00:56.02 | cybast1 | try printf("%X", *((unsigned char*) (cipertext + i))); |
| 00:56.04 | NormAst | P4 2.8gig. 512megs ram |
| 00:56.11 | cybast1 | just for s + g |
| 00:56.25 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 00:56.40 | *** join/#asterisk Grooby (~Grooby@66-205-89-194.in-addr.net1plus.com) |
| 00:56.41 | NormAst | remove the monitor from the dial plan and no echo |
| 00:57.08 | Murkyl | AgiNamu: Pure C code right? |
| 00:57.18 | AgiNamu | Murkyl, yea |
| 00:57.45 | AgiNamu | seems to be working now, with what cybast sent |
| 00:57.52 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 00:57.54 | cybast1 | AgiNami . . found vertical codes on the net bt can't find one for hold |
| 00:58.12 | Mavvie | AgiNamu: but keep in mind, that casting doesn't resolve the problem, only removes the symptons. |
| 00:58.19 | AgiNamu | cybast1, hold can't be implemented as a VSC |
| 00:58.32 | AgiNamu | Mavvie, yea. I'm trynna figure out wtf is going on. I guess I'll put it into the debugger. |
| 00:58.34 | cybast1 | I don;t have a hold buton on my phone |
| 00:58.55 | sudhir492 | What is the best way to delete a few voicemail files selectively from the /var/spool... directory? Even if I rm a few files from INBOX directory, voicemail application thinks there is no voicemail left |
| 00:59.12 | cybast1 | It sound like you've got some other than a pointer to a unsigne dchar |
| 00:59.15 | AgiNamu | printf("%hhx ", ciphertext[i]); seems to work |
| 00:59.25 | cybast1 | something is getting messed u[ |
| 00:59.37 | Murkyl | cybast1: I agree. It seems to think the pointer is larger than a byte. |
| 00:59.38 | AgiNamu | regardless WHAT it was, if I do "char tmp = fuckedPointer[i]" then there's no way tmp can be more than 0xff |
| 00:59.45 | cybast1 | yep |
| 00:59.58 | Murkyl | And char tmp still produces the wrong result? |
| 00:59.58 | Mavvie | or more than 0x7f |
| 01:00.03 | AgiNamu | yea |
| 01:00.17 | cybast1 | hmmm |
| 01:00.18 | AgiNamu | the char should be zero extended to the 32-bit int that "%x" expects. |
| 01:00.23 | Murkyl | Agi: Try unsigned char tmp. |
| 01:00.31 | Murkyl | Signed char tmp would produce 0xfffff7 |
| 01:00.38 | cybast1 | yes |
| 01:00.55 | cybast1 | for anything over F |
| 01:00.59 | cybast1 | 7F |
| 01:01.02 | cybast1 | that is |
| 01:01.08 | AgiNamu | oh yes, that's true |
| 01:01.21 | AgiNamu | i musta done char tmp instead of unsigned char tmp |
| 01:01.45 | AgiNamu | OK, thanks guys. I think i worked thru it |
| 01:01.52 | cybast1 | that would explain what you get the right result when you cast it |
| 01:02.03 | AgiNamu | Now i can get to real debuggin :P |
| 01:02.30 | cybast1 | So the only way to implement hold is througha physical hold button on a phone? |
| 01:03.42 | AgiNamu | no |
| 01:03.43 | AgiNamu | flash |
| 01:03.45 | AgiNamu | hookflash |
| 01:03.54 | AgiNamu | but your ATA or phone still needs to implement it |
| 01:03.59 | cybast1 | flash will do it |
| 01:04.06 | cybast1 | ?? |
| 01:04.11 | AgiNamu | Theoretically, you can do it by having Asterisk monitor all your audi |
| 01:04.21 | AgiNamu | and checking for DTMF tones, but that's a baaad hack |
| 01:04.27 | AgiNamu | it'll fuck up say, when you call someone's IVR |
| 01:04.31 | AgiNamu | and it's impractical and a waste of CPU |
| 01:04.40 | AgiNamu | right, your device's firmware must handle hold |
| 01:04.47 | AgiNamu | I suggest buying the PA168 |
| 01:04.55 | tainted- | AgiNamu how's res_mono? ;) |
| 01:04.55 | AgiNamu | The PA168 is going to be the best freaking phone out there, feature wisde. |
| 01:05.12 | tainted- | who makes pa168 |
| 01:05.17 | AgiNamu | tainted-, heh, forget that. A: Asterisk API is undocumented, making an efficient implmentation a HUGE task (i.e., must document API first) |
| 01:05.21 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:05.25 | AgiNamu | B: Asterisk API changes too much to make anything stable |
| 01:05.36 | tainted- | tell me about it |
| 01:05.36 | AgiNamu | C: bkw_ says there are some showstopper threading issues |
| 01:05.44 | AgiNamu | Centrality Communications |
| 01:05.48 | AgiNamu | PA168 does IAX2 |
| 01:05.51 | dikadika | my auto attendant message is a little on the quite side, is tehre a way to kick up the volume a couple notches? |
| 01:06.06 | AgiNamu | GSM, G723, G729, ULAW, ALAW, soon iLBC, and in the future, Speex is planned |
| 01:06.23 | AgiNamu | IAX2 is getting full features implemented. native transfer will be in the next firmware release |
| 01:06.30 | AgiNamu | attended and so on should follow soon |
| 01:06.40 | tainted- | AgiNamu there are a ton of manufacturers |
| 01:06.46 | AgiNamu | tons |
| 01:06.46 | tainted- | with all those codecs |
| 01:06.49 | AgiNamu | like 20 different reference designs |
| 01:06.50 | AgiNamu | yep |
| 01:06.56 | tainted- | what makes it good |
| 01:06.58 | AgiNamu | I also plan to exten it to handle new ideas for IAX2. i.e., adding a "redirect" IE to commands |
| 01:07.12 | AgiNamu | it's the only IAX2 device out there, for starters. It also does H323, SIP, MGCP, and N2P |
| 01:07.44 | tainted- | i'm not so sure on that (IAX2) |
| 01:07.53 | tainted- | i could swear i've seen a few out there |
| 01:08.16 | want561or772did | dikadika: use the "sox" audio processor to batch increase the volume (option -v) |
| 01:08.17 | file[laptop] | they all use the PA168 chipset |
| 01:08.27 | AgiNamu | tainted-, actually touch one? |
| 01:08.36 | file[laptop] | it's in ATAs, phones, whatever... |
| 01:08.37 | AgiNamu | Virbiage claims they will have one. FarFon too |
| 01:08.43 | dikadika | want561or772did, does that work even if i dont have a sound card working/installed |
| 01:08.53 | file[laptop] | I forgot about the farfon, eep |
| 01:08.56 | AgiNamu | I talked to another dev, who says he has multiple audio streams (i.e., 3-way calling) on the PA168 |
| 01:08.56 | file[laptop] | and virbiage... haha |
| 01:09.03 | AgiNamu | As well as MOH via ShoutCast |
| 01:09.06 | AgiNamu | for IAX2 on PA168 |
| 01:10.33 | want561or772did | yes dikadika |
| 01:10.39 | dikadika | want561or772did, thanks |
| 01:11.27 | hermie | _The Office_ is on tonight |
| 01:12.02 | want561or772did | voipjet lives. voipjet just answered my email in a very courteous way |
| 01:12.09 | want561or772did | maybe they're watching :o |
| 01:12.55 | *** join/#asterisk kielstirling (~kiel@knss.net) |
| 01:13.40 | kielstirling | Hi all, can anyone help me with a caller id problem ?? I'm in Australia? |
| 01:13.51 | Mavvie | sure. |
| 01:15.13 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
| 01:15.14 | *** join/#asterisk Legend (~Legend@24.244.142.134) |
| 01:15.58 | shido6 | ls ls ls |
| 01:16.03 | shido6 | la la la |
| 01:16.11 | shido6 | aussie land. |
| 01:17.00 | tainted- | shido6 rip any people off lately? |
| 01:17.11 | Damin | Third: I don't fell me as a stealer ! I'm felling well in my basket ! Unstead complain, feel free to make a donation for the hard *work* I have done ! And for the fact that a patch will be released to the public ! |
| 01:17.17 | dizzydiffi | please please |
| 01:17.33 | dizzydiffi | i need help with get H323 and Sip to talk on asterisk |
| 01:17.38 | dizzydiffi | someone |
| 01:18.04 | _GiGi_ | hm, asterisk have stable h323 channel ? |
| 01:18.10 | kielstirling | Sorry got abit lost .. How can I debug a caller id problem? I want to be able to identify if the modem I have support AU caller ID? I'm using FreeBSd |
| 01:19.01 | dizzydiffi | really |
| 01:19.17 | dizzydiffi | i get some crazy error when i compile the Oh323 code |
| 01:19.37 | dizzydiffi | asteriskauio.o cannot be found |
| 01:19.43 | kielstirling | In AU we use Bell however the delay before sending the CND is a bit longer and I have read some modems don't support this |
| 01:20.38 | Mavvie | oh oh... analogue lines. |
| 01:20.55 | Mavvie | not worth the hassle :-) |
| 01:21.03 | kielstirling | sorry yeah |
| 01:21.47 | kielstirling | so no one can help me ??? :( |
| 01:25.08 | *** part/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 01:31.16 | *** join/#asterisk PBXtech[mobile] (~upirc@wirelessdata-167-248.mycingular.net) |
| 01:31.37 | *** part/#asterisk Murkyl (~Murkyl@69.229.154.213) |
| 01:32.13 | want561or772did | voip-info.org is kvetching |
| 01:32.49 | *** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net) |
| 01:32.49 | tzanger | want561or772did: it always is |
| 01:33.33 | PBXtech[mobile] | drugs |
| 01:34.50 | Q-At-Home | greetings |
| 01:35.09 | PBXtech[mobile] | Q |
| 01:35.19 | Q-At-Home | been a long time :) |
| 01:35.26 | want561or772did | do you guys have a trick for shutting off Playtones(dial) when the first digit is hit |
| 01:35.43 | want561or772did | my current solution is a nest of extensions |
| 01:36.34 | Q-At-Home | I've got a really strange strange problem, all of a sudden, if I take an inbound call from a wctdm fxo via a wctdm fxs... and hang up the call, the fxs starts ringing with a phantom call... its driving me nuts. |
| 01:36.53 | NormAst | Can I do Dial(sip:username:password@host/${EXTEN}) ??? |
| 01:37.17 | want561or772did | i guess i just use DISA with no password |
| 01:37.21 | want561or772did | duh! |
| 01:37.22 | *** join/#asterisk tito (~tito@home.txzone.net) |
| 01:37.40 | tito | hi people |
| 01:38.05 | tito | i have some trouble with asterisk |
| 01:38.05 | PBXtech[mobile] | tito you suck. ken shamrock wooped ya |
| 01:38.12 | bkw_ | asdf |
| 01:38.14 | tito | mouhaha. |
| 01:38.22 | tito | Apr 13 03:34:53 NOTICE[5269]: app_dial.c:759 dial_exec: Unable to create channel of type 'Zap' |
| 01:38.35 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 01:38.44 | tito | i have try google, forum, wiki |
| 01:38.56 | tito | and no issue to ride this error |
| 01:39.15 | PBXtech[mobile] | not much info there tito |
| 01:39.30 | tito | what do you want ? |
| 01:39.43 | PBXtech[mobile] | could be in use could be configured bad |
| 01:39.47 | tito | asterisk-1.0.6 on 2.6.8-2-686 |
| 01:40.00 | PBXtech[mobile] | hmmm dinner arrived |
| 01:40.03 | tito | seem good configured |
| 01:40.16 | *** part/#asterisk PBXtech[mobile] (~upirc@wirelessdata-167-248.mycingular.net) |
| 01:42.02 | *** join/#asterisk stormfr (~StorM@82.66.251.138) |
| 01:45.56 | stormfr | hello, |
| 01:45.59 | *** join/#asterisk exonic (~exonic@c-24-11-127-28.hsd1.mi.comcast.net) |
| 01:46.29 | exonic | Hey all, i'm looking for a way to dial in a channel, making two bound together from AGI. Is this possible? Would it be possibel in the manager? |
| 01:46.30 | stormfr | how is it possible to tune sip communication for high latency network with * ? |
| 01:46.45 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 01:47.07 | exonic | stormfr, I would experiment with various codecs. |
| 01:47.20 | exonic | sorry not the best answer but I really don't know. |
| 01:47.56 | tito | exonic, can you help me ? |
| 01:47.57 | stormfr | for my case it's satellite communication |
| 01:48.06 | stormfr | 723 have the best quality for this latency |
| 01:48.27 | stormfr | exonic: you look for initiate to call and bridge them ? |
| 01:48.35 | stormfr | to/two |
| 01:48.44 | exonic | yea |
| 01:48.57 | stormfr | i have read something about it in ml or wiki |
| 01:49.04 | exonic | but the trick is i'd like to be inside an AGI (python script) |
| 01:49.20 | exonic | thanks, i'll do some digging |
| 01:49.25 | stormfr | if i remenber one said to put them in a conference but i guess there is another way |
| 01:49.30 | stormfr | else is to use callfile |
| 01:50.04 | exonic | oh, really. Hmm.. that's actually a good idea |
| 01:50.34 | stormfr | with callfile it's not a problem but you will not be able to handle return code etc :/ |
| 01:51.04 | exonic | callfile is something placed in /var/spool/asterisk/outoing/ correct? |
| 01:51.10 | stormfr | yes |
| 01:51.25 | stormfr | you can easyly generate it by agi |
| 01:51.33 | stormfr | depend of your application need. |
| 01:51.43 | stormfr | you can monitor the call after launch it by manager |
| 01:52.13 | *** join/#asterisk Hydroxide (user@Hydroxide.developer.debian) |
| 01:52.23 | exonic | cool, i'll look into that. Thanks |
| 01:52.35 | Hydroxide | when using the Record application, I'm having it record silence. (it's not a 0-byte file, it's silence that's as long as I recorded for.) |
| 01:53.08 | Hydroxide | what might be wrong? I can play pre-recorded or synthesized sounds OK, and I can use the phone and asterisk to have two-way conversations |
| 01:55.49 | exonic | Hydroxide, you expect it to be a 0-byte file? Why? what about background noise. |
| 01:56.05 | Hydroxide | exonic: I don't expect it to be a 0 byte file. I was pointing out that some recording is happening |
| 01:58.00 | exonic | sorry I didn't read it right. So .. I don't know the problem. I am confused. |
| 01:58.07 | yaboo | anyone own the soyo n400s gateway fxs unit? |
| 01:59.34 | *** join/#asterisk tessier (~treed@222.253.78.234) |
| 01:59.34 | Q-At-Home | speaking of hardware, hows the sipura 841 |
| 01:59.49 | file[laptop] | I dislike it's LCD |
| 02:00.21 | Q-At-Home | whats it worth? |
| 02:00.51 | file[laptop] | what does thou mean? |
| 02:00.58 | three55ml | Anyone heard anything about that new Grandstream? I think it looks pretty nice. |
| 02:01.05 | Q-At-Home | bucks to buy :) |
| 02:01.17 | file[laptop] | I saw one at VON... Brian Capouch had it... |
| 02:01.20 | Q-At-Home | ah |
| 02:01.27 | file[laptop] | besides the LCD and speakerphone, it was a decent phon |
| 02:01.29 | file[laptop] | er phone |
| 02:01.33 | Q-At-Home | I'm trying to get our place out of the "cisco is the only voip" |
| 02:01.36 | Q-At-Home | mentality |
| 02:01.43 | denon | file[laptop]: besides the LCD and speakerphone? so, what .. it had a nice handset cord? |
| 02:01.48 | Q-At-Home | I was going to bring in a budgetone :) |
| 02:01.58 | file[laptop] | denon: quality was fine, it worked :p |
| 02:02.09 | denon | heheh .. barbietones work too |
| 02:02.10 | denon | usually |
| 02:02.27 | Sedorox | Q-At-Home: do yourself a favor |
| 02:02.30 | Sedorox | for a qork enviroment |
| 02:02.33 | Sedorox | work* |
| 02:02.37 | Sedorox | don't get a budgetone |
| 02:02.39 | Q-At-Home | hahaha |
| 02:02.43 | Q-At-Home | I know, I was being an ass |
| 02:02.52 | Q-At-Home | I have 2 of em... in a drawer |
| 02:02.56 | Sedorox | The Aastra's look like nice phones... |
| 02:03.01 | three55ml | Sedorox: Yeah |
| 02:03.02 | Q-At-Home | I have a sayson |
| 02:03.05 | Q-At-Home | i480 |
| 02:03.07 | Q-At-Home | love it |
| 02:03.11 | Sedorox | lol.. I have one here.. probably gonna get a Aastra when I get the time and try it out... |
| 02:03.22 | Q-At-Home | poe is nice |
| 02:03.33 | tzanger | i has no voip phone... i's poe |
| 02:03.33 | Q-At-Home | in the sayson |
| 02:03.54 | Legend | Sedorox: read the wiki on the aastra, they have fallen on their asses with the firmware |
| 02:04.09 | Q-At-Home | I've heard rumor of an 8 call apperarance firmware |
| 02:04.14 | Q-At-Home | sayson == aastra? |
| 02:04.20 | Sedorox | hmmmm |
| 02:04.33 | three55ml | Q-At-Home: I believe so |
| 02:04.46 | Sedorox | Legend: well like I said.. they look nice |
| 02:04.50 | Sedorox | doesn't mean they work great tho :-p |
| 02:05.27 | Legend | Sedorox: yes, i have one, and it is built well, and has the potential to be a GREAT phone, but the firmware is barebones |
| 02:06.04 | Sedorox | well I heard it doesn't do xml the best.. but I mean.. I'm just looking for a more advanced phone then the budgetone with several lines too |
| 02:06.16 | Legend | it doesn't do xml |
| 02:07.13 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 02:07.24 | Hydroxide | does recording sound from a zap channel have any dependency on the system's sound card? |
| 02:07.34 | Sedorox | <PROTECTED> |
| 02:07.34 | Hydroxide | sorry |
| 02:07.36 | Hydroxide | sip channel |
| 02:07.38 | Sedorox | hehe.. that is the one I'm looking at |
| 02:07.46 | Hydroxide | does recording sound from a sip channel depend on the asterisk server's sound card? |
| 02:07.53 | Legend | Sedorox: yes, it doesn't do xml |
| 02:08.01 | Sedorox | yea.. but whats xml used for? |
| 02:08.10 | Sedorox | make custom stuff on the display? |
| 02:08.28 | Legend | Sedorox: yes, the phone is VERY limited right now |
| 02:08.30 | tito | open("/dev/zap/ctl", O_RDWR) = 3 |
| 02:08.33 | tito | ioctl(3, 0x40244a12, 0x8061280) = -1 EINVAL (Invalid argument) |
| 02:08.36 | tito | write(2, "ZT_SPANCONFIG failed on span 1: "..., 54ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 02:08.36 | Legend | conference doesn't work, redial doesn't work |
| 02:08.38 | tito | ) = 54 |
| 02:08.41 | tito | grrr :( |
| 02:08.44 | Sedorox | hmmm |
| 02:08.46 | Legend | haven't tried hold recenty |
| 02:09.25 | *** join/#asterisk DH-Kelly (none@pollo.cykotix.com) |
| 02:09.28 | DH-Kelly | Hi |
| 02:09.31 | Sedorox | hmmm |
| 02:12.01 | Sedorox | do you like it tho, overall? |
| 02:12.25 | DH-Kelly | We just got our Asterisk machine up and running today/last night, this is pretty neat. |
| 02:14.21 | three55ml | DH-Kelly: Glad to hear it |
| 02:15.10 | Q-At-Home | has anyone had any experience adding * to a meridian (nortel) box without using a PRI? |
| 02:15.53 | Q-At-Home | i.e use an fxs/fxo combo |
| 02:16.59 | DH-Kelly | Right now we have a pretty basic setup, still need to add minor details like voicemail, etc, right now calls ring for 20sec and drop :) |
| 02:17.25 | Silik0n | Q-At-Home are you using asterssk in front of it? |
| 02:17.26 | *** join/#asterisk boch (sdf@host200.200.61.129.ifxnw.com.ar) |
| 02:17.48 | DH-Kelly | I'm trying to get a better sip server setup though, BroadVoice has once that is local to us, but we need to setup a custom forward (hostname to ip) mapping to use it |
| 02:18.40 | *** join/#asterisk dalabera (~dalabera@adsl-9-151-98.mia.bellsouth.net) |
| 02:18.40 | *** join/#asterisk file[laptop] (~file@mctn1-6079.nb.aliant.net) |
| 02:19.07 | DH-Kelly | is there any way to do that short of running a local name server |
| 02:19.21 | three55ml | DH-Kelly: There are a ton of them, take a look at http://www.voip-info.org under "Providers" |
| 02:20.37 | DH-Kelly | Well, we're pretty happy with broadvoice, I jut need to get that host->ip mapping going |
| 02:20.42 | Q-At-Home | Silik0n: heres what I have, 2 locations with nortel boxes that want to forward calls to the other location during lunch, using voip |
| 02:20.57 | Q-At-Home | maximum 3 channels up at any given time |
| 02:22.13 | three55ml | DH-Kelly: You could always just put the entry in your hosts file |
| 02:22.29 | three55ml | DH-Kelly: /etc/hosts - would bypass you having to run a local nameserver. |
| 02:22.30 | DH-Kelly | I tried setting it in /etc/hosts, it seems thats only used for ip->hostname mapping |
| 02:22.52 | three55ml | You can just use the IP in Asterisk, what's wrong with that? |
| 02:24.03 | DH-Kelly | hrm, I had tried setting the host= line differently and I got 404 errors, I will try the ip :) |
| 02:24.17 | Q-At-Home | so, at lunch location1 forwards the phones to ext 5555 for example, which terminates on the remote * |
| 02:26.02 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 02:26.29 | DH-Kelly | -- Got SIP response 404 "Not Found" back from 147.135.8.128 |
| 02:26.53 | stormfr | how is it possible to tune sip communication for high latency network (satellite) with * (voice frame tx) ? Using G.729 or gsm but it's not enough. |
| 02:27.07 | tzafrir_laptop | I've just spent a couple of hours trying to make vmail.cgi strict, so I can feel safer changing it. Seems hopeless. |
| 02:27.22 | PTG123 | stormfr: i don't think asterisk can be tuned for that..however voip in general can |
| 02:27.31 | tzanger | stormfr: codec doesn't fix latency |
| 02:27.36 | tzanger | nothing fixes latency |
| 02:27.50 | stormfr | yes but we can maybe setup more voice frame per packet like some phone can do ? |
| 02:27.52 | tzafrir_laptop | That script badly needs a rewrite. |
| 02:27.57 | PTG123 | tzanger: there is actually a satellite inet provider that has their own voip, and its suppose to work awesome |
| 02:27.58 | stormfr | also new jitter can help too |
| 02:28.14 | PTG123 | tzanger however their jitterbuffer is designed to cache packets, to make it work well or some shit |
| 02:28.42 | tzanger | PTG123: there's no getting around the 600ms RTT, doesn't matter HOW many frames are in the packet or what the codec is |
| 02:28.55 | tzanger | PTG123: if it's geosync it's 600ms RTT |
| 02:28.59 | *** join/#asterisk iq (~iq@70-59-167-207.omah.qwest.net) |
| 02:29.03 | tzanger | PTG123: if it's LEO then yeah you can get amazing latency |
| 02:29.05 | PTG123 | tzanger: sure your convo will just be 600ms lagged, the killer is when you get packets FASTER then 600ms |
| 02:29.19 | PTG123 | which is why you need to cache packets, and make sure you always have a set latency |
| 02:29.21 | PTG123 | like 1sec.. |
| 02:30.02 | stormfr | sip to sip call seems ok, but sip to pstn are not well working, i was thinking maybe it's due of the transcoding ? any idea ? |
| 02:30.14 | tzanger | PTG123: how do you get packets faster than they're being sent? |
| 02:30.28 | DH-Kelly | gravity |
| 02:30.32 | PTG123 | tzanger: the problem with sat is the latency isn't consistant.. so packets speed up and slow down |
| 02:30.37 | want561or772did | wormhole |
| 02:30.46 | PTG123 | so the key is to make them consistant |
| 02:30.58 | tzanger | PTG123: that's what a jitter buffer is for |
| 02:31.07 | PTG123 | yes basically |
| 02:31.51 | mrproper_ | im getting a segmentation fault after starting asterisk -vvvvvvvc using asterisk-stable-1.0.7 |
| 02:32.18 | want561or772did | you can't begin decoding a packet until the whole thing is received, either, so shrinking packet size marginally improves latency |
| 02:32.18 | PTG123 | gdb it mrproper |
| 02:32.28 | mrproper_ | gdb? |
| 02:32.33 | JunK-Y | mrproper_: what's the latest line of ur CLI? www.pastebin.ca |
| 02:32.34 | tzafrir_laptop | mrproper_, after what exacly? |
| 02:32.42 | stormfr | PTG123 / tzanger : is there a way to modify number of voice frame with asterisk without recompile it ? (seems not) |
| 02:33.06 | PTG123 | no idea storm, i don't know much about the jitter buffer |
| 02:33.32 | stormfr | seems the new jitter only work with codec that's also been modified for use it |
| 02:33.41 | mrproper_ | JunK-Y: http://pastebin.ca/9454 |
| 02:34.42 | JunK-Y | mrproper_: start it with asterisk -vvvvvvvcg now |
| 02:34.48 | JunK-Y | it will gave ya a core dumpo |
| 02:34.50 | JunK-Y | dump |
| 02:34.58 | JunK-Y | we need the backtrace of that core. |
| 02:35.46 | mrproper_ | JunK-Y: how do i pull that info for you |
| 02:36.01 | Q-At-Home | yay! phantom ringing is solved in cvs... |
| 02:36.13 | JunK-Y | ya've the core file now? |
| 02:36.25 | Nugget | Q-At-Home: what branch? |
| 02:36.28 | mrproper_ | where do you get that from? |
| 02:36.49 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 02:36.49 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:36.54 | mrproper_ | same directory i ran -vvvvvcg in ? |
| 02:37.02 | Q-At-Home | no idea what branch... but I assume unstable |
| 02:37.06 | JunK-Y | after starting asterisk with -g option, when asterisk crashed, it will generate a file in the same dire where ya ran asterisk. |
| 02:37.07 | JunK-Y | yes |
| 02:37.18 | Q-At-Home | hrmm... who has the best rates on spa 3000's |
| 02:37.19 | mrproper_ | JunK-Y: its 17meg |
| 02:37.34 | Q-At-Home | everyone in the usa wants an arm and a leg to ship to canada |
| 02:37.34 | mrproper_ | 17M Apr 13 11:25 core.19206 |
| 02:37.56 | Q-At-Home | cvs-head |
| 02:38.08 | JunK-Y | when getting the core dump do: |
| 02:38.08 | JunK-Y | gdb -se "asterisk" -c <core_xyz> |
| 02:38.08 | JunK-Y | in gdb: do a bt full |
| 02:38.37 | JunK-Y | we need that gdb output. |
| 02:38.47 | mrproper_ | ok ill paste to a pastebin |
| 02:39.17 | JunK-Y | thanks. |
| 02:40.11 | mrproper_ | JunK-Y: http://pastebin.ca/9456 |
| 02:40.21 | tzafrir_laptop | mrproper_, from the looks of it, either the features module or the next one (more probable) is the one causing the crash. |
| 02:40.52 | JunK-Y | bt gives ya similar output? |
| 02:41.58 | jterrero | can someone help me out? I have "exten => s,1,Playback(/var/etc/soundfile)" in the context mainmenu... i also have "exten => 301,1,Goto(mainmenu)" but when i dial ext 301 i cannot hear my sounds.. I know the sound file is good because I have "exten => 303,1,Playback(/var/etc/soundfile" and when i go to 303 i can hear my sound |
| 02:42.03 | jterrero | can someone helpme out ? |
| 02:42.47 | tzafrir_laptop | <PROTECTED> |
| 02:43.33 | fugitivo | that's weird |
| 02:43.40 | jterrero | thats just an example, too lazy to type /var/lib/asterisk/sounds/ |
| 02:43.53 | Sedorox | just did... |
| 02:44.00 | jterrero | ;) |
| 02:44.03 | Sedorox | :p |
| 02:44.14 | tzafrir_laptop | Goto(mainmenu) sends you to a priority called "mainmenu". Try Goto(mainmenu,s,1) |
| 02:44.50 | tzafrir_laptop | jterrero, for a sound in the default sound path you don't have to write the full path. |
| 02:45.00 | mrproper_ | Junk-Y: any idea from that gdb whats going on? |
| 02:45.06 | jterrero | tzafrir_laptop: thx for the tip, really helpful |
| 02:45.20 | jterrero | tzafrir_laptop: what if i have a directory inside the default path |
| 02:45.24 | jterrero | can i still just use the sound name |
| 02:45.25 | jterrero | ? |
| 02:45.29 | tzafrir_laptop | mrproper_, a. you didn't give there a backtrace. |
| 02:45.46 | tzafrir_laptop | jterrero, I figure you could give a path relative to it. |
| 02:46.03 | mrproper_ | tzafrir_laptop: http://pastebin.ca/9456 |
| 02:46.18 | tzafrir_laptop | jterrero, don't forget that if you give a full path, the whole language-based sounds won't work |
| 02:46.57 | jterrero | lost me, not sure what you mean by language based sounds |
| 02:46.58 | jterrero | but ok |
| 02:47.14 | tzafrir_laptop | mrproper_, I don't see any backtrace there. Did you run bt? is 'strncpy' the only function in it? |
| 02:47.29 | tzafrir_laptop | Anyway, it traces the issue to chan_features. |
| 02:47.45 | JunK-Y | u should try: rm -f /usr/lib/asterisk/modules/; then make clean; make install |
| 02:48.09 | tzafrir_laptop | rm -f /usr/lib/asterisk/modules/* ; actually |
| 02:48.34 | tzafrir_laptop | bash: actually: command not found |
| 02:49.09 | mrproper_ | JunK-Y: http://pastebin.ca/9458 <----has bt info |
| 02:49.10 | JunK-Y | ya rm -rf :) |
| 02:49.36 | jterrero | do i have to specify in any of my asterisk config files to accept DTMF? i have sip phone that doesnt send dtmf (i can call a place with an ivr and when i push a button nothing happens) same is true when i dial into my pbx, it will not accept anything |
| 02:50.03 | JunK-Y | its in chan_features |
| 02:50.29 | jterrero | me? |
| 02:50.50 | dca | anyone use realtime? |
| 02:51.12 | tzafrir_laptop | mrproper_, are you sure that chan_features.so was installed in the current install and is not a left-over? |
| 02:51.18 | *** join/#asterisk adjacent (~scott@64.203.220.105) |
| 02:51.32 | boch | anyone here took the cvoice training? |
| 02:51.33 | want561or772did | iax media stream transfers don't work at all if one side can initiate to the other but not vice versa. which seems unnecessary |
| 02:51.39 | JunK-Y | tzafrir: that why i told him to rm all his modules, to make clean; make install |
| 02:51.45 | tzafrir_laptop | Anyway, another "quick fix" is to try to explicitly unload it in modules.conf and move on to see what would be the next problem |
| 02:52.09 | Q-At-Home | who other than xten(xlite/pro) makes a sip "phone" for pocketpc? |
| 02:52.28 | mrproper_ | tzafrir: it could be, i was running the cvs version with issues, so i ran a make clean all, and then started a new install of asterisk stable |
| 02:52.48 | mrproper_ | tzafrir: should i have manually removed all of /usr/lib/asterisk/modules? |
| 02:53.22 | mrproper_ | lol |
| 02:53.52 | mrproper_ | ill remove all of /usr/lib/asterisk* then make clean and recompile, ill be back if i have issues, thanks for the help guys |
| 02:54.13 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 02:54.34 | tzafrir_laptop | mrproper_, also make sure all of your "toolchain" (libpri etc.) is from stable when you build * |
| 02:55.04 | mrproper_ | tzafrir_laptop: how can i clean up the old libpri/zaptel etc? |
| 02:55.22 | JunK-Y | make clean; |
| 02:55.30 | mrproper_ | np |
| 02:56.02 | *** join/#asterisk teamjet (~teamjet@lfc.tor.istop.com) |
| 02:56.33 | teamjet | hi, if you have some understanding of JTAPI/TAPI, can you use it to write modules for asterisk? |
| 02:56.46 | teamjet | what does asterisk developers do to develop add ons? |
| 02:57.00 | teamjet | thx for any insight, mucho appreciation |
| 02:57.11 | JunK-Y | write ur own app |
| 02:57.18 | JunK-Y | based on app_skel.c if ya want. |
| 02:58.58 | Delmar | im not having much luck with this rxfax thingie. my fax extension is working.... i created an extension and when i dial it from a SIP phone i hear fax noise etc... and Asterisk consol says.. redirecting to fax extension... then it just does Timeout, but no rule 't' in context 'incomingFXO' and hangs up. |
| 02:59.23 | Hogie | does anybody know the best way to have 2 servers doing parking? I have 1 server at 2 offices each, and I want to be able to park calls and pick them up on the other system... |
| 02:59.24 | teamjet | is developing add-ons to asterisk-friendly hardware a popular topic? |
| 02:59.40 | JunK-Y | teamjet: depends of each ones. |
| 02:59.51 | *** join/#asterisk mog_home (~mog_home@146.229.181.169) |
| 03:00.46 | Hogie | just set one up to like 800 and one to 700? |
| 03:01.35 | dalabera | Hogie , would be better if you send an email to asterisk-users, and explain with every detail and there someone will answer you for sure.... |
| 03:01.48 | *** join/#asterisk salviadud (~dude@201.133.209.245) |
| 03:02.15 | salviadud | i've just bought a sipura 3000 |
| 03:02.46 | salviadud | where can i get some great faqs so i can set up a pbx, you think i can pull it off with the asterisk handbook? |
| 03:03.14 | teamjet | JunK-Y: lol what you are saying is very clear, but it sounds very confusing to mee |
| 03:03.40 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 03:04.00 | pigpen | I just got a new pri installed. I have added "exten => _9NXXXXXX,1,Dial(Zap/3/${EXTEN:1})" to my outgoing.... |
| 03:04.09 | want561or772did | pigpen: can i use it? |
| 03:04.33 | pigpen | well I gotta make it work first... |
| 03:04.37 | pigpen | geesh.. |
| 03:04.42 | want561or772did | oh. carry on |
| 03:05.04 | pigpen | ok..anyway...when I dial I see: == Spawn extension (ccnbi-ext, 95683553, 1) exited non-zero on 'SIP/mark-870a' |
| 03:05.05 | pigpen | <PROTECTED> |
| 03:05.05 | pigpen | <PROTECTED> |
| 03:05.05 | pigpen | <PROTECTED> |
| 03:05.06 | Delmar | anyone know why im getting Timeout, but no rule 't' in context 'incomingFXO' when * tries to redirect to the Fax extension? |
| 03:05.11 | pigpen | with the 568 number being local... |
| 03:05.18 | pigpen | I get only a busy tone... |
| 03:05.30 | pigpen | does it sound like the telco hasn't released the line yet? |
| 03:05.48 | Hogie | dalabera: I would do that, but I dont do email, though I read the lists on the archive server |
| 03:05.49 | Delmar | pigpen, its dialing too fast? |
| 03:05.59 | pigpen | Delmar: huh? |
| 03:06.20 | Delmar | oh its a Pri card.. tis all digital then |
| 03:06.29 | pigpen | yeah...digium... |
| 03:06.42 | PTG123 | anyone ever sold dvds on ebay? |
| 03:06.42 | pigpen | so, think the telco hasn't finished provisioning? |
| 03:06.44 | Delmar | no as in Primary Rate ISDN channel = digital. |
| 03:06.54 | pigpen | yeah...pri. |
| 03:07.04 | jterrero | do i have to specify in any of my asterisk config files to accept DTMF? i have sip phone that doesnt send dtmf (i can call a place with an ivr and when i push a button nothing happens) same is true when i dial into my pbx, it will not accept anything |
| 03:07.16 | Delmar | did you provision the channel in zapata? |
| 03:07.20 | pigpen | yep. |
| 03:07.35 | Delmar | ring the telco and confirm its all running their end. |
| 03:07.51 | Delmar | easier to check their end than it is to muck about with * .. pointlessly.. |
| 03:08.12 | pigpen | It is a very new circuit....I just haven't worked with * much to trouble shoot... |
| 03:08.13 | pigpen | thanks! |
| 03:08.39 | Delmar | yeah i reckon if they tell you their end is sweet.. then u can troubleshoot your end. |
| 03:08.56 | *** join/#asterisk LeoB (~chatzilla@c-66-31-41-1.hsd1.ma.comcast.net) |
| 03:09.07 | Delmar | troubleshoot by starting at the easiest point. you will shoot yourself if u end up spending hours trying to fix * when it aint broke. |
| 03:09.39 | Delmar | i dunno wht the hell is wrong with my rxfax thingie. |
| 03:09.50 | pigpen | k...so the same tactics as a cisco...start with the telco... |
| 03:09.55 | pigpen | thanks. |
| 03:09.58 | Delmar | yep. haha |
| 03:10.11 | LeoB | [novice] could not register xlite with asterisk. please help |
| 03:12.44 | jterrero | is there a command at the CLI to debug DTMF signals"? |
| 03:13.12 | Qwell | PTG123: ping |
| 03:13.17 | *** part/#asterisk teamjet (~teamjet@lfc.tor.istop.com) |
| 03:13.57 | *** join/#asterisk gongoputch (~kseel@pcp01486721pcs.limstn01.de.comcast.net) |
| 03:14.08 | LeoB | would anyone help me configure xlite so that it works with asterisk? |
| 03:14.39 | Nugget | only if you can demonstrate that you've already tried all of the readily available walk-throughs and howtos and other good documentation online. |
| 03:14.52 | Nugget | it' |
| 03:14.58 | Nugget | it is not our job to read web pages to you. |
| 03:15.02 | want561or772did | xlite only does SIP doesn't it |
| 03:15.23 | LeoB | how can I demonstrate that? :) I've spent a couple of hours and I couldn't figure it out... :( |
| 03:16.03 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 03:16.08 | LeoB | ... the thing is that sjphone is not working either... |
| 03:16.08 | want561or772did | send me a private message with your IQ |
| 03:16.26 | LeoB | :) |
| 03:17.06 | want561or772did | xlite had a wonky interface so i threw it away. i use diax on windows and iaxcomm in loonix |
| 03:17.11 | *** join/#asterisk outsidefactor (barf@203-206-247-72.dyn.iinet.net.au) |
| 03:18.41 | LeoB | so you think I should switch from sip to IAx? |
| 03:19.18 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 03:19.36 | Mazda-MX5 | Hi all~ |
| 03:20.29 | want561or772did | if you're using a softphone, why not |
| 03:22.40 | LeoB | I thought sip was more popular... |
| 03:26.23 | *** join/#asterisk Tuplink (~dsfsk@68-232-92-239.chvlva.adelphia.net) |
| 03:26.52 | Tuplink | what dose theis meen Apr 12 23:25:16 NOTICE[4343]: chan_iax2.c:5761 socket_read: Rejected connect attempt from 65.39.205.121, request '641726@default' does not exist |
| 03:27.05 | file[laptop] | the extension 641726 does not exist in context default |
| 03:27.14 | want561or772did | it means fwd is trying to connect to that extension in default but it's not there |
| 03:27.25 | PTG123 | anyone have a need for any extreme networks, ciscos catalysts, or hp procurve switches.. gonna list a bunch on ebay, want to get rid of them cheap.. just taking up space :) |
| 03:27.51 | Tuplink | ok...... |
| 03:27.57 | Tuplink | so how do i fix it? |
| 03:28.10 | Tuplink | extentions.conf and add an extention |
| 03:28.15 | want561or772did | create that extension and do something with it yeah |
| 03:28.22 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 03:28.27 | want561or772did | like have it dial your softphone or console |
| 03:29.19 | MrBelvedr | I am trying to slim down my system. I am editing modules.conf. What are the minimum modules that need to be loaded to terminate iax calls? |
| 03:29.32 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 03:30.08 | harryvv | Shaw cable router was down like 2 plus hours could not get on..and in thorey could not make a phone call if my phone service was all voip. |
| 03:30.46 | r0d3nt | not in theory.. you couldn't... |
| 03:31.06 | want561or772did | there should be a way to securely virtualhost asterisk dialplans for DIDs |
| 03:31.38 | denon | I dont think anyone's suggesting you replace all your pots lines with a shitty cable provider .. |
| 03:31.50 | denon | thats like saying you can replace your car with broken rollerblades |
| 03:32.01 | file[laptop] | but broken rollerblades are great! honest! |
| 03:32.04 | denon | voip is, however, a reality with commercial circuits and qos |
| 03:32.10 | denon | hehe file |
| 03:32.19 | NormAst | :( had to reset my asterisk box ... 1.1 Gigs of memory usages... |
| 03:32.24 | harryvv | deon :) belive it or not shaw is more reliable then telus dsl |
| 03:32.25 | NormAst | 6 weeks up time. |
| 03:32.49 | Qwell | NormAst: * was using 1.1gb? |
| 03:32.58 | Hogie | I had 2 year uptime on my speakeasy sdsl circuit, I miss that apartment:( |
| 03:33.00 | denon | harryvv: doesnt matter .. unless its designed to be a commercially used circuit, it probably wont suffice for exclusive and real voice traffic |
| 03:33.11 | harryvv | ohh sure I know. |
| 03:34.05 | DH-Kelly | You know, doing the basics in Asterisk is actually pretty easy once you force yourself to actually sit here and read the manuals :) |
| 03:34.42 | NormAst | Manuals...out of date. |
| 03:34.55 | NormAst | ~docs |
| 03:34.56 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 03:35.01 | NormAst | hee he |
| 03:35.03 | calvinhp | has anyone ever run into the problem where after about 7-9 minutes in a call it will blank out for about 10-15 seconds. It comes back just fine, but no one can hear each other for that time span |
| 03:35.05 | DH-Kelly | for the basic stuff this voip-info.org wiki seems to work |
| 03:35.15 | Hogie | I couldn't believe I got my T1 circuit to work with both our current key system and the new asterisk box at the same time. |
| 03:35.23 | calvinhp | this is with Cisco 7940 phones and Polycom IP300s |
| 03:35.24 | Tuplink | ok... i am able to recive calls from FWD now how do i make calls to it? |
| 03:35.33 | Hogie | no working late to "test" anymore |
| 03:35.33 | Qwell | Tuplink: by reading the rest of the howto. :) |
| 03:35.34 | Hogie | lol |
| 03:35.39 | harryvv | denon: this is specially bad if shaw wants to sell there shaw voip service. |
| 03:35.42 | Tuplink | i did. |
| 03:35.46 | PTG123 | calvinhp: running cvs? |
| 03:35.47 | calvinhp | they are all set to the same codecs too |
| 03:35.52 | calvinhp | 1.0.7 |
| 03:36.09 | *** join/#asterisk iq{tablet} (~iq@70-59-167-207.omah.qwest.net) |
| 03:36.12 | Tuplink | qwell its like 1 printed page i have read it |
| 03:36.21 | denon | harryvv: was their actual cable segment down? or just their routes to the outside? |
| 03:36.33 | denon | if it was just their outside routes, their voip service probably woulda kept working |
| 03:36.39 | calvinhp | PTG123: it happens between the phones so it doesn't really involve the TDM04B I have in the machine it seems |
| 03:37.08 | Kumbang | anyone experienced testing mfcr2 back to back with TE400P using chan_unicall? |
| 03:37.25 | Tuplink | so any one get FWD to work? |
| 03:37.35 | Sedorox | I have it working |
| 03:37.43 | Tuplink | i cant make calls to it |
| 03:37.50 | Tuplink | can recive them tho |
| 03:37.58 | PTG123 | calvinhp: are you reinviting each other? |
| 03:38.02 | Sedorox | both ways are fine for me... |
| 03:38.04 | DH-Kelly | One thing i'm not seeing, how do you make Asterisk (maybe my telephone bank?) not beep busy upon HangUp ? |
| 03:38.08 | Sedorox | using IAX2? |
| 03:38.13 | Tuplink | can you help me set it up? |
| 03:38.19 | Tuplink | yup |
| 03:38.24 | calvinhp | PTG123: the ciscos have reinvite turned off, but the polycom has it turned on |
| 03:38.36 | calvinhp | I'm not sure what that does really |
| 03:38.38 | Sedorox | did you click the box to switch your account over? |
| 03:38.43 | DH-Kelly | should I be running some other "sit and spin" command? |
| 03:38.45 | harryvv | deon, thay said it was not physical plant so it looks like there router went down for 2 plus hours. |
| 03:38.45 | Tuplink | yup |
| 03:38.56 | Sedorox | then just follow their setup on their site.. and it should work |
| 03:39.08 | Tuplink | i did |
| 03:39.20 | PTG123 | calvinhp: sounds lik ean asterisk problem then.. and i don;'t know since i dumped cvs i haven't had that issue |
| 03:39.20 | Sedorox | what do you get on the CLI when a call comes in? |
| 03:39.23 | Sedorox | errrr |
| 03:39.25 | Sedorox | try to make one |
| 03:39.30 | calvinhp | PTG123: this drop off will happen even when talking from a Cisco out using a Zap chanel |
| 03:39.58 | calvinhp | I figured it wasn't Zap since it will happen even when the Zap isn't involved |
| 03:40.13 | Tuplink | nothing |
| 03:40.33 | Sedorox | you should see something on the dialplan.... what happens on the phone? busy tone? |
| 03:40.54 | Tuplink | 404 bussy yes |
| 03:41.01 | PTG123 | don't know man.. could be a firewall issue |
| 03:41.06 | Sedorox | 404 is not found.. check your dial plan |
| 03:41.17 | Sedorox | make sure the context that the phone points to.. has the fwd included in it |
| 03:41.33 | Qwell | Tuplink: Do you have anything in your dialplan for fwd out? |
| 03:41.36 | dizzydiffi | hello |
| 03:41.48 | MrBelvedr | I am trying to slim down my system. I am editing modules.conf. What are the minimum modules that need to be loaded to terminate iax calls? |
| 03:41.54 | dizzydiffi | i need help with h323 and sip configs |
| 03:42.15 | Tuplink | i made a context caled fwd with the 393's in it and included it in local |
| 03:42.38 | Qwell | Tuplink: paste the relevant part to pastebin.ca (minus passwords...but, you shouldn't have a password in your dial line anyhow) |
| 03:43.18 | *** join/#asterisk alegh (~ag10@OL217-17.fibertel.com.ar) |
| 03:43.21 | Tuplink | pastebin.ca? |
| 03:43.27 | Qwell | http:// |
| 03:43.40 | MrBelvedr | http://pastebin.ca |
| 03:44.33 | alegh | Hi, Does anybody knows if there is any application to listen the from an extension the recorded files with the Monitor command? |
| 03:45.04 | JunK-Y | alegh: record ur monitor file, then playback it? |
| 03:45.14 | yxa | is there a reason to get a specific E1 or T1 card when a E1/T1 is the price? |
| 03:45.16 | Tuplink | 9640 |
| 03:45.23 | yxa | *same price |
| 03:45.47 | Sedorox | Tuplink: the phone is SIP? |
| 03:45.56 | Tuplink | yes |
| 03:45.58 | Qwell | Tuplink: link |
| 03:46.08 | *** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com) |
| 03:46.11 | Qwell | 9460? |
| 03:46.20 | Sedorox | Qwell: yes |
| 03:46.20 | Tuplink | yea |
| 03:46.23 | robl^ | 6234 |
| 03:46.24 | |Vulture| | can anyone send me a testfax? |
| 03:46.25 | Qwell | Tuplink: part of it is missing |
| 03:46.27 | Qwell | }) |
| 03:46.39 | Qwell | and, you really shouldn't have user/pass in the dial line |
| 03:46.42 | alegh | JunK-Y: what I want is to playback the files and listen it from an extension |
| 03:46.43 | Sedorox | Tuplink: what do you have in sip.conf for context= for either [general] or for the phone itself? |
| 03:46.50 | Tuplink | 3},60,r) |
| 03:47.23 | JunK-Y | alegh: then just exten => 1234,1,Playback(monitor_output); to play ur file, no? |
| 03:47.37 | Sedorox | .... |
| 03:47.41 | Tuplink | [20001] |
| 03:47.41 | Tuplink | type=friend |
| 03:47.41 | Tuplink | username=20001 |
| 03:47.41 | Tuplink | host=dynamic |
| 03:47.48 | Qwell | no context? |
| 03:47.56 | Sedorox | in general.. what is context= |
| 03:48.22 | Tuplink | default |
| 03:48.27 | Sedorox | ok |
| 03:48.35 | Sedorox | under the default context... include fwd |
| 03:48.39 | Sedorox | reload.. and it'll work |
| 03:48.55 | Tuplink | in sip? |
| 03:48.57 | alegh | JunK-Y: something like that but I have to first browse the folder and find the names of the files to playback |
| 03:49.01 | Qwell | in extensions.conf |
| 03:49.08 | JunK-Y | alegh: just save it. |
| 03:49.17 | Sedorox | what Qwell said |
| 03:49.36 | alegh | JunK-Y: maybe a menu for listen again, go back and forward, etc |
| 03:49.48 | JunK-Y | then do an agi. |
| 03:50.16 | Tuplink | can i just include local |
| 03:50.35 | want561or772did | it's unfortunate that chan_alsa and chan_oss hold open the sound card. otherwise i'd leave em running |
| 03:50.40 | alegh | I just looking if there is some app out there to used. Thanks. |
| 03:50.42 | *** part/#asterisk DH-Kelly (none@pollo.cykotix.com) |
| 03:50.57 | Sedorox | no |
| 03:51.07 | JunK-Y | u can do it easily with an agi. |
| 03:51.08 | Sedorox | unless you change the context in sip.conf to local |
| 03:51.09 | Qwell | recursive include, heh |
| 03:51.10 | Tuplink | kool that worked |
| 03:51.20 | Tuplink | thank you qwell |
| 03:51.26 | JunK-Y | i dont work so much with dialplan, im almost directly in agi. |
| 03:53.25 | niZon | Can voicemail be configured to execute an external program/script when someone leaves a message? |
| 03:54.24 | Tuplink | in IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:3} after the / is the actual # that is diald rite |
| 03:54.24 | Sedorox | yes |
| 03:54.24 | Sedorox | the :3 tells it to remove the first three digits |
| 03:54.24 | Tuplink | kool |
| 03:54.24 | Sedorox | the 393 |
| 03:54.25 | Sedorox | so then its the number... |
| 03:54.34 | niZon | this looks interesting: http://www.laser.com/dante/ |
| 03:54.38 | Sedorox | so if you did :4.. it would remove 393 and then whatever the next was |
| 03:56.34 | Tuplink | kool |
| 04:00.17 | want561or772did | can someone call 638271 on FWD please, then choose option 1 |
| 04:01.21 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
| 04:01.29 | Tuplink | i called it... no mic here ;) |
| 04:01.30 | want561or772did | so my console rings really loud. thank you |
| 04:01.38 | PTG123 | holy crap i just had the problem with audio disappearing for 15 seconds |
| 04:02.07 | PTG123 | on stable 1.0.7 who was just saying something about that |
| 04:02.48 | Sedorox | hmm |
| 04:03.05 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
| 04:03.42 | jterrero | can someone help me out? I am trying to have one of my incoming DIDs go to a mainmenu context. when i call I cannot hear nor can i send DTMF, or receive DTMF |
| 04:03.45 | jterrero | http://pastebin.ca/9461 |
| 04:04.54 | TomL | ~seen ManxPower |
| 04:04.57 | jbot | manxpower is currently on #asterisk (48m 54s) |
| 04:05.02 | TomL | ManxPower? |
| 04:05.29 | niZon | bah wtf |
| 04:05.34 | niZon | my CD rom is giving me IO errors |
| 04:10.33 | DaLion | yo lall |
| 04:10.56 | ManxPower | I don't suppose anyone knows of a way to get Asterisk to write the cdr csv files as a specific user without running asterisk as that user? |
| 04:11.14 | DaLion | hmm |
| 04:11.15 | DaLion | nope |
| 04:11.25 | DaLion | why not just use same group |
| 04:12.17 | ManxPower | I guess I could do that. |
| 04:12.50 | ManxPower | I have a cgi script that runs as, oddly enough, user "apache", group "apache", and I want the scrip to access the Asterisk CDR logs |
| 04:13.06 | Hogie | su! |
| 04:13.07 | Hogie | :P |
| 04:15.02 | Mazda-MX5 | in SIP , why the some client must set dtmfmode=inband ? |
| 04:15.17 | Mazda-MX5 | why not is rfc2833 |
| 04:16.18 | three55ml | ManxPower: Does Asterisk recreate the file, or could you try putting them into a group together? |
| 04:16.29 | three55ml | ManxPower: Or you could always use cdr_mysql, pgsql, or similar |
| 04:16.36 | harryvv | mazda, the wiki talks about it |
| 04:16.38 | ManxPower | Mazda-MX5, no client must set that unless the other side sets it. |
| 04:17.32 | ManxPower | three55ml, I do not the added complexity of a real database for this application. Asterisk no only recreates the file, it will change the ownership at random times back to what it thinks it's supposed to be at random times. |
| 04:17.50 | harryvv | wierd |
| 04:18.01 | *** join/#asterisk tylorflys (~tylorflys@ip68-104-178-155.ph.ph.cox.net) |
| 04:18.10 | odie_flocon_ | it's muted me |
| 04:18.25 | odie_flocon_ | BRC... |
| 04:18.42 | three55ml | ManxPower: Hehe, well that sucks :) (The changing of the permissions) |
| 04:18.44 | odie_flocon_ | brc_ I am that IAX user. |
| 04:18.49 | brc_ | ah |
| 04:19.03 | odie_flocon_ | It says i'm muted when I enter? |
| 04:19.03 | brc_ | *1 dude |
| 04:19.08 | three55ml | ManxPower: Horrible way of doing it, but you could always use sudo and make a copy of it or similar. |
| 04:19.08 | odie_flocon_ | how do I fix that? |
| 04:19.45 | *** part/#asterisk gongoputch (~kseel@pcp01486721pcs.limstn01.de.comcast.net) |
| 04:23.01 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 04:23.30 | ManxPower | odie_flocon_, If you read the docs for meetme you would know the answer to that. |
| 04:23.41 | ManxPower | "show application meetme" in the Asterisk CLI. |
| 04:24.02 | odie_flocon_ | ok thanks ManxPower |
| 04:24.14 | `Sauron | Mmmm. |
| 04:24.18 | `Sauron | Yummy powerbook. |
| 04:24.22 | ManxPower | Or RTFW |
| 04:24.50 | sivana | W = wiki? |
| 04:25.22 | ManxPower | sivana, yes |
| 04:25.26 | sivana | heh |
| 04:28.06 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 04:29.28 | elric | hey any news on NVMachineDetect()? theres nothing on the wiki for it |
| 04:30.35 | JunK-Y | NV means? |
| 04:30.57 | Corydon76-home | Nevada |
| 04:31.05 | elric | its the abbreviation for the company that coded it. |
| 04:31.13 | elric | its answering machine detection |
| 04:31.27 | jterrero | what does an ! mean in front of a context |
| 04:31.34 | jterrero | [!context] |
| 04:31.40 | *** join/#asterisk linsys (~non@67.42.246.62) |
| 04:31.57 | three55ml | I'm playing with NVFaxDetect right now without much lick |
| 04:31.59 | three55ml | luck |
| 04:32.05 | elric | ah ok |
| 04:32.08 | linsys | Can anyone tell me what app_directory.so does? It looks like it has something to do with vmail? |
| 04:32.20 | elric | i dont like the way answering machine detection happens now |
| 04:32.29 | elric | waiting a few seconds |
| 04:32.34 | elric | :| |
| 04:34.22 | elric | has anyone else tried answering machine detection and succeeded? |
| 04:35.19 | linsys | the reason I ask is because when I try and load asterisk I get the following error message "app_directory.so: symbol strcasestr: referenced symbol not found"; when I set a noload in the modules.conf asterisk loads fine.. |
| 04:35.27 | linsys | just wanted to be sure what I was turning off.. |
| 04:37.28 | want561or772did | is there a way to make asterisk like a traditional old answering machine where i hear messages as they are left |
| 04:37.43 | ManxPower | BTW, if I was to give a talk at a conference which do you think would be better: 1) Introduction to QoS or 2) discussion of a 60 phone Asterisk deployment |
| 04:38.08 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 04:38.17 | sivana | hrmm... |
| 04:38.19 | zoa | second one |
| 04:38.23 | sivana | both are good |
| 04:38.38 | sivana | I'd like to know QoS |
| 04:38.40 | sivana | :) |
| 04:38.47 | sivana | how intro? |
| 04:39.04 | sivana | workable implementation of QoS or more overview type thing? |
| 04:39.11 | linsys | it probably depends on the audience |
| 04:39.32 | sivana | second one is more broad |
| 04:40.10 | sivana | want561or772did: yes, the voicemail plays back oldest first I think |
| 04:40.43 | ManxPower | sivana, Over of issues and problems, as well as some practical examples. |
| 04:41.04 | ManxPower | For example: QoS and Frame Relay issues. |
| 04:41.25 | sivana | the second one would be more broad.. and the QoS more specialized for seasoned * users |
| 04:42.06 | sivana | if you're charging a cover, then go with the second.. hehe |
| 04:43.05 | ManxPower | sivana, it would be at a conference. |
| 04:43.09 | sivana | :) |
| 04:43.17 | ManxPower | I could do both. |
| 04:43.38 | sivana | I think both topics are good and would be useful |
| 04:43.46 | ManxPower | I'll have to look at it. |
| 04:44.38 | *** join/#asterisk syle (~blah@wnpgmb02dc1-156-248.dynamic.mts.net) |
| 04:44.54 | drumkilla | yeah, depends on the audience ... but I would vote to attend the QoS talk :) |
| 04:52.15 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 04:54.13 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 04:54.39 | want561or772did | ah. chanspy all. |
| 04:58.12 | drumkilla | how's it going want561or772did ? |
| 05:00.34 | want561or772did | i've turned asterisk into a jukebox/door buzzer/sex line/panopticon |
| 05:00.53 | want561or772did | the only thing is chan_alsa doesn't seem to close the audio device on unload |
| 05:02.41 | drumkilla | what?! ha ... |
| 05:03.18 | want561or772did | also i've tried to make media stream transfers work against two sessions coming from the same IP but behind NAT |
| 05:03.31 | want561or772did | by telling that ip to try txing to localhost |
| 05:03.50 | want561or772did | i'm waiting for a real phone line to test this with |
| 05:12.17 | want561or772did | drumkilla do you know of a way to force an answered channel to transfer to a specific context/extension |
| 05:13.35 | ManxPower | want561or772did, Um, Dial( |
| 05:13.42 | ManxPower | Or Goto |
| 05:14.12 | brc_ | mod_ssl is a PITA |
| 05:16.33 | want561or772did | ManxPower i mean for an already answered call that i want to transfer to my softphone on the spur of a moment, for instance |
| 05:16.52 | *** join/#asterisk TheEmperor (~TheEmpero@203.121.47.165) |
| 05:16.55 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 05:17.28 | ManxPower | want561or772did, Then you use the Transfer feature of the device you are using. |
| 05:18.19 | ManxPower | For Zap or ATAs, you use FLASH, for other VoIP devices you use the TRANSFER key/button of the device. For devices too stupid to have a TRANSFER key, you use t and T options to Dial and use # |
| 05:18.44 | Qwell | All you have to do is hookswitch, to transfer? |
| 05:18.52 | want561or772did | whose transfer button? i'm talking about a caller whose buttons i can't press |
| 05:18.59 | ManxPower | Qwell, on ATAs and Zap, yes. |
| 05:19.00 | Qwell | no t or T in the dial line? |
| 05:19.07 | Qwell | for Zap I mean, yeah |
| 05:19.07 | ManxPower | Qwell, Correct. |
| 05:19.13 | salviadud | im a complete newbie, does anyone here own a sipura 3000? |
| 05:19.17 | ManxPower | T and t should be only use a a last resort. |
| 05:19.17 | Qwell | wow...flash, dial, flash? |
| 05:19.18 | want561or772did | he's leaving me a message in voicemail for instance. but i'd like to yank him out of voicemail and transfer him to my softphone |
| 05:19.28 | ManxPower | want561or772did, You can't do that. |
| 05:19.55 | Qwell | hmm, gonna have to try that out |
| 05:21.03 | want561or772did | well there's this add queue member command on the console.. |
| 05:21.05 | Qwell | wow... |
| 05:21.19 | Qwell | I feel dumb for not knowing that. :P |
| 05:21.30 | salviadud | don't feel dumb, im the newbie here |
| 05:21.37 | *** part/#asterisk DrJolo (~chatzilla@217.153.194.10) |
| 05:21.48 | Qwell | ManxPower: sadly, I've even RTFW |
| 05:22.08 | TheEmperor | ManxPower:in the extensions file,do you need to put t in to be able to transfer a call? |
| 05:22.26 | salviadud | well here's a question for ya guys. can i run asterisk under a 2.6.x kernel? |
| 05:22.36 | Qwell | TheEmperor: <ManxPower> For Zap or ATAs, you use FLASH, for other VoIP devices you use the TRANSFER key/button of the device. For devices too stupid to have a TRANSFER key, you use t and T options to Dial and use # |
| 05:23.14 | TheEmperor | Qwell:on my ip phone, there is no transfer key, just flash key which doesn't work |
| 05:23.28 | Qwell | TheEmperor: <ManxPower> T and t should be only use a a last resort. |
| 05:24.02 | syle | i been wondering same thing salviadud |
| 05:24.16 | salviadud | im gonna read the faqs, and i'll be back!!! |
| 05:24.32 | Silik0n | anyone heard of a problem withSPA-841s where if youhave a call onbutton1 and a call comesin (on button 2) you try to use the builtin softkey to transfer the call andit bredges to thecalls together |
| 05:24.32 | salviadud | honestly, i want to learn, but i don't know where to start... |
| 05:24.44 | syle | people bitch at me to move to fedora core 3 and to not stay with old school 2.4.x kernels yet i read they only support 2.4.x kernel somewhere hehee |
| 05:25.21 | salviadud | ohhhh, i got slackware 10.1 on kernel 2.6.7.11 |
| 05:25.29 | salviadud | so you're saying, it wont work? |
| 05:25.33 | Silik0n | 2.6.x kernels work fine with asterisk |
| 05:25.38 | three55ml | Yep |
| 05:25.40 | Silik0n | FC3 sux tho |
| 05:25.52 | salviadud | how about slackware, whats your opinion on that distro? |
| 05:25.59 | Silik0n | imho anyway |
| 05:26.13 | three55ml | In my opinion, for a purely server install Debian is your best choice |
| 05:26.18 | syle | yeah well try installing new sata drives on old installs and tell me how much it sucks then silik0n :) |
| 05:26.36 | Silik0n | well I work at a company that does pretty mcuh nothing but * and we use either RHEL3 (for support contracts that want it) or ricer linux errr gentoo |
| 05:26.48 | Silik0n | syle: do it all the time w/ gentoo heh |
| 05:26.55 | ManxPower | Qwell, Thank you, Qwell. It's nice that you pasted that when some asked the question THIRTY SECONDS after I answered it. |
| 05:27.03 | ManxPower | some = someone |
| 05:27.25 | syle | i got use to redhat but i refused to ever pay for linux so fedora seemed viable solution |
| 05:27.30 | ManxPower | salviadud, All distros work. I happen to prefer Mandrake, but that's not an Asterisk thing that's a ME thing. |
| 05:27.38 | Silik0n | syle: then use WBEL or CentOS |
| 05:28.00 | Silik0n | both are RHELs w/ the redhat logos and stuff stripped out |
| 05:28.06 | Qwell | ManxPower: without redundancy, we wouldn't have anything to live for. :p |
| 05:28.10 | salviadud | yeah, linux is so cool, i like to play with nmap, hehe |
| 05:28.35 | syle | there are so many different version of linux you never know what to pick anymore |
| 05:28.45 | Silik0n | nmapping the wrong boxes will make your box disappear from the internet |
| 05:29.01 | salviadud | are you sure silikon? |
| 05:29.10 | syle | always some new idiot comming out with installation scripts and his own kernel and creating a new linux distro somewhere |
| 05:29.12 | Silik0n | salviadud yes I am |
| 05:29.17 | salviadud | actually, one day |
| 05:29.21 | salviadud | i nmapped this site |
| 05:29.31 | salviadud | and they had the telnet open |
| 05:29.37 | salviadud | i tried to enter as root |
| 05:29.44 | Silik0n | salviadud: i know several people that fire off hostile responves to nmap scans |
| 05:29.58 | salviadud | y always use -sS |
| 05:30.02 | salviadud | its kinda sneaky |
| 05:30.17 | Silik0n | ahhhh -sS dont mask you |
| 05:30.32 | salviadud | still, i live in mexico, its a different country |
| 05:30.41 | Silik0n | maybe 6 or 7 years ago that was ok but not today |
| 05:30.44 | three55ml | Same Internet :) |
| 05:30.44 | ManxPower | I have 7 Asterisk servers running on Mandrake 9.2 |
| 05:31.19 | salviadud | ah come on, they're probably gonna see the log file and check out some ip from mexico tried to enter as root. its ridiculous |
| 05:31.52 | salviadud | anyways. i just like to know what OS people are running |
| 05:31.55 | Silik0n | salviadud: actually you try to connect as root to some of my boxes and you get null routed |
| 05:32.12 | *** part/#asterisk tylorflys (~tylorflys@ip68-104-178-155.ph.ph.cox.net) |
| 05:32.16 | salviadud | i don't know what null routed means |
| 05:32.20 | salviadud | what happens? |
| 05:32.23 | Silik0n | salviadud: then nmap some of the MX servers for hotmail.com and get a good laff |
| 05:32.40 | Silik0n | nullroute? that means my boxes disappear from the internet from your perspective |
| 05:32.51 | harryvv | sounds like nuts rolled |
| 05:33.06 | salviadud | how do i set up my firewall to do that? |
| 05:33.21 | Silik0n | ask google |
| 05:33.26 | salviadud | alright |
| 05:33.26 | ManxPower | Qwell, I could use a redundant liver. |
| 05:33.33 | Qwell | ManxPower: couldn't we all... |
| 05:33.55 | ManxPower | Qwell, I could use one more than most people. 8-( |
| 05:34.02 | Qwell | ahh... |
| 05:34.47 | syle | hmm |
| 05:34.50 | syle | fedora core 4 out |
| 05:34.57 | Qwell | syle: oh? |
| 05:34.59 | syle | anyone tested gcc 4.0? |
| 05:35.14 | Qwell | syle: its not out, its in test still |
| 05:35.18 | salviadud | can asterisk run on solaris? |
| 05:35.34 | Qwell | salviadud: don't expect the hardware/drivers to work |
| 05:35.55 | syle | i did have a question relevant to this channel when i came in lol |
| 05:36.50 | salviadud | im just curios, i don't think i'll ever get my hands on a solaris |
| 05:36.51 | syle | oww yeah....what if you wanted to run a VOIP business...hardware i see supported with asterisk has like 4 ports etc, what bigger solutions for many phone lines are there? |
| 05:37.02 | three55ml | syle: T1 interfaces |
| 05:37.07 | three55ml | channel banks |
| 05:37.08 | Qwell | syle: quad T1, and now a DS3 card |
| 05:37.16 | salviadud | i actually work at a call center, we got like 3 channel banks |
| 05:37.17 | Silik0n | T1 cards, quad t1 cards, DS3 card one of these days |
| 05:37.23 | salviadud | but we use freakin windows |
| 05:37.42 | salviadud | its not cool... windows is lame |
| 05:37.51 | linsys | salviadud: Yes |
| 05:37.52 | Silik0n | or you can use hardware like a MAX-TNT or a APX-8000 to get very high density w/ hardware codecs |
| 05:37.57 | linsys | salviadud: Asterisk runs on Solaris.. |
| 05:38.08 | linsys | salviadud: I just set it up on an e4500 |
| 05:38.11 | Qwell | Silik0n: What are those? |
| 05:38.14 | salviadud | it does eh... its very flexible then |
| 05:38.14 | syle | what kind of channel banks are you using? |
| 05:38.38 | salviadud | i haven't seen them yet. the quality guy told me they were german |
| 05:38.47 | salviadud | and started with a letter V or something |
| 05:38.49 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:38.54 | linsys | salviadud: like someone else said the FXO, or FXS, or T1 etc.. cards don't work, but if you have a channel bank or are providing and receiving just SIP or IAX traffic you are fine.. |
| 05:38.58 | salviadud | i don't have access to the server , hehe |
| 05:39.03 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 05:39.16 | ManxPower | I figured out why my TDM400P stoped working ocasionally. |
| 05:39.18 | Silik0n | Qwell: well a MAX-TNT is a box that takes a channelized DS0 and plays modem bank or PSTN->VoIP bridge... an APX8K is the same thing but w/ support for 4 DS3 |
| 05:39.31 | Qwell | oh |
| 05:39.57 | Silik0n | when dialup was all the rage MAXTNTs were the shit |
| 05:40.01 | ManxPower | I had the original verison of the card, the one without the power connector |
| 05:40.34 | salviadud | i want to implement open source on that company |
| 05:40.53 | salviadud | but... its a longshot, i don't know * about asterisk |
| 05:41.23 | Silik0n | so download it and start playing with it |
| 05:41.29 | salviadud | still, im very very positive |
| 05:41.34 | salviadud | yeah, i got hardware |
| 05:41.36 | salviadud | sipura 3000 |
| 05:41.46 | salviadud | you think i could setup like a small office at my home? |
| 05:41.53 | Silik0n | and get a copy of xlite or firefly and go have fun |
| 05:41.57 | Silik0n | yeah |
| 05:42.13 | salviadud | i need to know this though |
| 05:42.18 | salviadud | where is the best documentation? |
| 05:42.18 | Silik0n | i use a mix of sipuras polycoms firefly xlite xpro and eyebeam at home |
| 05:42.25 | Silik0n | voip-info.org |
| 05:42.25 | brc_ | ManxPower, that, or you had the static discharge issue |
| 05:42.26 | salviadud | is the asterisk manual good enough? |
| 05:42.34 | salviadud | all righty |
| 05:42.38 | Silik0n | jbot docs |
| 05:42.39 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 05:42.39 | linsys | <PROTECTED> |
| 05:42.40 | iq | ph9kz |
| 05:42.54 | salviadud | i'll start over there |
| 05:42.54 | brc_ | salviadud, what asterisk manual? |
| 05:42.54 | Silik0n | hah |
| 05:43.01 | salviadud | the big pdf from digium |
| 05:43.01 | Silik0n | brc_ |
| 05:43.02 | brc_ | whatup yo |
| 05:43.08 | Silik0n | nadda |
| 05:43.20 | brc_ | that is about 3 years outdated |
| 05:43.22 | Silik0n | setting up the laptop w/ eclipse |
| 05:43.26 | brc_ | fun fun |
| 05:43.29 | Silik0n | yeah |
| 05:43.37 | brc_ | there's a subversion plugin for eclipse |
| 05:43.45 | Silik0n | finally got eclipse php apache and postgres all talking |
| 05:43.46 | brc_ | be nifty to make a extensions.conf eclipse plugin |
| 05:43.51 | Silik0n | yeah |
| 05:43.53 | brc_ | you ever setup mod_ssl? |
| 05:43.58 | Silik0n | yeah its easy |
| 05:44.04 | brc_ | it's being a pita to me |
| 05:44.08 | DaLion | digium |
| 05:44.13 | Silik0n | cd /usr/ports/www/mod_ssl && make && make install |
| 05:44.24 | brc_ | I'm talking about the configs |
| 05:44.25 | DaLion | oups.. thats was a 6 hours late answer .. |
| 05:44.41 | Silik0n | brc_ look at instantssl.com they have a nice cheatsheet on that shit |
| 05:45.07 | Silik0n | or if you installed from source theres a target in the Makefile for generating selfsigned certs |
| 05:45.25 | brc_ | nah, I've got the cert crap figured out |
| 05:45.27 | Silik0n | and if you are using a recent release of apache its a autoconf option to enable it now |
| 05:45.32 | DaLion | btw anyone ever gets tons of select * from exten where ? etc etc like requests |
| 05:45.44 | Silik0n | and then apache{2}ctl startssl |
| 05:46.10 | Silik0n | DaLion: realtime sucks |
| 05:46.25 | DaLion | yeah i know.. some dumbass is dosing me |
| 05:46.32 | brc_ | debian:/etc/apache2# openssl s_client -connect localhost:443 -state -debug |
| 05:46.33 | brc_ | <PROTECTED> |
| 05:46.37 | DaLion | dont know why cos asterisk logs suck |
| 05:47.01 | brc_ | uh |
| 05:47.16 | Qwell | paranoiaLevel++; |
| 05:47.23 | DaLion | and alll i can find in line 277 of res_odbc_config.c |
| 05:47.23 | DaLion | and thats says its using SQLFecth(stmt) .. but SQLFetch nowehre to be found nowehre |
| 05:47.23 | DaLion | i sassumed its from myodbc or someshit |
| 05:48.38 | DaLion | Silik0n what would one use if not realtime .. lol its needed for provider like |
| 05:48.51 | Silik0n | DaLion: i use agi i wrote |
| 05:50.26 | three55ml | DaLion: You can also do dynamic config files |
| 05:50.31 | three55ml | DaLion: And reload |
| 05:50.38 | Silik0n | well yeah I do a combination of both... |
| 05:50.55 | three55ml | Silik0n: Me too |
| 05:51.21 | DaLion | http://bugs.digium.com/bug_view_page.php?bug_id=0002979 |
| 05:51.28 | three55ml | Silik0n: Support for full config files, but I like RealTime for now :) Queues, agents, and basic extension set sending off to RT is in dynamic configs. |
| 05:51.36 | DaLion | think thats whats happening |
| 05:51.36 | DaLion | Is not problem in that we are selecting with NULL criteria on NOT NULL column? |
| 05:52.02 | *** join/#asterisk shantanoo (~shantanoo@shantanoo.user) |
| 05:52.28 | DaLion | so .. res_odbc_config.c says around line 270.. that all passed trough.. but then it dies on a null returning a null value and asterisk trying to play with the null and spwes |
| 05:52.47 | shantanoo | hi! |
| 05:53.09 | three55ml | DaLion: Are you just trying to get RT to work? |
| 05:53.35 | Silik0n | three55ml theres a variety of ways to do it... realtime limits stuff tho like MWI dont work just to name 1 |
| 05:53.41 | three55ml | DaLion: Because if you're going to use res_config_mysql just comment out res_config_odbc from the Makefiles and you'll be fine. |
| 05:53.48 | three55ml | Silik0n: Agreed |
| 05:53.58 | DaLion | no its was working for months |
| 05:53.58 | DaLion | al of a suddent its not |
| 05:53.58 | DaLion | we thinkg its some shit in extensions |
| 05:53.58 | DaLion | and asterisk not liking what it is. expet theres 3000 extensions minimum andone by one not an option |
| 05:53.59 | three55ml | Silik0n: Actually VMWI does work now in RT |
| 05:54.21 | DaLion | we using odbc i thnk |
| 05:54.25 | three55ml | DaLion: I had a similar issue but don't use ODBC for anything so I don't even load it. |
| 05:54.37 | Silik0n | and dont even get me started on ODBC storage for app_vm |
| 05:55.23 | DaLion | nah |
| 05:55.31 | DaLion | res_odbc is what is used for mysql |
| 05:55.36 | DaLion | vm is fine |
| 05:55.43 | DaLion | only that stupid exention problem |
| 05:55.47 | three55ml | DaLion: There are two ways to do it |
| 05:55.59 | three55ml | DaLion: MySQL through ODBC or directly through res_config_mysql |
| 05:56.02 | DaLion | makes asterisk loose connection and cant regain it so goes wild.. 150 mbit |
| 05:56.11 | DaLion | we use first |
| 05:56.18 | DaLion | and worked grewat till around hight load |
| 05:56.23 | Silik0n | vm uses part of res_odbc but not much... it has its own odbc calls for storage (not configs) and its nastily slow |
| 05:57.06 | DaLion | problem is i cant see what makes it barf in mysql binlogs and the asterisk full log doesnt show shit neither |
| 05:58.09 | three55ml | DaLion: Try a slow query log or something along those lines |
| 05:58.24 | DaLion | k |
| 05:58.46 | three55ml | Does the MySQL load shoot up or does Asterisk just die/ |
| 05:58.46 | DaLion | now i got nagios checking this shit with odbc show and paging me if happens |
| 05:58.47 | three55ml | ? |
| 05:59.03 | DaLion | heu |
| 05:59.04 | DaLion | asterisk load goes to 1400000 connectiosn per sec |
| 05:59.13 | DaLion | since asterisk is fucking up |
| 05:59.17 | three55ml | Ah |
| 06:00.00 | DaLion | i get millions of these all o sudden |
| 06:00.01 | DaLion | Apr 12 15:07:03 WARNING[677]: SQL Fetch error! |
| 06:00.01 | DaLion | [SELECT * FROM extensions WHERE exten LIKE ? AND context = ? AND priority = ? ORDER BY exten] |
| 06:00.01 | DaLion | Apr 12 15:07:03 WARNING[677]: SQL Fetch error! |
| 06:00.01 | DaLion | [SELECT * FROM extensions WHERE exten LIKE ? AND context = ? AND priority = ? ORDER BY exten] |
| 06:00.09 | zoa | hehe |
| 06:00.12 | zoa | still there dalion ? |
| 06:00.18 | DaLion | sorry |
| 06:00.52 | DaLion | sorry afain ;) |
| 06:00.59 | syle | you created your indexes properly on db? |
| 06:01.22 | DaLion | ? yes |
| 06:01.26 | DaLion | let me check |
| 06:01.43 | DaLion | hmm no indexe |
| 06:02.00 | Silik0n | indexs onDBs are overrated |
| 06:02.15 | three55ml | I disagree |
| 06:02.17 | three55ml | A lot |
| 06:02.25 | syle | well if your using the like command your on drugs lol |
| 06:02.38 | DaLion | hmm why |
| 06:02.43 | syle | you definately want indexes |
| 06:02.50 | DaLion | its asteirsk like command |
| 06:02.51 | zoa | the indexes wont help |
| 06:03.01 | zoa | you do want them though |
| 06:03.02 | three55ml | Not in this case, no |
| 06:03.04 | DaLion | and extensions table is made form wiki examples i think |
| 06:03.13 | three55ml | I'm just saying in general that statement isn't true |
| 06:03.27 | syle | well i was just thinking of a way to speed up his db, what i thing is happening is his max connections variable for db is possibly to low |
| 06:03.34 | syle | you check mysql debug logs? |
| 06:03.48 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 06:04.07 | DaLion | i cant |
| 06:04.07 | DaLion | i can only look in bin logs |
| 06:04.11 | DaLion | mysqlbinlog |
| 06:04.16 | three55ml | That's a good thought. What's your max connection count? |
| 06:04.29 | DaLion | even is --log=/var/log/mysql.log is on .. |
| 06:04.31 | three55ml | Setup a MySQL monitor, there's a bunch out there...and see what's happening. |
| 06:04.41 | syle | what does 'mysqladmin -pyourpassword status' show? |
| 06:05.04 | DaLion | Uptime: 12175 Threads: 6 Questions: 259590 Slow queries: 0 Opens: 0 Flush tables: 2 Open tables: 64 Queries per second avg: 21.322 |
| 06:05.16 | Qwell | 0 slow queries, so... |
| 06:05.22 | three55ml | What's your max connection count? |
| 06:05.24 | DaLion | only 6 theards all the time 2 local 4 asterisk with one being for cdr |
| 06:05.36 | DaLion | i dont know |
| 06:05.36 | DaLion | maxconnect is |
| 06:05.39 | syle | mysqladmin -pyourpassword variables |
| 06:05.40 | DaLion | default |
| 06:06.11 | three55ml | Does this happen when you experience higher call volume? |
| 06:06.27 | DaLion | <PROTECTED> |
| 06:06.33 | syle | hmmmm |
| 06:06.37 | syle | theres your problem i think |
| 06:06.38 | DaLion | yes it does around 50-60 concurent calls it does it |
| 06:06.42 | three55ml | That's pretty low, what's your limit? |
| 06:06.44 | three55ml | sorry |
| 06:06.49 | three55ml | What are your system specs |
| 06:06.54 | three55ml | RAM and CPU |
| 06:07.10 | *** join/#asterisk dec (~tom@203.87.91.78) |
| 06:07.14 | syle | set max_connections = 1500 in /etc/my.cnf |
| 06:07.15 | DaLion | CPU: Intel(R) Xeon(TM) CPU 2.80GHz (2800.12-MHz 686-class CPU) real memory = 1073479680 (1023 MB) |
| 06:07.41 | three55ml | You can easily set max_connections to 1024 or more |
| 06:07.45 | DaLion | | max_user_connections | 0 |
| 06:07.47 | three55ml | Like syle said |
| 06:07.50 | DaLion | ok 1500 it is |
| 06:08.01 | DaLion | i dont think its reading /etc/mycnf |
| 06:08.01 | Qwell | DaLion: That one just limits the number of time each user can connect. 0 is unlimited |
| 06:08.07 | three55ml | Most people never optimize MySQL :) |
| 06:08.18 | Qwell | three55ml: until its too late. ;] |
| 06:08.54 | DaLion | yes i was looking for how to change it |
| 06:08.58 | three55ml | Qwell: Yeah. I have a lot of customer doing several million records a day, you start to learn pretty quick. |
| 06:09.18 | DaLion | only thing that seem to work is if i hardcode this stuff in /usr/local/share/mysql/mysql.server |
| 06:09.28 | syle | you should have a debug log though |
| 06:09.36 | syle | cd /usr/local/mysql/data |
| 06:09.42 | DaLion | it doesnt load the conf from it i think |
| 06:09.45 | syle | should be a .err file there |
| 06:10.01 | DaLion | no data there |
| 06:10.06 | DaLion | no data dir |
| 06:10.23 | syle | well whereever dir your data is in |
| 06:10.25 | dec | Hi all, I'm running asterisk 1.0.5 - having a slight problem. I'm dialling into my Asterisk box via a PSTN landline phone, through an IAX termination provider. This works fine. From there, I type in an extension which dials another asterisk box via IAX. For some reason, I get two-way communication for about 10 seconds, and then it drops back to one-way.. audio from the PSTN -> asterisk does not come through. Any ideas why? |
| 06:10.26 | DaLion | my data is /var/db/mysql |
| 06:10.35 | DaLion | and .err says shit |
| 06:11.05 | *** join/#asterisk heison (~heison@p230.n-lapop08.stsn.com) |
| 06:11.17 | syle | i been optimizing mysql servers for years look me up if you run into any real trouble |
| 06:11.19 | *** join/#asterisk Nest0r (sdf@200.10.66.31) |
| 06:11.55 | Silik0n | syle: i have a mysql server running on a 486 that needs to handle 2gigaqueries/sec can you help me there? |
| 06:12.05 | syle | yes |
| 06:12.11 | syle | buy a new machine lol |
| 06:12.14 | Silik0n | heh |
| 06:12.47 | Silik0n | and it runs mysql 2.3.2 |
| 06:12.59 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 06:13.50 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 06:13.53 | DaLion | can max_connections be added to cmd line ? |
| 06:13.53 | DaLion | --max_conections i assume ? |
| 06:13.59 | DaLion | 2 n's |
| 06:14.14 | syle | in your startup script sure |
| 06:14.32 | syle | but that is old way to do it |
| 06:14.33 | Qwell | Why not put it in the conf file, and tell it where it is? |
| 06:14.40 | syle | my.cnf is the new thing |
| 06:14.49 | Nest0r | ŋalguien en espņaol? |
| 06:15.01 | DaLion | its not reading my /etc/my.cnf |
| 06:15.12 | clive- | hi all |
| 06:15.16 | Qwell | DaLion: maybe you aren't putting the settings in there properly |
| 06:15.24 | Qwell | [mysqld] |
| 06:15.26 | Qwell | setting=value |
| 06:15.27 | dec | anyone got any suggestions for my question above? :) |
| 06:15.28 | Qwell | right? |
| 06:15.40 | clive- | just wondering if anyone also has noticed climbing memory usage with time on asterisk |
| 06:15.51 | Qwell | dec: does latest stable, or cvs head do the same thing? |
| 06:16.04 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 06:16.07 | dec | Qwell: - i haven't tried anything above 1.0.5 yet, do you think it will help? |
| 06:16.14 | Qwell | it might. |
| 06:16.15 | DaLion | yes its already like it |
| 06:16.15 | DaLion | but i see that ps -auwx says mysqld is runing from a /bin/sh .. and not from safe_mysql |
| 06:16.18 | syle | idk my mysql server is a dual xeon 3.0 cpus, scsi drives and 8 gigs of ram, now that is fun to optimize :) |
| 06:16.19 | DaLion | so i dont know |
| 06:16.26 | three55ml | dec: I was going to recommend the same thing. Depending on what version your IAX providers are using it could have an affect. |
| 06:16.38 | dec | ahh good point. |
| 06:16.55 | drumkilla | dec: does it happen when it tries to native bridge? |
| 06:17.04 | drumkilla | you can try notransfer=yes |
| 06:17.05 | dec | it happens before it bridges |
| 06:17.07 | drumkilla | or whatever the option is |
| 06:17.13 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 06:17.13 | dec | the same problem occurs with notransfer=yes |
| 06:17.21 | three55ml | I would at least upgrade to the latest stable and see what happens. |
| 06:17.35 | dec | at first I thought it was a codec incompatibility... but i've got everything running disallow=all and allow=gsm |
| 06:17.47 | DaLion | can we mod variables runtime ? |
| 06:17.49 | DaLion | for mysql |
| 06:17.54 | DaLion | i cant afford to stop it |
| 06:18.10 | dec | you cant afford a 1 second downtime of mysql? :P |
| 06:18.23 | dec | must be pretty high availability production server |
| 06:18.27 | three55ml | sending an HUP signal shouldn't mess up too much, and I think it'll reload config. syle's the expert though |
| 06:19.02 | *** join/#asterisk ccfiel (ccfiel@210.5.72.36) |
| 06:19.11 | ccfiel | hello ppl. |
| 06:19.13 | DaLion | no i cant |
| 06:19.24 | dec | DaLion: ok. cool. :) |
| 06:19.41 | dec | as far as i know, the only way to update the max_connections is by restarting it. |
| 06:19.53 | DaLion | load is only 20 queries per sec but its still too loaded for me to stop it |
| 06:20.11 | three55ml | No slave in such a HA scenario? |
| 06:20.40 | DaLion | -O, --set-variable=name |
| 06:20.40 | DaLion | <PROTECTED> |
| 06:20.40 | DaLion | <PROTECTED> |
| 06:20.40 | DaLion | <PROTECTED> |
| 06:20.55 | DaLion | no im working on clustering this shit |
| 06:21.01 | DaLion | toomrowow |
| 06:21.05 | three55ml | Ah |
| 06:21.44 | Mazda-MX5 | ? |
| 06:21.47 | syle | you don;t have to shut it down |
| 06:21.57 | DaLion | doesnt work |
| 06:21.59 | syle | just issue a reread of the config file |
| 06:22.09 | DaLion | mysqladmin: unknown variable 'max_connections=1024' |
| 06:22.44 | syle | -O -Dmax_connections=1024 |
| 06:22.51 | syle | if your doing it that way i believe |
| 06:23.28 | DaLion | same |
| 06:23.38 | DaLion | ysqladmin: unknown variable '-Dmax_connections=1024' |
| 06:23.43 | syle | just do --max_connection=blah |
| 06:23.51 | syle | just do --max_connections=blah |
| 06:24.19 | syle | screw the -O |
| 06:24.54 | syle | i still don;t know why you just don;t use my.cnf |
| 06:25.01 | DaLion | mysqladmin: unknown variable 'max_connections=1024' |
| 06:25.04 | DaLion | same |
| 06:25.07 | Zeeek | join #farfon |
| 06:25.08 | DaLion | im on 5.02 |
| 06:25.59 | syle | echo 'max_connections=1024' > /etc/my.cnf |
| 06:26.03 | syle | if you don;t have the file |
| 06:26.16 | DaLion | i do its loaded but not used by mysqkl it seems |
| 06:26.38 | DaLion | bah fucke it ill just restart mysql in 1 hour |
| 06:26.55 | syle | do you know this for sure? mysql by default in its own binary checks for that file |
| 06:27.02 | syle | change a varaiable and find out |
| 06:27.38 | DaLion | cos i have log=/var/log/mysql and never worked |
| 06:27.54 | ccfiel | can somebody help me with my problem. i have a remote iax connection when it tries to connect to my * server and try to connect a sip client that is local. there where no delays in the coversation. but when the remote iax connects to an iax local connection in my *... there is a lag in the voice at least 2 mins.. what would be the problem? |
| 06:27.56 | syle | does the dir exist? |
| 06:28.25 | syle | #1 you should not be using 5.x series mysql in production |
| 06:28.34 | syle | thats a devel version |
| 06:28.43 | DaLion | i know |
| 06:28.45 | syle | #2 4.x should be only ones you use |
| 06:28.54 | DaLion | i didnt install them |
| 06:29.03 | DaLion | i never use latest in anything |
| 06:29.06 | want561or772did | you know what would be great? MusicOnHold on a large corporate PBX that allowed you to talk to the other disgruntled customers during the hours of waiting |
| 06:29.14 | DaLion | i use apache 1.X php 4.X etc |
| 06:29.22 | dec | hehe want561or772did |
| 06:29.24 | dec | sounds good |
| 06:29.46 | DaLion | should i use the medium file or large.cnf ? |
| 06:30.01 | want561or772did | craigslist would have to start a MusicOnHold missed connections board |
| 06:30.05 | syle | i;d use large since you have a gig of ram |
| 06:30.12 | *** join/#asterisk kenshinblade (~kenshinbl@62-101-126-208.fastres.net) |
| 06:30.23 | DaLion | k |
| 06:30.31 | DaLion | huge for 2 gigeR+ and 2 cpu + |
| 06:30.36 | DaLion | ok |
| 06:31.03 | DaLion | thread_concurrency = 8 |
| 06:31.15 | syle | set that to 2 if you have only 1 cpu |
| 06:31.46 | mog_home | can any one explain to me what the p option is for in voicemailmain |
| 06:32.31 | Qwell | If the mailbox is preceded by 'p' then the supplied mailbox is prepended to the user's entry and the resulting string is used as the mailbox number. |
| 06:32.38 | syle | only real bottlenecks you run into with mysql is write locks |
| 06:32.40 | drumkilla | there you go. |
| 06:32.42 | drumkilla | :) |
| 06:32.47 | Qwell | You too can figure out your easy answers, by typing "show application myapp" |
| 06:32.49 | syle | table locks |
| 06:33.18 | syle | this is why it is so important to optimize mysql |
| 06:33.22 | Zeeek | I can find any application called "myapp" :( |
| 06:33.32 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 06:33.35 | three55ml | Haha |
| 06:33.45 | syle | cause if a read query is taking to long it can lock out all other queries on a table |
| 06:34.09 | shantanoo | elric: you there? |
| 06:34.20 | shantanoo | with some time to spare :) |
| 06:34.39 | syle | personally i use innodb table format for high write tables |
| 06:34.50 | syle | and myisam for mostly read tables |
| 06:35.04 | Qwell | want561or772did: That actually might be a good idea, in the right situation |
| 06:35.09 | DaLion | k rebooted |
| 06:36.05 | syle | i didn;t plan on comming in here to give advice actually, i came in here as an asterisk noob to learn lol |
| 06:36.09 | shantanoo | installed asterisk and kphone on the same machine. but the kphone isn't able to register. how can i allow registration? |
| 06:36.34 | DaLion | fuck no max_connections still |
| 06:37.18 | DaLion | ok all ok |
| 06:38.00 | syle | so what does everyone do for work |
| 06:38.16 | syle | i am guessing mostly run your own voip companies etc |
| 06:38.26 | three55ml | A bit of everything |
| 06:39.20 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
| 06:39.25 | DaLion | ah |
| 06:39.49 | syle | does anyone in here actually work for someone else lol |
| 06:39.55 | DaLion | it didnt show in ps but it worked.. im now on /etc/my.cnf.. seem u need to start with rc.conf and not from rc.local |
| 06:40.03 | elric | shantanoo, yeah sort of |
| 06:40.14 | three55ml | syle: Not in a long time |
| 06:40.18 | three55ml | What do you do? |
| 06:40.56 | syle | well i finished up with spam after all the lawsuits going on, made alot of cash, investing in real estate, and thinking of a voip business now |
| 06:41.17 | shantanoo | elric: installed asterisk and kphone on same machine. how do i start now? :) |
| 06:41.28 | elric | shantanoo, pm me |
| 06:41.28 | three55ml | I just sold an Ironport :) |
| 06:41.32 | elric | :) |
| 06:41.50 | dec | an Ironport? |
| 06:41.59 | DaLion | thanks syle |
| 06:42.04 | DaLion | i owe u one or 2 |
| 06:42.20 | DaLion | thread_concurrency = does want ? excactly paralele threads ? |
| 06:42.38 | syle | if you have 1 cpu set it to 2 |
| 06:42.43 | syle | remember hyperthreading |
| 06:43.08 | DaLion | and does asterisk have that thread shit.. i see 5 asterisk connections on mysql.. cant i get more ? |
| 06:43.24 | DaLion | seems like pconnects but cant figure if it is or not |
| 06:46.25 | *** join/#asterisk pgpkeys (~pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 06:47.04 | DaLion | k thanks good night lly |
| 06:47.10 | *** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
| 06:49.48 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
| 06:49.56 | DaLion | old_passwords | OFF |
| 06:50.20 | DaLion | if i turn this on i can use them without adding set password=old_password('something') right ? |
| 06:50.30 | DaLion | but can i use both ? |
| 06:51.20 | *** join/#asterisk pgpkeys (~pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 06:51.48 | Qwell | DaLion: http://dev.mysql.com/doc/ or #mysql |
| 06:52.32 | DaLion | quwell only that syle is the expert |
| 06:52.48 | DaLion | and never got any infod on there on that |
| 06:52.52 | Qwell | DaLion: So are http://dev.mysql.com/doc/ or #mysql |
| 06:52.55 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:54.43 | pgpkeys | /clear;/n |
| 06:56.38 | *** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
| 06:56.53 | pgpkeys | sorry about that. |
| 07:02.17 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 07:05.16 | want561or772did | ManxPower, using asterisk like a traditional answering machine (being able to answer a call that's leaving a message) is possible with the manager api |
| 07:13.42 | ccfiel | can somebody help me with my problem. i have a remote iax connection when it tries to connect to my * server and try to connect a sip client that is local. there where no delays in the coversation. but when the remote iax connects to an iax local connection in my *... there is a lag in the voice at least 2 mins.. what would be the problem? |
| 07:17.35 | dec | ccfiel: codec incompatibilities? |
| 07:18.16 | ccfiel | dec: what do you mean? |
| 07:21.10 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
| 07:22.05 | dec | ccfiel: make sure that all connections are using the same codecs |
| 07:22.23 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 07:22.28 | shepherd | hi |
| 07:22.30 | dec | GSM, g723.1, etc |
| 07:22.34 | ccfiel | dec: yes it uses gsm |
| 07:22.40 | dec | oh k |
| 07:22.46 | dec | no idea then, sorry :) |
| 07:22.56 | ccfiel | dec: ok thanks anyway |
| 07:23.34 | pbxjunkie | anybody in here have a quadbri w/ a 2.6 kernel?:) |
| 07:24.47 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 07:24.51 | Qwell | oh, thats freaking stupid...if I call my work with my cidnum set to my (valid) number, it can't get through. If I set it to a fake number, it works fine |
| 07:25.02 | shantanoo | 'sip show peers' shows me the 2 users |
| 07:25.13 | shantanoo | but they can't call each other |
| 07:25.17 | *** join/#asterisk rainfall (~blah@wnpgmb02dc1-57-192.dynamic.mts.net) |
| 07:25.29 | shantanoo | 'call failed: Not Found' <--- error |
| 07:25.34 | shantanoo | any idea? |
| 07:26.05 | shepherd | shantanoo: same network? |
| 07:26.15 | shantanoo | shepherd: yes |
| 07:26.18 | shantanoo | its on intranet |
| 07:26.20 | shepherd | nope sorry! |
| 07:26.21 | shepherd | :) |
| 07:26.53 | *** join/#asterisk Cronus-sct (~none@d5152D609.access.telenet.be) |
| 07:27.06 | shantanoo | :(( |
| 07:27.08 | shantanoo | :) |
| 07:29.25 | Cronus-sct | i'm doing some resaerch on how to use a quadBRI card, to connect to a legacy PBX, before I buy such a rather expensive card, but I don't seem to find a lot on voip-info nor google |
| 07:29.36 | Cronus-sct | can anyone give me some pointers to search? |
| 07:31.44 | Cronus-sct | i would like to put asterisk in between the pstn and the panasonic PBX, do I have enough with the quadBRI or do I have to buy a power supply thing for the NT lines? |
| 07:31.58 | Cronus-sct | i've seen this on some sites |
| 07:33.45 | Cronus-sct | thx in advance |
| 07:34.57 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 07:36.58 | pbxjunkie | Cronus-sct: I've just bought a quad bri |
| 07:38.11 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 07:38.35 | *** join/#asterisk pgpkeys (~pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 07:38.53 | pbxjunkie | if you want to connect asterisk w/ a panasonix pbx then you probably need a card with analog ports |
| 07:38.54 | shantanoo | kphone needs which port open? |
| 07:39.02 | shantanoo | i think its firewall problem now. |
| 07:42.24 | Cronus-sct | pbxjunkie We get 4 ISDN lines from the telephone company. Was it easy to install the card? |
| 07:44.09 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 07:44.40 | pbxjunkie | i have yet to install the card, the documentation is VERY little, I've experienced compile errors with their driver and they have yet to answer my support e-mails |
| 07:44.59 | pbxjunkie | actually.. there's NO documentation on their card or the bristuff driver |
| 07:45.13 | syle | how do you take an existing phone line a user has and turn it into a voip number exactly |
| 07:45.18 | syle | call up the phone company? |
| 07:46.35 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 07:46.52 | pbxjunkie | the problem with european ISDN equipment manufacturers is that they're mostly german |
| 07:46.56 | brc_ | syle, there is *NO* such thing as a "voip number" |
| 07:47.46 | pbxjunkie | AVM (who makes C1, C2, C4) all docs save a few, are in German, forum posts... internet sites.. all in german.. and I can't figure out anything |
| 07:48.38 | syle | brc_ : how do you convert an exising anolog number to a digital number so you can call it up with sip |
| 07:49.05 | syle | hmmmm |
| 07:49.09 | *** join/#asterisk kore (kore@mindwipe.org) |
| 07:49.10 | pbxjunkie | your phone provider must do that |
| 07:49.13 | brc_ | IT DOESN"T WORK THAT WAY |
| 07:49.13 | cypromis | pbxjunkie: a junghanns quadbri card ? |
| 07:49.16 | syle | ata convertor my guess |
| 07:49.17 | pbxjunkie | cypromis: yes |
| 07:49.21 | cypromis | the junghanns btistuff has enough docs |
| 07:49.23 | brc_ | cypromis, duuuude |
| 07:49.27 | cypromis | also examples etc |
| 07:49.27 | pbxjunkie | cypromis: really?:) where? :D |
| 07:49.31 | cypromis | morn brc_ :) |
| 07:49.38 | cypromis | pbxjunkie: in the tarball for bristuff |
| 07:49.38 | brc_ | how was the weekend? |
| 07:49.39 | brc_ | :p |
| 07:49.51 | cypromis | plus on the pdf that describes the jumper settings |
| 07:49.52 | pbxjunkie | the tarball for bristuff has 1 file, called install .. and it hass. .let me tell you |
| 07:49.54 | cypromis | and it is in english |
| 07:49.56 | *** join/#asterisk ta[i]nted (~tainted@adsl-69-108-101-61.dsl.irvnca.pacbell.net) |
| 07:50.10 | cypromis | pbxjunkie: don't tell me I distribute the stuff |
| 07:50.11 | cypromis | lol |
| 07:50.17 | cypromis | brc_: nice :) |
| 07:50.19 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l03m-16-26.d4.club-internet.fr) |
| 07:50.27 | pbxjunkie | 52 lines of docs :D |
| 07:50.42 | pbxjunkie | cypromis: you distribute the stuff? meaning? |
| 07:51.11 | cypromis | I sell it for example |
| 07:51.17 | pbxjunkie | ooh.. i see |
| 07:51.20 | cypromis | is a good definition of distributing it |
| 07:51.42 | newl | 15oz will get ya 20? :) |
| 07:52.24 | pbxjunkie | cypromis: so you can tell me why the latest bristuff tarball doesn't compile then?:) |
| 07:52.26 | cypromis | it doesn't ? |
| 07:52.29 | pbxjunkie | it fetches it's own stuff.. zaptel.. asterisks..patches them |
| 07:53.00 | pbxjunkie | then the driver compiles w/ 4-5 warnings and can't be inserted into the kernel |
| 07:53.12 | cypromis | hmm seems we are running a financial companies callcenter on a non compiled software than |
| 07:53.48 | cypromis | how about a paste of the error message into pastebin.ca ? |
| 07:55.50 | *** join/#asterisk Koshatul (~evangelio@inf-203-132-65-157.bne.ipnetworks.net.au) |
| 07:56.09 | *** join/#asterisk jeffik (jefik@69.158.0.103) |
| 07:56.46 | Cronus-sct | When I succesful connect the quadBRI to my panasonic PBX, do I need extra software to be able to forward a fax to the panasonic? |
| 07:57.14 | Cronus-sct | can asterisk detect a fax signal? |
| 07:57.28 | cypromis | pbxjunkie: ? |
| 07:57.33 | cypromis | Cronus-sct: yes |
| 07:57.46 | Cronus-sct | nice |
| 07:57.52 | pbxjunkie | http://pastebin.ca/9465 |
| 07:57.53 | cypromis | what do you mean by forwarding |
| 07:58.22 | Cronus-sct | i receive a fax from the outside and asterisk has to detect it and send it to the panasonic pbx |
| 07:58.31 | cypromis | pbxjunkie: those warnings are no problem |
| 07:58.48 | pbxjunkie | cypromis: they are when you try to insmod the driver |
| 07:58.48 | cypromis | can I have the error on inserting into the kernel plus your zaptel.conf ?? |
| 07:58.48 | three55ml | Cronus-sct: Yes, goto http://www.voip-info.org and search for Fax |
| 07:58.57 | pbxjunkie | eer.. sure hang on |
| 08:00.17 | *** join/#asterisk shantanoo (~shantanoo@shantanoo.user) |
| 08:00.38 | shantanoo | back ;) |
| 08:01.14 | PTG123 | anyone here know ip500s well? |
| 08:01.35 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 08:02.40 | pbxjunkie | cypromis: u see the pastebin? |
| 08:03.24 | cypromis | checking sorry was on the phone |
| 08:03.27 | pbxjunkie | np |
| 08:03.57 | cypromis | pbxjunkie: that is compile time wanrings |
| 08:04.02 | cypromis | I need to see zaptel.conf |
| 08:04.02 | cypromis | and |
| 08:04.09 | pbxjunkie | it's there! |
| 08:04.11 | cypromis | errors showing after modules loading |
| 08:04.26 | pbxjunkie | after the second seperator |
| 08:04.30 | pbxjunkie | module never loads |
| 08:05.11 | cypromis | aah it's in 9466 |
| 08:05.24 | pbxjunkie | yes |
| 08:05.41 | cypromis | run a depmod -ae |
| 08:05.50 | cypromis | and use modprobe instead of insmod |
| 08:05.54 | PTG123 | someone must know polycom phones :) |
| 08:06.11 | cypromis | PTG123: I know the conferencing ones |
| 08:06.12 | cypromis | ;) |
| 08:07.10 | pbxjunkie | hmm... |
| 08:07.43 | pbxjunkie | that seemed to work :D |
| 08:08.34 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 08:09.10 | cypromis | yeah you where not loading zaptel with it |
| 08:09.25 | pbxjunkie | cypromis: check out the bottom : http://pastebin.ca/9467 |
| 08:09.25 | cypromis | so the dependencies where not fullfilled and the kernel could not find the zaptel stuff |
| 08:09.58 | pbxjunkie | every doc I've read anywhere said insmod, both voip-info and their ./INSTALL file |
| 08:10.08 | cypromis | pbxjunkie: are you using terminators on the bus ? |
| 08:10.11 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 08:10.35 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 08:10.35 | pbxjunkie | cypromis: all the buses are going to be NT mode |
| 08:10.39 | pbxjunkie | all the ports I mean |
| 08:10.42 | newl | insmod should only be used if you know there are no other dependancies by the module you're loading. |
| 08:12.12 | cypromis | pbxjunkie: I run a setup here with 4 octobri's per box |
| 08:12.23 | cypromis | you occasionally get crc errors but they are harmless |
| 08:12.34 | cypromis | caused by cable lengths and also by initialisation of phones |
| 08:12.46 | cypromis | yours sounds more like the wires are not connected to anything yet ? |
| 08:14.02 | pbxjunkie | i've got 2 wires on ports 1-2 atm connected to S0 buses.. ports 3-4 got nothing on them |
| 08:14.56 | pbxjunkie | or it could be the other way round (3-4 w cables , 1-2 nothing) |
| 08:15.28 | facek_ | cypromis hi |
| 08:15.35 | cypromis | morn |
| 08:15.59 | cypromis | yu will get CRC errors on initialisation of equippment or on non connected wires |
| 08:16.06 | cypromis | typically at least |
| 08:16.11 | *** join/#asterisk TheEmperor (~TheEmpero@203.121.47.165) |
| 08:16.21 | PTG123 | anyone use a polycom phone here? |
| 08:16.36 | *** join/#asterisk pino (~z@host167-115.pool80116.interbusiness.it) |
| 08:19.32 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:32.36 | *** join/#asterisk zoa (~zoa@213.16.46.130) |
| 08:41.06 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:44.12 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 08:50.43 | *** join/#asterisk ta[i]nted (~tainted@adsl-69-108-101-61.dsl.irvnca.pacbell.net) |
| 08:52.43 | pbxjunkie | cypromis: thanks for the tips. |
| 08:52.54 | pbxjunkie | :D |
| 09:05.44 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 09:08.36 | RoyK | ~seen inspired |
| 09:08.38 | jbot | inspired <mikael@213.197.167.61> was last seen on IRC in channel #asterisk, 1d 20h 58m 1s ago, saying: 'coc'. |
| 09:08.53 | Zeeek | ~lart RoyK |
| 09:11.57 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:17.51 | newl | ><> |
| 09:21.46 | *** join/#asterisk Cronus-sct (~none@d5152D609.access.telenet.be) |
| 09:23.39 | RoyK | pino: I'd guess herring could be held in an aquarium, given, say, 100 herring and a 100.000 litre tank to start with... a little more than what I've got |
| 09:23.59 | Zeeek | What CPU and RAM are needed for a 100 herring system? |
| 09:25.11 | zoa | wtf is a herring ? |
| 09:25.19 | Zeeek | ~herring |
| 09:25.21 | zoa | fish ? |
| 09:25.25 | Zeeek | ya |
| 09:25.32 | Zeeek | delicious smoked |
| 09:25.33 | zoa | NOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO |
| 09:26.41 | RoyK | zoa: http://en.wikipedia.org/wiki/Herring |
| 09:26.58 | Zeeek | I like 'em in vinegar |
| 09:27.18 | RoyK | :) |
| 09:29.01 | *** join/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
| 09:29.05 | MuppetMaster | Hello. |
| 09:29.18 | MuppetMaster | Anyone know what the AMP uname/passwd for Asterisk@Home v0.8 us? |
| 09:29.24 | MuppetMaster | suppose to be admin/password but that does not seem to work. |
| 09:31.23 | MuppetMaster | AMP = Asterisk Managment Portal...http://amp.coalescentsystems.ca/ |
| 09:31.34 | MuppetMaster | RoyK: Were you really talking to me? |
| 09:33.11 | MuppetMaster | Found it on the AMP website, as the Asterisk@Home documentation is wrong. Should be wwwadmin/password |
| 09:33.20 | *** join/#asterisk pbxjunkie (~stormtroo@213.5.44.113) |
| 09:34.18 | *** join/#asterisk TheEmperor (~user@203.121.47.165) |
| 09:37.25 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 09:42.29 | Zeeek | what is my root password? |
| 09:42.39 | Zeeek | I have asterisk running on this machine |
| 09:42.48 | MuppetMaster | Zeeek: ? |
| 09:42.54 | Zeeek | <joke> |
| 09:43.15 | MuppetMaster | Ah, making fun of me. |
| 09:43.42 | MuppetMaster | Can't help it if the Asterisk@Home documentation is wrong. |
| 09:43.48 | MuppetMaster | But laugh if you must... |
| 09:43.49 | Zeeek | I know |
| 09:44.06 | Zeeek | but someone asks that question often. I don't laugh... honest |
| 09:44.22 | MuppetMaster | Yes, hence why the documentation needs to be fixed. |
| 09:44.25 | Zeeek | but @hole needs its own doc and info resources |
| 09:44.38 | Zeeek | errr |
| 09:44.41 | MuppetMaster | I have never used AMP...prefer the command line, but helping someone else who wants the GUI... |
| 09:44.42 | Zeeek | @home |
| 09:44.50 | MuppetMaster | @home does have it. |
| 09:44.52 | Zeeek | there is a definite need for gui |
| 09:45.10 | Zeeek | but the damn things need to have their own documentation and it needs to be good |
| 09:45.18 | Zeeek | since people who want this kind of thing need it |
| 09:45.21 | MuppetMaster | @home does have it's own and it is quite good. |
| 09:45.25 | Zeeek | need, need, need |
| 09:45.26 | MuppetMaster | Just happen to have the uname/passwd wrong. |
| 09:45.38 | Zeeek | that sicks. Oh well, you found it |
| 09:45.41 | Zeeek | sucks |
| 09:48.07 | MuppetMaster | I don't think so. |
| 09:48.18 | MuppetMaster | I think a phenomenal job has been done with Asterisk@Home, including the docs. |
| 09:48.25 | MuppetMaster | Just happen to have a significant error. |
| 09:48.28 | MuppetMaster | To err is human... |
| 09:49.32 | *** part/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
| 09:50.26 | *** join/#asterisk CiNzAs (~ashes@83.240.144.145) |
| 09:51.58 | CiNzAs | morning |
| 09:52.31 | Zeeek | the "sucks" was that the doc is wrong, not the products |
| 09:54.15 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 09:57.09 | clive- | does anyone know if the newer versions of asterisk will work without libpri compilling correctly ? |
| 09:57.34 | zoa | i think they do |
| 09:58.17 | clive- | zoa thanks:), I am trying my best not to recomplie the kernel another 25 times before I get it working again |
| 09:58.36 | clive- | esp since I am using chan_capi |
| 09:59.25 | *** join/#asterisk abracsas (~abuono@217.9.64.150) |
| 10:05.28 | *** join/#asterisk jackthe (~jesse@d594f03e.ftth.concepts.nl) |
| 10:15.37 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 10:18.05 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 10:27.19 | *** join/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 10:28.22 | saabluvr | Hi everyone... Doe anyone know why the "serveremail" in voicemail.conf is ignored ? |
| 10:29.23 | saabluvr | my mailserver does not like senders without fq domains :( |
| 10:31.09 | saabluvr | the command "hostname" shows the complete server address, but although i set the serveremail in voicemail.conf , the sender is root@machinename and not the address set in voicemail.conf ... |
| 10:31.15 | tafazzi | ciao |
| 10:31.43 | saabluvr | ciao |
| 10:32.13 | Alexi1 | <PROTECTED> |
| 10:32.42 | Alexi1 | because the voicemail are own by root only |
| 10:34.15 | saabluvr | that would be fine with me, but the receiving mailserver sees as sender "root@voip-machine" and not "root@voip-machine.somwhere.com" |
| 10:34.27 | saabluvr | and my SPF says : deny ... |
| 10:39.33 | pbxjunkie | can somebody look at my zapata.conf and tell me WHY asterisk doesn't see my channels? http://pastebin.ca/9469 |
| 10:43.21 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
| 10:43.55 | pbxjunkie | I get no channel type registered for Zap :/ |
| 10:53.19 | pbxjunkie | cypromis?:) |
| 10:53.21 | cj-rm | Has anyone here used asterisk to programatically instantiate a telephone call between two parties (3rd pary call control)??? i.e. Some software tells Asterisk to ring phone A, then phone B and upon both parties answering Asterisk joins the calls together? |
| 10:53.55 | cj-rm | But where Asterisk is not an end-point for the calls |
| 10:54.08 | cj-rm | Is that possible? |
| 10:56.27 | cypromis | are you sure you have chan_zap.so loaded ? |
| 10:57.28 | cypromis | and the card jumpered to NT mode ? |
| 10:57.53 | cypromis | sorry you have it in te mode as I see now |
| 10:58.00 | mozrat | Guys could anyone help me load the SIP image onto a Cisco 7960 phone? I have a working TFTP server with the files as detailed in the Cisco doc, but when I boot the phone it doesn't even look for the OS79XX.TXT file |
| 11:01.25 | cj-rm | Has anyone here used Asterisk for 3rd party call control with SIP??? |
| 11:02.09 | Kumbang | hello, anyone works with mfc r2 here? |
| 11:02.15 | pbxjunkie | cypromis: chan_zap.so is not in modules.conf :D |
| 11:02.23 | pbxjunkie | cypromis: shouldn't it be there by default ? :/ |
| 11:04.01 | cypromis | no |
| 11:05.22 | cj-rm | is 3rd party call control even possible with Asterisk? |
| 11:09.11 | zoa | signalling = bri_cpe_ptmp |
| 11:09.11 | zoa | 019 ; p2p TE mode (for connecting ISDN lines in point-to-point mode) |
| 11:09.11 | zoa | 020 ;signalling = bri_cpe |
| 11:09.11 | zoa | 021 ; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode) |
| 11:09.11 | zoa | 022 ;signalling = bri_net_ptmp |
| 11:09.12 | zoa | 023 ; p2p NT mode (for connecting an ISDN pbx in point-to-point mode) |
| 11:09.13 | zoa | 024 signalling = bri_net |
| 11:09.24 | zoa | why oh why do you have 2 signallings ? |
| 11:10.04 | zoa | is that for chan_capi ? |
| 11:12.25 | pbxjunkie | I changed that I only use one |
| 11:12.30 | pbxjunkie | the thing is I'm not sure which one to use |
| 11:13.07 | pbxjunkie | I jumpered the card in NT mode, so I connect the telco's S0 bus onto the card |
| 11:13.43 | *** join/#asterisk The_Ball (~alex@static-227.35.240.220.dsl.comindico.com.au) |
| 11:14.33 | *** join/#asterisk LeoB (~chatzilla@c-66-31-41-1.hsd1.ma.comcast.net) |
| 11:15.23 | pbxjunkie | check out: http://pastebin.ca/9471 , the message I get when * tries to load chan_zap.so .. I think it's a compilation problem |
| 11:16.26 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 11:17.00 | CiNzAs | is there any command to unset a variable ? |
| 11:17.02 | CiNzAs | via CLI |
| 11:17.29 | *** join/#asterisk che (~che@che.user) |
| 11:18.29 | che | heyyas. i am at a buying decision maybe someone got some first hand experience with the 4 channel isdn pci cards. either i am gonna take this one: http://www.sirrix.de/content/pages/pci4s0.htm or this one: http://shop.beronet.com/product_info.php/cPath/21_25/products_id/39?osCsid=9fe9c2b0b295cf55e8ebfa5cd971e8aa comments and suggestions very welcome. |
| 11:18.29 | CiNzAs | I would like to unser a TRANSFER_CONTEXT variable |
| 11:26.13 | *** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au) |
| 11:26.14 | Delmar | if I have a single line such as exten => fax,1,rxfax(/var/spool/asterisk/fax/test.tif) .. it works fine and I end up with a fax received called test.tif but if i change the line to exten => fax,1,Goto(ext-fax,in_fax,1) which is really what I need, it breaks and wont even spawn the fax extension...* console says .. redirecting to fax extension, but then it just causes a time out. |
| 11:26.23 | Delmar | Goto broken? |
| 11:27.23 | webman | so you have a context called ext-fax and a exten called in_fax ?? |
| 11:27.45 | Delmar | yep |
| 11:27.49 | webman | what does show dialplan ext-fax show? |
| 11:28.17 | Delmar | sec. im gonna hak up the ext-fax context and make it basic then retest, to prove the goto. |
| 11:29.23 | webman | delmar: could also try something like "exten => fax,1,noop(here I am)\nexten => fax,2,goto(etc...) |
| 11:29.41 | webman | then on console, you should see the output from the noop, and then the goto..... |
| 11:31.32 | Delmar | and its intermittantly saying... Fax detected, but no fax extension... i changed nothing at all, and retried, and the fax received. |
| 11:31.50 | Delmar | that was when i had just the line exten => fax,1,rxfax(/var/spool/asterisk/fax/test.tif) |
| 11:31.54 | Delmar | and nothing else. |
| 11:32.10 | Delmar | so its broken half the time without me doing anything for a start. |
| 11:33.48 | Delmar | so now i have under my [incomingFXO] context... exten => fax,1,Goto(ext-fax,in_fax,1) and then under context [ext-fax] exten => in_fax,1,rxfax(/var/spool/asterisk/fax/test.tif) |
| 11:34.03 | Delmar | just failed claiming there is no fax extension |
| 11:34.29 | pbxjunkie | arrghhh :/ |
| 11:34.29 | pbxjunkie | Apr 13 14:34:01 WARNING[21295]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death |
| 11:34.29 | pbxjunkie | Apr 13 14:34:01 WARNING[21295]: loader.c:391 load_modules: Loading module chan_zap.so failed! |
| 11:35.07 | Delmar | and its doing that again. |
| 11:36.32 | *** join/#asterisk ccfiel (ccfiel@210.5.72.36) |
| 11:36.32 | ccfiel | hello ppl.. |
| 11:36.32 | ccfiel | can somebody help me with my problem. i have a remote iax connection when it tries to connect to my * server and try to connect a sip client that is local. there where no delays in the coversation. but when the remote iax connects to an iax local connection in my *... there is a lag in the voice at least 2 mins.. what would be the problem? :( |
| 11:36.32 | zoa | native bridging |
| 11:36.32 | zoa | NEXT! |
| 11:36.40 | webman | pbxjunkie: rm /usr/lib/asterisk/modules/*; cd /usr/src/asterisk;make install |
| 11:37.08 | webman | Delmar, I don't really know, I don't use the fax auto-detect, but sounds like you need to solve that problem first .... |
| 11:37.51 | *** part/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 11:38.19 | ccfiel | hello?? |
| 11:38.29 | Mavvie | *CLI> sip show peer test5 |
| 11:38.30 | Mavvie | <PROTECTED> |
| 11:38.30 | ccfiel | can somebody help me with my problem.. |
| 11:38.37 | Mavvie | still have no idea what that number there is :-/ |
| 11:38.46 | webman | hmmm, could someone explain gain values to me please... I am looking at adjusting the tx gain on a polycom phone, the default value is 3, and I want to make it quieter... should I make it a higher or lower number? |
| 11:38.50 | RoyK | ccfiel: if you ask, perhaps. |
| 11:38.55 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 11:39.25 | ccfiel | can somebody help me with my problem. i have a remote iax connection when it tries to connect to my * server and try to connect a sip client that is local. there where no delays in the coversation. but when the remote iax connects to an iax local connection in my *... there is a lag in the voice at least 2 mins.. what would be the problem? :( |
| 11:39.28 | webman | Mavvie: Wouldn't that be the time until the registration expires? |
| 11:39.36 | ccfiel | RoyK: :) |
| 11:39.41 | tzanger | ccfiel: two minutes?? |
| 11:39.55 | tzanger | ccfiel: you have a DNS lookup that is timing out? |
| 11:40.08 | RoyK | 2 minutes lag????? |
| 11:40.12 | Mavvie | webman: that's what I thought, but the expiry is 1 hour (3600 seconds), which makes this a weird number. |
| 11:40.21 | ccfiel | tzanger : what do you mean.. |
| 11:40.27 | Mavvie | webman: and it doesn't decrease neither. |
| 11:40.34 | ccfiel | yes 2mins...sometimes 1 min... |
| 11:40.46 | RoyK | ccfiel: not seconds? |
| 11:40.58 | ccfiel | no minutes.. |
| 11:41.02 | webman | Mavvie: in that case, I can only suggest you "Use the source, Luke" :).... or just ignore it .... |
| 11:41.06 | ccfiel | not seconds.. |
| 11:41.24 | ccfiel | but if you will wait you can hear the other end..but its very lag.. |
| 11:42.26 | ccfiel | it happen between iax remote ---> * server -----> iax local |
| 11:42.53 | ccfiel | but when i do iax remot ----> * server -----> sip local theres no lag... |
| 11:43.16 | ccfiel | the communication is good |
| 11:44.41 | jackthe | ccfiel: question, are you using the new jitterbuffer? |
| 11:45.01 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:45.14 | ccfiel | jackthe: what do you mean? |
| 11:45.35 | ccfiel | jackthe: where can i find that? :( |
| 11:45.45 | Mavvie | webman: heh... it's the Id of the expire, not the time. |
| 11:45.51 | jackthe | I mean, wich program do you use for iax local? |
| 11:46.10 | webman | mavvie: of course it is... why didn't we both know that :) |
| 11:46.34 | Delmar | the fax autodetection is working ok. its something to do with the extension. |
| 11:46.36 | Mavvie | useless information to print.... |
| 11:47.22 | Delmar | if I start using a Goto to another context and such, it breaks .. claiming there is no fax extension. |
| 11:47.27 | ccfiel | jackthe: you mean the softphone..i used media X both local iax and remote iax connection.. http://www.marccharbonneau.com/asterisk/mediaxphone.php#Support |
| 11:47.48 | ccfiel | in my sip connection i used x-lite |
| 11:48.02 | RoyK | ccfiel: using HEAD or what? |
| 11:49.50 | ccfiel | RoyK: sorry for my ignorance... what do you mean by HEAD? |
| 11:50.09 | jackthe | ccfiel: If possible, can you make a packettrace off the callsetup and the first 2 min of call (lets say until you hear audio) |
| 11:50.41 | webman | ccfiel: type show version and paste the response you see |
| 11:54.38 | jackthe | to much to handle for ccfiel :P |
| 11:55.03 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 11:59.58 | *** join/#asterisk smiley- (~smiley@h239n2fls33o1123.telia.com) |
| 12:04.49 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 12:07.34 | smiley- | is it possible to customize the voicemail-app without modfify the source code? I have a few options I want to disable |
| 12:08.20 | smiley- | I have been thinking about replacing the audiofiles with silence as one solution.. but if it is possible to disable functions it's much better |
| 12:09.52 | webman | smiley: well, you can customise the source code, or the sound files, or you can use some options from show application voicemailmain if they will do what you want.... |
| 12:11.30 | smiley- | ok.. I guess I have to do the sound file trick then |
| 12:15.42 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:18.40 | Cronus-sct | in your extensions.conf, do you always have to do an answer and a hangup? |
| 12:19.07 | Mavvie | no. |
| 12:19.48 | Cronus-sct | thx |
| 12:19.55 | webman | cronus-sct: but often you want to .... |
| 12:19.56 | dizzydiffi | hello |
| 12:20.00 | dizzydiffi | nyone there |
| 12:20.08 | webman | nope |
| 12:20.12 | dizzydiffi | nope |
| 12:20.14 | Cronus-sct | :) |
| 12:20.25 | dizzydiffi | pls i need help |
| 12:20.35 | dizzydiffi | with sip to h323 translation |
| 12:21.14 | dizzydiffi | has anyone done this |
| 12:21.27 | pepzi | what is this flash-function in x-ten eyebeam and on many sip-phones? |
| 12:21.40 | RoyK | dizzydiffi: asterisk h.323 channels SUCK |
| 12:21.47 | RoyK | there really are no good solutions for it |
| 12:21.51 | dizzydiffi | yea it might but i need to do it |
| 12:22.10 | dizzydiffi | what of Open h323 with gnungk |
| 12:22.16 | Mavvie | webman: http://bugs.digium.com/bug_view_page.php?bug_id=0004022 <- fixed! |
| 12:23.33 | webman | mavvie: neat :) now that is good turn around time :) |
| 12:23.48 | smiley- | oh.. that reminds me.. show sip peers isn't working perfectly when using a mix of entries from real time config mysql and some static ones in sip.conf |
| 12:26.16 | smiley- | if I remove that static one from sip.conf all the peers from real time are showing up.. |
| 12:26.29 | smiley- | maybe I should update to the latest CVS |
| 12:34.23 | webman | if gain is 3, how do I make the volume quieter, increase to 4, or decrease to 2 ?? |
| 12:34.53 | smiley- | decrease seems logical.. |
| 12:36.40 | webman | smiley-: well. most things to do with audio don't seem to be so logical for me :) I'll try and see... |
| 12:37.55 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 12:39.46 | webman | well, time for sleep for me now... cya... |
| 12:43.55 | *** join/#asterisk phpboy (~sj@tpr-165-252-251.telkomadsl.co.za) |
| 12:44.05 | phpboy | guys, what's a nice softphone that supports IAX? |
| 12:44.11 | phpboy | X-ten doesn't seem to support it |
| 12:44.11 | phpboy | <PROTECTED> |
| 12:44.19 | ariel_ | diax |
| 12:44.37 | phpboy | u got a URL for that? |
| 12:45.57 | oden | phpboy: i have packaged some for mandriva (cooker), iaxcomm, tkiaxphone and kiax. |
| 12:45.57 | ariel_ | phpboy, http://www.laser.com/dante/ |
| 12:46.58 | oden | phpboy: ehh, sorry. that is linux :) |
| 12:47.55 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 12:48.03 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 12:49.33 | phpboy | all good :) |
| 12:50.26 | newl | oden: uploaded to contrib or maintained elsewhere? |
| 12:51.01 | oden | newl: cooker. |
| 12:51.25 | newl | those are in main? |
| 12:51.25 | oden | newl: i mean, yes. into the mandriva cooker contrib rep. |
| 12:51.34 | oden | newl: no. |
| 12:51.41 | newl | ahh, okay. I was gonna say, I didn't recall seeing them in main. hehe |
| 12:52.57 | oden | newl: they're fresh, i did it monday. iaxclient is a shared lib used by kiax too. |
| 12:53.23 | newl | grabbing now. |
| 12:53.34 | newl | whoops..libiax0 is missing |
| 12:56.55 | Cronus-sct | if you connect an asterisk server to a legacy PBX, is it possible to use Music on Hold? |
| 12:57.12 | Cronus-sct | is it also possible to give each internal number a voicemail? |
| 12:57.16 | *** part/#asterisk JunK-H (~grepmoo@65.39.228.5) |
| 12:57.30 | RoyK | I guess that depends on the other pbx |
| 12:57.38 | Cronus-sct | it's a panasonic |
| 12:58.07 | Cronus-sct | if that helps |
| 12:58.08 | RoyK | I have no idea what's inside that..... |
| 12:58.52 | Cronus-sct | i'll find out that when i implement it |
| 12:59.41 | ManxPower | ~docs |
| 12:59.44 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:59.45 | ManxPower | ~mailinglist |
| 12:59.46 | jbot | i heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 13:01.16 | ManxPower | ariel_, I figured out the problem with the TDM400P that I had. |
| 13:02.08 | ariel_ | what problem? and how? |
| 13:02.33 | ManxPower | ariel_, turns ou the card was the original first version of the TDM400P,l the one without the power connector |
| 13:03.06 | phpboy | hmmm |
| 13:03.18 | phpboy | diax isn't even trying to connect to my asterisk server |
| 13:03.19 | phpboy | :/ |
| 13:04.02 | ariel_ | ManxPower, ahh I see. I have never seen one without the plug. |
| 13:04.31 | oden | are there other open source iax soft phones i don't know about besides iaxcomm, tkiaxphone and kiax? |
| 13:04.55 | Moonwick | tried ikilledemwithaniax? |
| 13:05.24 | ariel_ | oden, for linux or windows? |
| 13:05.41 | oden | that builds under linux. |
| 13:06.36 | ariel_ | don't know of any other. there is an xlite sip beta out there for linux |
| 13:07.17 | *** join/#asterisk Chad-wl (~asdf@207.164.188.10) |
| 13:08.13 | *** join/#asterisk bet (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:08.26 | phpboy | anybody got some nice asterisk DOCS? |
| 13:08.36 | phpboy | asteriskdocs.org isn't all that informative :/ |
| 13:08.45 | oden | X-Lite? seems for ms or mac |
| 13:09.02 | Chad-wl | I'm trying to dial out over a TM400 card on the first channel, should my TRUNk=Zap/1-1 ? |
| 13:10.32 | ariel_ | Chad-wl, no TRUNK-Zap/1 |
| 13:14.52 | *** join/#asterisk easimon (~easimon@balu.kawo2.RWTH-Aachen.DE) |
| 13:15.06 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 13:15.34 | *** join/#asterisk betul (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:15.39 | *** part/#asterisk betul (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:15.53 | *** join/#asterisk illek (~Mike@ip68-13-238-168.ok.ok.cox.net) |
| 13:16.04 | *** join/#asterisk DrJolo (~chatzilla@217.153.194.10) |
| 13:16.07 | *** join/#asterisk betul (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:16.11 | *** part/#asterisk betul (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:16.31 | *** join/#asterisk betul (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:17.14 | *** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
| 13:17.17 | ManxPower | All SoftPhones Suck! |
| 13:17.46 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 13:17.50 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 13:17.56 | phpboy | when I phone using SIP |
| 13:18.07 | phpboy | sometimes funny noices come over the line |
| 13:18.18 | phpboy | could this possibly be hardware related? |
| 13:18.25 | phpboy | compression issues of some sort? |
| 13:18.33 | *** join/#asterisk betul (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:18.45 | *** join/#asterisk jt_ (~jt@66.28.34.162) |
| 13:18.48 | tzanger | phpboy: not compression but perhaps ethernet weirdness |
| 13:19.05 | phpboy | hmmm, u think? |
| 13:19.05 | tzanger | phpboy: i've seen that before where the card (or hdd controller) holds interrupts for too long and you get "chirps" |
| 13:19.19 | phpboy | yeah |
| 13:19.26 | phpboy | exactly that |
| 13:19.28 | tzanger | phpboy: try playing with the PCI latency timers; set them all to 32 or 64 to start |
| 13:19.38 | tzanger | I bet tehre's something sitting in your system with a latency value of like 255 |
| 13:19.40 | phpboy | so possibly hdd related |
| 13:19.42 | *** join/#asterisk easimon (~easimon@balu.kawo2.RWTH-Aachen.DE) |
| 13:19.50 | phpboy | hmmm, how can I check? |
| 13:19.50 | *** join/#asterisk betul (~bet@slip139-92-59-114.ist.tr.prserv.net) |
| 13:19.55 | phpboy | or where, rather |
| 13:20.06 | tzanger | lspci -v | grep latency= |
| 13:20.12 | *** join/#asterisk pif (~pif@mail.conceptbau.de) |
| 13:20.22 | tzanger | ~google pci latency tune asterisk |
| 13:20.27 | phpboy | Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 |
| 13:20.30 | phpboy | that's what I get back? |
| 13:20.37 | tzanger | that's all you have? |
| 13:20.43 | phpboy | yeah |
| 13:21.01 | tzanger | ahh not latency= sorry latency\ (backslash and space) |
| 13:21.07 | tzanger | don't paste it here |
| 13:21.09 | tzanger | use pastebin |
| 13:21.41 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 13:26.36 | phpboy | tzanger: http://pastebin.ca/9472 |
| 13:27.05 | tzanger | actually there's nothing too bad there, but what all's on IRQ 12? cat /proc/interrupts |
| 13:29.20 | tzanger | what else is the asterisk system doing? |
| 13:29.36 | phpboy | http://pastebin.ca/9473 |
| 13:29.44 | phpboy | what do u mean? |
| 13:29.58 | tzanger | phpboy: why do you run ztdummy if you already ahve a timing source (wctdm) ? |
| 13:30.13 | tzanger | you also have your USB controllers on the same interrupt; do you use them ofr anything? |
| 13:30.36 | phpboy | now |
| 13:30.38 | phpboy | nope |
| 13:30.40 | phpboy | <PROTECTED> |
| 13:30.42 | tzanger | ok |
| 13:30.45 | tzanger | remove ztdummy |
| 13:30.50 | tzanger | and try it see if the chiping ocntinues |
| 13:31.05 | tzanger | what kidn of systme is it anyway (processor) |
| 13:31.07 | phpboy | how do I remove/disable ztdummy? |
| 13:31.13 | phpboy | PIII 733 |
| 13:31.16 | tzanger | rmmod ztdummy |
| 13:31.18 | tzanger | and odn't load it :-) |
| 13:32.47 | phpboy | I'm not all that clued up with Linux |
| 13:32.59 | phpboy | how do I stop a module from loading on boot |
| 13:33.06 | tzanger | I can tell. :-) But that's not a problem if you're willing to learn |
| 13:33.44 | CiNzAs | vi /etc/rc.d/init.d/zaptel |
| 13:34.05 | CiNzAs | and there you haver a line like this |
| 13:34.17 | tzanger | CiNzAs: only on some distros |
| 13:34.19 | CiNzAs | MODULES = "torisa tor ztdummy" |
| 13:34.20 | CiNzAs | etc |
| 13:34.24 | tzanger | CiNzAs: you can't assume everyone uses the same distro |
| 13:34.30 | CiNzAs | Ok |
| 13:34.34 | CiNzAs | What distro do u use ? |
| 13:34.34 | AvengerX | slack: /etc/rc.d/rc.modules |
| 13:34.39 | CiNzAs | whatever ... |
| 13:34.39 | tzanger | AvengerX: untrue |
| 13:34.40 | AvengerX | (or whenever kmod requests it) |
| 13:34.49 | newl | If I define an extension for say *78 (or some other internal extension) will asterisk use the one I define or the internal one first? |
| 13:34.50 | tzanger | slackware uses hotplug |
| 13:34.57 | tzanger | or rc.modules, or rc.local |
| 13:35.01 | tzanger | depends on how you have it set up |
| 13:35.06 | AvengerX | tzanger: normally, ok? |
| 13:35.45 | tzanger | AvengerX: heh. again it depends but rc.modules is a good place to look, although I'd be hard pressed to find an installation that puts it there since it's not a standard module |
| 13:35.56 | phpboy | #uname -a |
| 13:35.57 | phpboy | Linux asterisk1.local 2.4.21-27.0.1.EL #1 Fri Dec 24 02:04:03 GMT 2004 i686 i686 i386 GNU/Linux |
| 13:36.06 | tzanger | phpboy: what distro, not what kernel |
| 13:36.06 | phpboy | so how would I go about disabling it on boot? |
| 13:36.57 | phpboy | tzanger: I'm using asterisk at home on cent os |
| 13:37.00 | tzanger | ok |
| 13:37.05 | phpboy | I think it's a bread of Redhat |
| 13:37.06 | tzanger | that's what we needed to know |
| 13:37.11 | phpboy | ok, cool |
| 13:37.14 | tzanger | now we need someone who knows *@~ |
| 13:37.23 | phpboy | tzanger: it seems to have fixed the problem... |
| 13:37.33 | tzanger | phpboy: hey, I'm good, what can I say. :-0 |
| 13:37.40 | phpboy | tzanger: perhaps you can help me with another pressing issue |
| 13:37.42 | phpboy | thanks a mil |
| 13:37.43 | phpboy | u rock :D |
| 13:37.44 | phpboy | :P |
| 13:38.05 | tzanger | phpboy: try the asterisk@home pages to see if anyone describes how to disable that |
| 13:38.08 | phpboy | I add a SIP user |
| 13:38.12 | newl | oden: heh that kiax is pretty nifty |
| 13:38.18 | tzanger | kiax? |
| 13:38.31 | phpboy | and then an extention to contact that user |
| 13:38.38 | phpboy | but it doesn't seem to work all that well :/ |
| 13:38.52 | tzanger | phpboy: did you do an extensions reload and a sip reload? |
| 13:39.27 | newl | tzanger: kde iax client. |
| 13:39.33 | tzanger | newl: I guessed that much :-p |
| 13:39.39 | phpboy | I did an entire reload |
| 13:39.42 | newl | 8) |
| 13:39.55 | tzanger | well now that meetup.com's got their collective heads up their asses torastricon's gonna need another way to schedule meetings |
| 13:40.05 | tzanger | phpboy: hmm ok well "it doesn't work all that well" is very vague |
| 13:40.14 | phpboy | tzanger: doesn't work |
| 13:40.16 | phpboy | the SIP user |
| 13:40.17 | phpboy | works |
| 13:40.19 | phpboy | I can login |
| 13:40.21 | phpboy | the works |
| 13:40.34 | phpboy | but dialing that extention |
| 13:40.38 | phpboy | no can do :/ |
| 13:41.00 | tzanger | phpboy: well do you have an [extensions] context with all the extensions in it that is inlcuded from the default context of the sip users/ |
| 13:41.04 | tzanger | ? |
| 13:41.16 | phpboy | I do |
| 13:41.22 | tzanger | obviously not :-) |
| 13:41.41 | phpboy | pomple |
| 13:41.42 | phpboy | :/ |
| 13:41.58 | tzanger | what does the * console say when you try to dial the new extension? |
| 13:42.36 | phpboy | Not found |
| 13:42.37 | phpboy | :/ |
| 13:42.46 | tzanger | then you haven't created the extension correctly |
| 13:42.49 | phpboy | I'm clearly doing something VERY VERY stupid here |
| 13:42.49 | phpboy | :/ |
| 13:42.56 | tzanger | and I am not exactly sure what wrappers asterisk @ home uses so I am not much help |
| 13:43.07 | phpboy | tzanger: u know what would make you a HACKER of note and put me forever in ur debt |
| 13:43.13 | *** join/#asterisk malbech (Phils@m199.net81-66-243.noos.fr) |
| 13:43.20 | tzanger | phpboy: let me guess, if I figured this out for you |
| 13:43.24 | phpboy | is if you could quickly login to my box remotely |
| 13:43.26 | tzanger | that's not hacking, that's just helping out |
| 13:43.28 | phpboy | and just had a look |
| 13:43.30 | phpboy | I'll fix it |
| 13:43.32 | malbech | hello |
| 13:43.38 | tzanger | I am already a hacker of note. :-) |
| 13:43.40 | phpboy | I just need you to point out the problem |
| 13:43.45 | phpboy | IF you don't mind |
| 13:43.45 | phpboy | :P |
| 13:43.48 | phpboy | I know, i know :P |
| 13:43.50 | tzanger | phpboy: I'm trying to help you already |
| 13:43.58 | phpboy | I know :D |
| 13:43.59 | tzanger | what context do your sip users get dumped in to |
| 13:44.41 | phpboy | [ext-local] |
| 13:44.42 | *** join/#asterisk RyanW (~fuckyou@myjoint.id.au) |
| 13:44.45 | tzanger | ok |
| 13:44.47 | phpboy | in the extentions.conf file |
| 13:44.56 | tzanger | at the * console type show dialplan [ext-local] |
| 13:44.59 | tzanger | er without the [] |
| 13:45.05 | tzanger | does the extension that doesn't work show up in there? |
| 13:45.09 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 13:46.14 | RyanW | i'm trying to ring a mobile phone via zap and also a sip extension simultaneously and my problem is the Telco answers the zap channel immediately even if the mobile phone is unreachable. |
| 13:46.22 | RyanW | does anyone have a work around for this ? |
| 13:46.26 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 13:46.33 | Moonwick | heh. new mobile provider? |
| 13:46.41 | Moonwick | what do they do, answer and then play a ring noise? |
| 13:46.41 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 13:46.43 | phpboy | asterisk1*CLI> show dialplan [ext-local] |
| 13:46.43 | phpboy | There is no existence of '[ext-local]' context |
| 13:46.44 | phpboy | :< |
| 13:46.51 | boch | anyone here took the Cisco CVoice training? |
| 13:46.51 | RyanW | Moonwick...thats exactly what they do |
| 13:46.55 | phpboy | I've obviosly done something wrong :/ |
| 13:46.58 | Moonwick | that's really lame. |
| 13:47.04 | Moonwick | I don't know if there is much you can do |
| 13:47.20 | RyanW | Every telco in Australia does that |
| 13:47.38 | Moonwick | any idea why? |
| 13:47.43 | phpboy | tzanger: I've obviously done something very stupid :/ |
| 13:47.46 | RyanW | they answer then play silencem followed by a recorded message "phone out of range or switched off" or they play a ringing sound |
| 13:48.08 | Pj386 | phpboy: yeah, try reading everything tzanger said, he said "without the []" :) |
| 13:49.47 | phpboy | soz :/ |
| 13:49.49 | newl | "Please wait while this call is diverted..." |
| 13:49.53 | RyanW | is there a way i can make asterisk place a call then identify the tone being heard and act accordingly |
| 13:51.23 | phpboy | asterisk1*CLI> show dialplan ext-local |
| 13:51.24 | phpboy | [ Context 'ext-local' created by 'pbx_config' ] |
| 13:51.24 | phpboy | <PROTECTED> |
| 13:52.01 | tzanger | phpboy: ok so what is a sip extension that works? |
| 13:52.18 | tzanger | (sorry was on the phone) |
| 13:52.32 | phpboy | none of them do |
| 13:52.33 | phpboy | :/ |
| 13:52.39 | phpboy | I'm 300 |
| 13:52.43 | tzanger | ok |
| 13:52.44 | phpboy | so what I'm going to do |
| 13:52.50 | phpboy | is add 301 quickly |
| 13:52.54 | phpboy | at another place on my network |
| 13:52.57 | phpboy | and see if it works |
| 13:52.58 | tzanger | well that tells me that that macro is either not getting invoked correctly or it's the wrong macro altogether |
| 13:53.11 | oden | newl: cool. works ok? |
| 13:53.22 | newl | oden: ahhyeppers |
| 13:53.39 | tzanger | phpboy: just for shits and giggles, try Dial(SIP/blah), where blah is in sip.conf with [blah] and type=user or =friend |
| 13:53.54 | zoa | tzanger should be given an award today again :) |
| 13:54.00 | tzanger | I should? |
| 13:54.03 | newl | oden: though installing the libiax0 package porked my cvs Asterisk install..nothing a quick make install couldn't fix. |
| 13:54.06 | zoa | yeah |
| 13:54.08 | zoa | for helping out people |
| 13:54.09 | zoa | :) |
| 13:54.25 | tzanger | zoa haha yeah but I burn up the karma I accrue by flaming in the lists :-) |
| 13:54.31 | zoa | haha |
| 13:55.00 | zoa | i stopped doing free support, well for larger things like login in etc at least |
| 13:55.02 | *** join/#asterisk moy (~kvirc@201.135.105.124) |
| 13:55.14 | zoa | just cant keep spending all that time on it |
| 13:55.27 | zoa | houston we have a problem with the sip jitter buffer |
| 13:55.36 | tzanger | zoa: agreed. but I enjoy helping and I keep this and -dev in the background while I work |
| 13:55.40 | zoa | its taking too damn long |
| 13:55.53 | zoa | anyone willing to take over ? |
| 13:55.56 | tzanger | it gives me a healthy distraction and it (can) help others |
| 13:56.10 | tzanger | zoa there is nohting about the sip jitter buffer in -head, is there? |
| 13:56.25 | tzanger | jerjer had (numerous) sigsegv's on switch-3 which runs the new jitter buffer |
| 13:56.38 | zoa | aha he also had ? |
| 13:56.38 | *** join/#asterisk brc-tux (~brc-tux@pD9E9A160.dip0.t-ipconnect.de) |
| 13:56.46 | zoa | we have some crashes on the sip jitter buffer |
| 13:56.53 | zoa | once every some hundred thousand calls |
| 13:57.15 | tzanger | zoa: yes but it the code in -HEAD? I thought the sip jitter buffer was not merged |
| 13:57.23 | zoa | no its not in there |
| 13:57.32 | zoa | the last version is only here in the office |
| 13:57.41 | zoa | there is a patch for the last approach on mantis |
| 13:57.48 | zoa | most recent version doesnt even compile for now |
| 13:58.00 | phpboy | tzanger: I've added 301 onto my network now |
| 13:58.02 | *** join/#asterisk ajx (~root@46-80.200-68.tampabay.res.rr.com) |
| 13:58.07 | phpboy | we still can't seem to contact one another |
| 13:58.10 | phpboy | "Not found" |
| 13:58.11 | phpboy | :/ |
| 13:58.22 | tzanger | zoa: ahh okay... jerjer's running -HEAD from yesterday when he got it |
| 13:58.27 | tzanger | phpboy: did you do what I asked? |
| 13:58.34 | tzanger | what's your SIP user in sip.conf |
| 13:58.44 | phpboy | My personal one is 300 |
| 13:59.02 | phpboy | and the new user is 301 on the network |
| 13:59.03 | tzanger | right |
| 13:59.07 | tzanger | what is it in sip.conf |
| 13:59.08 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 13:59.08 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:59.11 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 13:59.15 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 13:59.15 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:59.19 | phpboy | 300 |
| 13:59.40 | RoyK | 400 |
| 13:59.48 | bkw_ | 500 |
| 13:59.54 | tzanger | phpboy: you have a [300] in sip.conf? |
| 13:59.55 | RoyK | 42 |
| 14:00.06 | tzanger | with type=user or type=friend? |
| 14:00.08 | phpboy | I do |
| 14:00.21 | tzanger | ok I wonder if that's allowed |
| 14:00.26 | tzanger | Dial(SIP/300) |
| 14:00.39 | phpboy | type=friend |
| 14:00.39 | tzanger | I would have thought that * would try and interpret '300' as in IP and fail |
| 14:00.58 | phpboy | nope |
| 14:01.04 | phpboy | I authenticate |
| 14:01.05 | phpboy | just fine |
| 14:01.07 | phpboy | etc.. |
| 14:01.11 | tzanger | I'm ont talking about authentication, I'm talking about dialing |
| 14:01.13 | phpboy | Dial(SIP/300) |
| 14:01.18 | phpboy | won't work on my console |
| 14:01.20 | phpboy | :/ |
| 14:01.26 | phpboy | na, I can dial fine with it |
| 14:01.28 | tzanger | I didn't say on the console |
| 14:01.32 | tzanger | phpboy: that is outgoing |
| 14:01.34 | tzanger | that's totally different |
| 14:01.40 | phpboy | soz :< |
| 14:01.50 | phpboy | I dial 300 |
| 14:01.52 | tzanger | replace that exten => 300,1,Macro() with exten => 300,1,Dial(SIP/300) |
| 14:01.54 | phpboy | from my softphone |
| 14:01.54 | tzanger | extensions 300 |
| 14:01.57 | phpboy | ahhh |
| 14:01.58 | phpboy | soz |
| 14:01.59 | tzanger | and try dialing 300 from the soft phone |
| 14:02.58 | phpboy | nope |
| 14:02.59 | phpboy | still |
| 14:03.02 | phpboy | "Not found" |
| 14:03.03 | *** join/#asterisk SirPrize (~blah@host-212-158-241-184.bulldogdsl.com) |
| 14:03.08 | tzanger | ok do this |
| 14:03.14 | tzanger | exten => 300,1,NoOp(I am trying to dial 300) |
| 14:03.19 | tzanger | exten => 300,2,Dial(SIP/300) |
| 14:03.23 | tzanger | extensions reload |
| 14:03.24 | tzanger | and try again |
| 14:03.29 | SirPrize | how could I make asterisk only make one outgoing call per available SIP account ? |
| 14:04.17 | SirPrize | I've tried using ChanIsAvailable("SIP/outgoingaccount"), but this always says there ARE available channels, even when I'm using the outgoing account for a call |
| 14:04.25 | phpboy | nope |
| 14:04.28 | phpboy | still not |
| 14:04.51 | phpboy | perhaps I should paste you my debug? |
| 14:04.52 | [TK]D-Fender | phpboy : could you put your sip.conf and extensions.conf in a pastebin for us |
| 14:04.59 | [TK]D-Fender | would make things a lot easier |
| 14:05.08 | bjohnson | SirPrize: setgroup and checkgroup .. look at the superdial macro on the wiki |
| 14:05.08 | tzanger | phpboy: do you see the NoOp() text on the console? |
| 14:05.15 | phpboy | [TK]D-Fender: good idea |
| 14:05.18 | tzanger | [TK]D-Fender: nah I like doing it this way |
| 14:05.20 | tzanger | he's learning more |
| 14:05.25 | RyanW | SirPrize use the hangup extension to place the next outgoing call in the spool folder |
| 14:05.27 | phpboy | that's also true |
| 14:05.35 | phpboy | tzafrir: I had sip debug running |
| 14:05.39 | phpboy | lemme disable quick |
| 14:05.44 | tzanger | sip no debug |
| 14:05.54 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 14:05.56 | [TK]D-Fender | tzanger = Proponent of masochism ;) |
| 14:06.04 | SirPrize | bjohnson: thanks for that - will read and see |
| 14:06.26 | tzanger | [TK]D-Fender: nonsense; it's easy to just point out what's wrong when you have the configs but he's willing to learn and I am trying to cultivate that; there are far too many people already that want it the easy way |
| 14:06.47 | SirPrize | RyanW: thanks, but I think you're trying to queue the calls to take place one after the other? It's enough if I can identify if there's a call already in progress and just play a recording that the channel isn't available |
| 14:06.48 | phpboy | tzanger: I do not :< |
| 14:07.02 | tzanger | phpboy: you do not see NoOp(blah blah blah) when you dial 300? |
| 14:07.12 | phpboy | I prefer learning, the hard way |
| 14:07.17 | phpboy | that way I won't forget :D |
| 14:07.19 | tzanger | then your softphone does not start out in the [ext-local] context |
| 14:07.24 | *** part/#asterisk brc-tux (~brc-tux@pD9E9A160.dip0.t-ipconnect.de) |
| 14:07.29 | [TK]D-Fender | I suppose. its a mixed blessing of sorts. Depends how long you can be frustrated trying to help when it turns out to be the absolute last thing you'd look at |
| 14:07.30 | phpboy | tzafrir: I'm search my debug for it |
| 14:07.32 | phpboy | non |
| 14:07.33 | phpboy | :/ |
| 14:07.40 | tzanger | [TK]D-Fender: this isn't frustrating at all |
| 14:07.41 | *** join/#asterisk quigleymd (~quigleymd@24-53-142-5.chvlva.adelphia.net) |
| 14:07.43 | tzanger | at least not at the moment |
| 14:07.46 | tzanger | we're not going in circles |
| 14:07.47 | [TK]D-Fender | heh |
| 14:07.56 | [TK]D-Fender | thats a bonus ;) |
| 14:08.11 | phpboy | tzanger: what are we looking at next then |
| 14:08.12 | phpboy | ? |
| 14:08.27 | RoyK | ~docs |
| 14:08.28 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:08.32 | tzanger | your soft phone sip.conf entry, what's its context= line |
| 14:08.36 | tzanger | ~royk |
| 14:08.37 | jbot | it has been said that royk is my one and only sex toy |
| 14:08.42 | sivana | heh |
| 14:09.04 | tzanger | heh |
| 14:09.08 | RoyK | ~tzanger |
| 14:09.09 | jbot | it has been said that tzanger is some #asterisk resident, although he doesn't know too much... |
| 14:09.10 | tzanger | I forgot about that one |
| 14:09.29 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 14:09.33 | Nugget | heh |
| 14:09.38 | RoyK | jbot: no, RoyK is that nice Asterisk consultant from .no :P |
| 14:09.39 | jbot | okay, RoyK |
| 14:09.50 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 14:09.51 | phpboy | context = ext-local |
| 14:09.53 | *** part/#asterisk SirPrize (~blah@host-212-158-241-184.bulldogdsl.com) |
| 14:09.57 | phpboy | in my sip.conf file |
| 14:10.06 | tzanger | phpboy: for your soft phone entry? |
| 14:10.09 | sivana | ~royk |
| 14:10.12 | jbot | somebody said royk was that nice Asterisk consultant from .no :P |
| 14:10.12 | tzanger | what is your softphone's entry |
| 14:10.15 | sivana | blah |
| 14:10.16 | sivana | heh |
| 14:10.23 | phpboy | tzanger: what do you mean by that? |
| 14:10.51 | tzanger | phpboy: the softphone entry in sip.conf |
| 14:10.52 | phpboy | ah, I see what u mean |
| 14:10.53 | newl | hot grits haha |
| 14:10.54 | phpboy | hang ten... |
| 14:10.54 | tzanger | what is its [name] |
| 14:11.05 | Silik0n | <PROTECTED> |
| 14:11.46 | *** join/#asterisk coppice (~chatzilla@235.193.17.210.dyn.pacific.net.hk) |
| 14:11.53 | phpboy | tzanger |
| 14:11.59 | phpboy | it was the wrong context |
| 14:12.02 | phpboy | I just learnt something |
| 14:12.07 | phpboy | that I WILL NOT forget :D |
| 14:12.17 | *** join/#asterisk jmacz (~jmacz@63.245.86.104) |
| 14:12.23 | RoyK | coppice: guten abend, herr coppice |
| 14:12.32 | tzanger | guten abend? |
| 14:12.48 | coppice | it means "good crash" |
| 14:12.54 | tzanger | guten morgen, guten nacht... abend? |
| 14:12.55 | mmlj4 | heh |
| 14:13.05 | mmlj4 | abend = evening |
| 14:13.09 | tzanger | ahh |
| 14:13.23 | coppice | abend == abnormal ending |
| 14:13.30 | RoyK | coppice: remember those old novell servers ABENDing |
| 14:13.32 | RoyK | shit |
| 14:13.44 | *** part/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
| 14:13.45 | coppice | IBM did it first :-) |
| 14:13.50 | RoyK | I know |
| 14:14.05 | phpboy | tzanger: last Q |
| 14:14.12 | phpboy | before I continue through these DOCS |
| 14:14.21 | phpboy | how do I make it |
| 14:14.26 | phpboy | that after 5 rings |
| 14:14.30 | phpboy | it goes to my mailbox? |
| 14:14.52 | phpboy | voicemail |
| 14:14.54 | phpboy | that is |
| 14:14.56 | RoyK | coppice: did you get to try that firmware yet? |
| 14:15.06 | phpboy | I think I've added the voice mail box properly |
| 14:15.07 | RoyK | phpboy: it's all in the docs |
| 14:15.11 | RoyK | phpboy: all of it |
| 14:15.13 | phpboy | now all I need is the redirect |
| 14:15.17 | tzanger | phpboy: well that's all in that macro |
| 14:15.17 | RyanW | phpboy......go read the macro vmdial ... |
| 14:15.20 | RoyK | phpboy: rtfm |
| 14:15.22 | tzanger | look at the macro and see what it does |
| 14:15.24 | coppice | RoyK: yep. it seems little different from the previous version |
| 14:15.25 | RoyK | ~rtfm |
| 14:15.26 | jbot | extra, extra, read all about it, rtfm is read the f*cking manual... try asking me about "FAQ" |
| 14:15.26 | tzanger | RoyK: he's doing very well |
| 14:15.32 | phpboy | RoyK: I'm reading on www.asteriskdocs.org |
| 14:15.36 | phpboy | am I at the right place? |
| 14:15.36 | RoyK | coppice: tzangerok... |
| 14:15.37 | tzanger | that's blitzrage's site |
| 14:15.45 | RyanW | phpboy look in the default extensions.conf there is a macro called vmdial |
| 14:15.46 | sivana | ~faq |
| 14:15.47 | RoyK | coppice: meaning just as bad signal etc? |
| 14:16.00 | Chad-wl | Why is x-ten getting a 404 error trying to dial local on a default asterisk configuration? I'm on the console with -vvvvv and I still don't see any errors. How can I tell what is failing? |
| 14:16.12 | phpboy | RoyK: no need to get hectic d00d |
| 14:16.14 | coppice | yep. the fax machine either marginally decode or fail |
| 14:16.18 | phpboy | we're all here to learn/teach |
| 14:16.24 | phpboy | if it was ONLY up to the docs |
| 14:16.29 | phpboy | we wouldn't need this channel |
| 14:16.29 | phpboy | :P |
| 14:16.39 | RyanW | phpboy. you have not read the docs. you are lazy. |
| 14:16.54 | phpboy | RyanW: I got the system going via the docs |
| 14:16.55 | phpboy | :P |
| 14:16.59 | phpboy | now I need to relax a bit |
| 14:17.00 | phpboy | :PP |
| 14:17.01 | newl | Chad-wl: are you sure the client is registered? |
| 14:17.04 | RoyK | coppice: could you send an email to yoda about it, please? |
| 14:17.12 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 14:17.14 | coppice | OK |
| 14:17.20 | RoyK | coppice: and I'll see if I can ship down that 4port micronet |
| 14:17.37 | Chad-wl | newl: It's a SIP client registering fine, I can call out and access the main menu (s context) is there another place to register? |
| 14:17.38 | phpboy | tzanger: www.asteriskdocs.org |
| 14:17.42 | phpboy | got doc source? |
| 14:17.56 | phpboy | good doc source |
| 14:17.56 | phpboy | I mean |
| 14:18.27 | tzanger | yes blitzrage et al have put a lot of effort into that |
| 14:18.41 | tzanger | I keep meaning ot help them out with it but you know what they say about good intentions |
| 14:18.42 | RyanW | phpboy. go configure the voicemail accounts http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 14:18.56 | newl | Chad-wl: sounds like it's registering fine then. Make sure the clients sip.conf entry points to the correct context. |
| 14:19.15 | Chad-wl | newl: what context should it point too exactly? |
| 14:19.16 | phpboy | RyanW: in your opinion... what's the best source for asterisk DOCs overall? |
| 14:19.23 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 14:19.35 | RyanW | phpboy....http://voip-info.org |
| 14:20.20 | newl | Chad-wl: whichever you decide that client should have access to. e.g. all my test bed clients here use my test context. |
| 14:20.23 | *** join/#asterisk Corydon76-home (black@pcp08665860pcs.500ash01.tn.comcast.net) |
| 14:20.29 | phpboy | RyanW: thanks : |
| 14:20.31 | phpboy | :) |
| 14:21.39 | Chad-wl | newl: I'd like to have them access the sample "from-sip" context. Where is that context defiend? |
| 14:23.12 | tzanger | RyanW: don't give him the wiki... jeez that'll throw him into a tailspin |
| 14:23.31 | newl | Chad-wl: try the local context (presuming you're using the default example configurations) |
| 14:23.51 | newl | or demo |
| 14:23.51 | *** join/#asterisk sergiovel (~sergio@200.68.89.177) |
| 14:23.55 | RyanW | tzanger...if he goes and reads the wiki asterisk configuration from the "start here" bit onwards he'll do just fine |
| 14:24.03 | sergiovel | Hello everyone |
| 14:24.18 | Godsey | is there an elegant way to use something like stdexten macro but have the DIAL command include W or possibly wW depending on if the call originated from a stdext? |
| 14:24.21 | tzanger | RyanW: I dunno, I have found the wiki to be a wealth of (very disorganized, often stale and many times conflicting) information |
| 14:24.33 | DrWho17 | tzanger: exactly so |
| 14:24.47 | tzanger | which is a problem with wikis in general |
| 14:25.06 | tzanger | they need a flock of maintainers to keep it coherent |
| 14:25.07 | RyanW | i work for an ISP managing 3 asterisk boxes and 100 extensions in 2 countries and the wiki has taught me how to compile/configure and manage asterisk |
| 14:25.11 | DrWho17 | that is why I try to help if someone ask a question on the irc support |
| 14:25.18 | DrWho17 | rather then reference to a wiki |
| 14:25.19 | Godsey | I was thinking GotoIf[${LEN{CALLERIDNUM}}==4] |
| 14:25.32 | Godsey | but haven't done it yet thinking someone here may have a good idea :) |
| 14:25.45 | tzanger | Godsey: that is right |
| 14:25.45 | Chad-wl | newl: Thanks, I got it to connect, |
| 14:25.46 | tzanger | I do that |
| 14:25.59 | Godsey | RyanW: neat :) I manage 3 asterisk machines too but only 30 ext in an isp |
| 14:26.21 | newl | Chad-wl: cool |
| 14:26.51 | RyanW | we have a $50 000 Avaya PABX sitting in a pile in the kitchen at work because Asterisk shits all over it when it comes to reliability and functionality. |
| 14:26.55 | Godsey | the ISP wants to start providing voip to customers |
| 14:26.57 | sergiovel | I have a question guys, is there any webphone or softphone that will work behind a firewall. I have tried firefly and others without sucess. I want to install it in a company like IBM, HP, etc that usually only have port 80 open. I know that Skype works in this company...any suggestions? Most users are on Windows |
| 14:27.12 | oden | newl: yes, i see. i added the new jitterbuffer code to it from the iaxclient stuff. |
| 14:27.21 | Godsey | I'm having admin overload :) |
| 14:27.28 | CiNzAs | sergiovel: no luck |
| 14:27.30 | Godsey | I manage all work desktops, servers, routers |
| 14:27.35 | Godsey | and now asterisk :) |
| 14:27.35 | CiNzAs | skype is peer-to-peer |
| 14:27.38 | DrWho17 | sergiovel: the sip clients/iax clients should work fine |
| 14:27.45 | Godsey | there is almost too much for 1 person to do anymore |
| 14:27.46 | RyanW | Godsey. what company do you work for? |
| 14:27.52 | CiNzAs | DrWho17: i doubt |
| 14:27.54 | Godsey | RyanW: local isp near Seattle |
| 14:27.57 | RyanW | Godsey. i work fot http://www.tsninternet.com.au |
| 14:27.59 | sergiovel | i have tried but did not work |
| 14:28.04 | *** part/#asterisk JohnnyST ([U2FsdGVkX@av8.netikka.fi) |
| 14:28.07 | Godsey | http://www.fidalgo.net/ |
| 14:28.19 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfjsm.dialup.mindspring.com) |
| 14:28.34 | CiNzAs | If they block udp ports |
| 14:28.46 | sergiovel | how does skype manage to make it work |
| 14:28.56 | DrWho17 | oh, yes firewalling is a problem, missed that part |
| 14:29.04 | CiNzAs | Skype is a peer-to-peer |
| 14:29.06 | DrWho17 | of course the ports need to be open |
| 14:29.06 | newl | using a tunneling client (such as freedom or stunnel) on the local machine and connecting to that should allow anyone through a proxy. :) |
| 14:29.13 | sergiovel | this company has a firewall that cannot be administered |
| 14:29.20 | sergiovel | easily |
| 14:29.20 | CiNzAs | newl: maybe |
| 14:29.29 | Godsey | sergiovel: IAX :) |
| 14:29.36 | newl | CiNzAs: right, in theory anyway. :) |
| 14:29.45 | CiNzAs | Hehe |
| 14:29.46 | sergiovel | firefly using iax did not pass :( |
| 14:29.52 | Godsey | no |
| 14:29.56 | Godsey | put asterisk behind the firewall |
| 14:30.02 | CiNzAs | Maybe they have a ISA server or eles |
| 14:30.07 | sergiovel | let me expand a bit then... |
| 14:30.08 | RyanW | Godsey, i'm not in irc much but i do use msn messenger if you wanna pm me i'll give it to you, feel free to ask for advice with asterisk if u need to |
| 14:30.16 | Godsey | talking to outside world (or other asterisk machine) via iax |
| 14:30.18 | CiNzAs | i can't even do ssh tunnelling |
| 14:30.22 | Godsey | and sip internally to asterisk |
| 14:30.29 | sergiovel | the server has to be outside their company for a conferencing solution |
| 14:30.42 | sergiovel | so 20 people need to access it from diferent companies |
| 14:30.53 | sergiovel | usually this big companies with firewall |
| 14:30.57 | Godsey | and? |
| 14:31.02 | RyanW | Godsey...or routing/bgp/linux/java/perl/asp for that matter. |
| 14:31.04 | sergiovel | they all need to access the asterisk box |
| 14:31.05 | Godsey | use hub and spoke topography |
| 14:31.20 | sergiovel | using a web or softphone |
| 14:31.57 | DrWho17 | then they need to allow the service through their firewall |
| 14:32.09 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 14:32.20 | CiNzAs | Yeap that's right |
| 14:32.38 | *** join/#asterisk Eranpe (~eran_perl@104.Red-81-34-236.pooles.rima-tde.net) |
| 14:32.42 | CiNzAs | Ask the firewall addmin to open SIP ports |
| 14:32.48 | sergiovel | that is going to be very difficult |
| 14:32.50 | CiNzAs | During the conference |
| 14:32.58 | CiNzAs | Otherwise .. i doubt. |
| 14:33.00 | sergiovel | because I have to ask 20 companies to do that |
| 14:33.09 | CiNzAs | And if those are "big" companies ... i reall y doubt |
| 14:33.10 | sergiovel | and the companies change everytime |
| 14:33.34 | sergiovel | cause this is a guy who will send an email inviting people to join the conference |
| 14:33.47 | DrWho17 | oh, well it would be easier then getting each of those companies to have some custom configged client |
| 14:33.48 | sergiovel | he want them to listen to them |
| 14:33.54 | DrWho17 | or tunnelling solution |
| 14:34.25 | CiNzAs | Well .... Theres is other way |
| 14:34.25 | CiNzAs | lol |
| 14:34.39 | sergiovel | how does skype work? they work over the 80 port? |
| 14:34.54 | CiNzAs | If they cant'use the internet connection ... The client use the traditional phone system |
| 14:35.22 | sergiovel | but the cost goes way up, i did the numbers |
| 14:36.13 | DrWho17 | sergiovel: well it tries a few things I think |
| 14:36.20 | DrWho17 | any, then 443, then 80 |
| 14:36.48 | sergiovel | that is nice...but we cant use skype with asterisk right? |
| 14:37.14 | CiNzAs | The minimum requirement is that Skype needs unrestricted outgoing TCP access to all destination ports above 1024 or to ports 80 and 443 (the former is better, however). If you don't allow either of those, Skype will not work reliably at all. |
| 14:37.19 | CiNzAs | from skype.com |
| 14:37.28 | DrWho17 | can you run a sip proxy on port 80 maybe? |
| 14:37.42 | DrWho17 | point their sip clients to it, and have it talk to asterisk |
| 14:37.44 | sergiovel | you mean at each customer? |
| 14:37.52 | DrWho17 | sergiovel: no on the outside |
| 14:37.59 | sergiovel | ah..that is a good idea |
| 14:38.04 | CiNzAs | Hmmmm |
| 14:38.12 | DrWho17 | I'm not sure how the rtp would work though |
| 14:38.13 | CiNzAs | Sip cliente ---> Port 80 --> SIP PROXY ---> * |
| 14:38.20 | DrWho17 | CiNzAs: yea |
| 14:38.22 | CiNzAs | Yeap ! |
| 14:38.27 | CiNzAs | Hmmmmm |
| 14:38.47 | coppice | what works is what is not blocked |
| 14:38.49 | coppice | what is not blocks varies enormously :-) |
| 14:39.03 | DrWho17 | right, and it's variable on every company |
| 14:39.03 | fenlander | Often these companies don't really have port 80 open - it is just an http proxy, so a tunnel won't work |
| 14:39.16 | DrWho17 | right, that too |
| 14:39.19 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.res.rr.com) |
| 14:39.26 | sergiovel | hmm |
| 14:39.40 | DrWho17 | fenlander: although some of the streaming audio needs to go straight out, and company executives like that |
| 14:39.48 | coppice | I think most big companies proxy port 80 |
| 14:39.50 | DrWho17 | so many firewalls have holes punched in them for htis |
| 14:40.24 | fenlander | Yes- but that is often handled directly by the proxy understanding the streaming protocol |
| 14:40.54 | sergiovel | now the fact that skype works well in this one company that has 60 offices all over the world...I guess it shows that it can happen |
| 14:41.10 | phpboy | tzanger: I must be over looking something |
| 14:41.18 | fenlander | sergiovel: yes - worth a try |
| 14:41.19 | phpboy | I can't see in that Doc that ryan pasted me |
| 14:41.31 | sergiovel | so the sip proxy idea might just work |
| 14:41.34 | phpboy | how to make it go to voice mail after what ever number of rings |
| 14:42.04 | fenlander | sergiovel: you need to proxy the rtp traffic as well |
| 14:42.22 | sergiovel | it would be smth like ... softclient >>Port 80 >>SER>>Asterisk |
| 14:42.30 | coppice | does skype do anything to mitigate lost packets? |
| 14:42.31 | DrWho17 | phpboy: unavailable |
| 14:42.52 | phpboy | DrWho17 ? |
| 14:42.56 | fenlander | sergiovel: and this is TCP port 80, not udp |
| 14:43.02 | sergiovel | correct |
| 14:43.05 | sergiovel | yes tcp |
| 14:43.13 | DrWho17 | phpboy: add an extension for it, the priority after the dial |
| 14:43.36 | phpboy | yeah, I thought so |
| 14:43.44 | phpboy | but how do you specify the ammount of rings? |
| 14:43.52 | *** join/#asterisk bratner (~kvirc@bzq-179-152-71.pop.bezeqint.net) |
| 14:44.15 | sergiovel | Ok, I will give that setup a try...thanks guys for your feedback |
| 14:45.21 | BuckRogers | f |
| 14:45.26 | Chad-wl | When I call out from a sip client it reports: Called 1/9056235555 Does that mean that it dialed a 1/ in the number? |
| 14:45.35 | BuckRogers | good morning |
| 14:45.36 | *** join/#asterisk jeffik (jefik@69.158.26.125) |
| 14:46.22 | jeffik | need help getting x-lite to work |
| 14:46.53 | BuckRogers | x-lite there is a lot that you need to leave blank' |
| 14:46.54 | mozrat | Any Debian Testing users here?? Who might know why I get [chan_modem_bestdata.so]/usr/lib/asterisk/modules/chan_modem_bestdata.so: undefined symbol: ast_unregister_modem_driver |
| 14:49.45 | zoa | http://www.asteriskguru.com/xlite.html# |
| 14:49.49 | zoa | for xlite setup |
| 14:49.49 | BuckRogers | phpboy: have you worked with mysql databases? |
| 14:50.02 | *** join/#asterisk delYsid (~user@delYsid.developer.debian) |
| 14:50.41 | smiley- | speaking of softphones.. I can't get the softphones to match an extensions with #nn# *nn* works.. but no #nn# can't figure out why |
| 14:50.51 | delYsid | Does anyone know docs which describe how to use asterisk just as a SIP proxy to connect one cisco phone to a public SIP gateway? |
| 14:51.29 | zoa | mozrat: comment it in modules.conf |
| 14:51.41 | zoa | delysid, asterisk is not a proxy |
| 14:51.55 | zoa | but the thing you want can be done easilt |
| 14:52.02 | foobos | delysid, if you need SIP proxy, use SER |
| 14:52.08 | zoa | but dunno for a place to look for |
| 14:52.59 | delYsid | hrm |
| 14:53.11 | delYsid | foobos: SER? |
| 14:53.14 | foobos | delysid, http://www.voip-info.org/tiki-index.php?page=SIP+Express+Router |
| 14:53.33 | newl | heh ser would be a bit overkill |
| 14:54.07 | foobos | well if you don't need the pbx, then why configure one |
| 14:55.01 | newl | don't load the module then. |
| 14:55.36 | boch | anyone here took the Cisco CVoice training? |
| 14:55.51 | *** join/#asterisk Romik (~romik@router-net.ser.netvision.net.il) |
| 14:57.31 | mozrat | zoa, trying that now, thanks |
| 14:57.45 | *** part/#asterisk RyanW (~fuckyou@myjoint.id.au) |
| 14:59.15 | Essobi | I'm loving this new NiN Downward Spiral Remaster. |
| 14:59.34 | Essobi | I think I'm going to whip up some new MOH. :) |
| 15:00.48 | *** join/#asterisk mwgbc (mwallacegb@adsl-69-109-116-236.dsl.pltn13.pacbell.net) |
| 15:01.28 | mwgbc | Will MoH play if you flash over to dial a three way call? |
| 15:02.14 | ChkDigit | mwgbc: It should play for the party that you flashed into the background. |
| 15:03.55 | mwgbc | ChkDigit: Thanks that's what I needed to know. |
| 15:03.58 | *** part/#asterisk mwgbc (mwallacegb@adsl-69-109-116-236.dsl.pltn13.pacbell.net) |
| 15:04.27 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 15:07.31 | bkw_ | asdf |
| 15:09.18 | Essobi | mwg I believe so. |
| 15:10.01 | Essobi | It does on my 7960s. |
| 15:10.15 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 15:10.20 | Essobi | Never used an FXS device. |
| 15:13.55 | CiNzAs | Anyone using CM 4.x with asterisk ? |
| 15:14.01 | CiNzAs | SIP Trunk |
| 15:14.09 | sudhir492 | this may not be related to asterisk, but with so many unix experts here, someone might answer my question. My Asterisk server restarts for some reason. In the syslog, I see syslogd 1.4.1: restart. |
| 15:14.13 | CiNzAs | My SIp trunk has gone crazy right now .... grrr... |
| 15:14.31 | CiNzAs | That is no asterisk restarting |
| 15:14.36 | CiNzAs | thar is syslogd restarting |
| 15:14.48 | sudhir492 | Where to find out more about the casue of this. |
| 15:14.50 | Moc____ | why are all my International keep broking all the time !!!! |
| 15:14.57 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 15:15.12 | sudhir492 | I know, it is not asterisk, it is machine rebooting |
| 15:15.14 | Moc____ | the last working provider I have force me to use g729 |
| 15:15.16 | bprice20 | does anyone have an agi for implementing callback when busy |
| 15:15.56 | bprice20 | so when a user calls a number and its busy it will keep calling back until there not busy then dial the user back |
| 15:16.10 | sudhir492 | bpriec20: I wrote an AGI to callback after capturing callerid. You can use it after busy |
| 15:16.15 | Nugget | bprice20: that's really difficult to do, since it's hard for asterisk to know the difference between "busy" and "ringing" |
| 15:16.36 | Nugget | (for many channel types) |
| 15:16.49 | Moc____ | anyone can dial with voiceconduits internationally ? |
| 15:17.01 | bprice20 | sudir492 that may work will you send it to me? |
| 15:17.22 | bprice20 | nugget really? |
| 15:17.27 | sudhir492 | CiNzAs: Yes. My machine is restarting for some reason. How to figure out why is it rebooting? |
| 15:17.39 | Nugget | yes, really! |
| 15:17.42 | CiNzAs | check syslog |
| 15:19.33 | sudhir492 | CiNzAs: Thats what I am doing. In my syslog, I see message at 15:11, and then at 15:50 syslogd 1.4.1: restart |
| 15:19.51 | bprice20 | I'd imagine when it tosses back congestion thats where I would pick that up, play the user a message give them an option to retry and run the agi |
| 15:20.24 | *** join/#asterisk fbw (~tyco@c-67-181-56-5.hsd1.ca.comcast.net) |
| 15:20.32 | Moc____ | ok jerjer is not here, file wont talk to me !!! that suck.. |
| 15:21.18 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 15:21.18 | *** mode/#asterisk [+o twisted] by ChanServ |
| 15:22.33 | CiNzAs | sudhir492: cant tell |
| 15:22.37 | sudhir492 | ChkDigit: Haha. I wish it was that simple :-) Then I could knock on someone's head and fix the problem. No, for some weird reason the machine reboots itself. I dont know where to find the cause of this |
| 15:22.42 | CiNzAs | you have to check that |
| 15:24.10 | *** join/#asterisk goobster (goobster@c-67-168-105-166.hsd1.wa.comcast.net) |
| 15:24.56 | *** join/#asterisk ellvis (~root@195.98.29.34) |
| 15:25.03 | bprice20 | sudhir492 try /var/log/acpid might gve you some clues |
| 15:25.03 | ellvis | hi people |
| 15:25.34 | bprice20 | maybe someone accidentally set up a cron |
| 15:25.53 | fbw | Is there a way to test the server using a pots modem to dial in or out, no voip |
| 15:26.40 | bprice20 | yes you need fxo, or was it fxs hardware |
| 15:27.13 | bprice20 | sudhir492-- get that e-mail addy? |
| 15:27.30 | DrWho17 | sudhir492: right, add the kernel options noapic pci=noacpi acpi=off |
| 15:27.46 | DrWho17 | if you use linux and are getting hardlocks |
| 15:27.51 | Nugget | fxo talks TO a dialtone. FXS creats a dialtone. |
| 15:27.58 | sudhir492 | bprice20: Thats the first thing I checked. No entry in crontab. in /var/log/acpid all I see is at 15:08 1 rule loaded, and at 15:51 starting up |
| 15:28.27 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
| 15:28.29 | CiNzAs | damm it |
| 15:28.35 | sudhir492 | bprice20: yes, go the email address |
| 15:28.36 | CiNzAs | i do a weekly restart to asterisk |
| 15:28.42 | CiNzAs | Everytime i do that restart |
| 15:28.49 | CiNzAs | MPG123 goes to 100% cpu |
| 15:29.04 | CiNzAs | just one process, and only when i do a restart |
| 15:29.29 | bprice20 | CiNzAs I had the same thing for a while I just added killall -9 mpg123 to the cron |
| 15:29.38 | CiNzAs | thats not a solution |
| 15:30.05 | DrWho17 | haha |
| 15:30.51 | jero-aw | lol |
| 15:30.51 | bprice20 | well it IS A solution maybe not a good one |
| 15:30.58 | CiNzAs | Hehe .. or that |
| 15:31.20 | CiNzAs | Maybe ill do that. |
| 15:32.43 | bprice20 | Wyhos an asterisk consultant I need a feature, don't know how to develop it and am willing to pay for expertise |
| 15:32.52 | bprice20 | Whois |
| 15:33.03 | tzanger | bprice20: we're all asterisk consultants to one degree or another |
| 15:33.09 | tzanger | what are you looking for |
| 15:33.13 | jero | :) |
| 15:33.31 | bprice20 | I need a feature that will callback on busy |
| 15:34.14 | bprice20 | So asterisk gets busy, plays back a message user presses something to confirm then asterisk calls every 3 minutes and when its not busy calls back user |
| 15:35.13 | tzanger | bprice20: hmm |
| 15:35.17 | Essobi | bprice20 someone wrote a busy redial app I think. |
| 15:35.26 | tzanger | that would be tricky |
| 15:35.32 | Essobi | I seen it mentioned somewhere before.. maybe the wiki. |
| 15:35.37 | bprice20 | after some googling I apparently am not the only person looking for it. I'll pay someone for their time then post it online in voip-info |
| 15:35.39 | tzanger | because you can't tell if it's busy until you Dial() |
| 15:35.40 | Essobi | It was a dirty hack if I remember right. |
| 15:35.49 | *** join/#asterisk easimon (~easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 15:35.50 | tzanger | and you don't want to bridge until Dial() says NOT busy |
| 15:36.03 | tzanger | I can think of a few ways to try |
| 15:36.22 | Essobi | Umm. |
| 15:36.39 | Essobi | exten => s-BUSY,1,answer() |
| 15:36.59 | Essobi | exten => s-BUSY,2,AGI(dial-back-i-wrote) |
| 15:37.02 | Nugget | sometimes you can't tell if it's busy even after you dial. |
| 15:37.03 | bprice20 | tru you can't tell if its busy until you dial but when you get Congestion (forgot the sip code) you can run the agi or whatever |
| 15:37.10 | Nugget | I don't get BUSY back from my FXO |
| 15:37.28 | Essobi | Nugget You need a better FXO then. |
| 15:37.30 | Essobi | :) |
| 15:37.34 | bprice20 | I get busy from my sip provider |
| 15:37.43 | Nugget | I just get bridged to the PSTN busy tone. |
| 15:37.44 | bprice20 | Essobi can I have that? |
| 15:37.47 | Essobi | Ayup.. sip supports it. |
| 15:38.13 | tzanger | actually I think I have a not so dirty trick |
| 15:38.16 | Essobi | bprice20 lol.. I was being silly.. that's just an easy way to do it.. write an AGI to perform the call backs and you're golden. |
| 15:38.29 | Essobi | tzanger True that? |
| 15:38.33 | Essobi | Let's hear it. |
| 15:38.34 | bprice20 | oh see thats what I figured |
| 15:38.59 | bprice20 | I need someone to write the agi |
| 15:39.02 | Essobi | A new option on Dial()? That'd be super sweet. |
| 15:39.24 | bprice20 | well theres the retrydial option but I don't know what thats used for |
| 15:39.27 | bprice20 | not this application |
| 15:40.41 | tzanger | Essobi: this is just thinking out loud at this point |
| 15:40.41 | tzanger | my internet connections' acting up too |
| 15:41.04 | tzanger | callfile calls the busy # |
| 15:41.15 | tzanger | and bridges to a context that wait()s and checks a var |
| 15:41.25 | tzanger | and only dial()'s the extension waiting for nonbusy |
| 15:41.38 | tzanger | if the var shows the dial() is in progress |
| 15:41.40 | tzanger | it may be simpler |
| 15:41.48 | tzanger | because the channel wouldn't come up at all on busy |
| 15:42.11 | tzanger | and you'd just have an 'h' extension that checked for busy and did NOT clear the redial attempt if so |
| 15:42.18 | Essobi | Roger that.. you could kruft something similar to the retrydial function in the dialer, but Dial() would have to be able to continue operating after the initial call leg was dropped, then recreate it. |
| 15:42.48 | tzanger | Essobi: nah there's no need to alter Dial() |
| 15:42.59 | Essobi | :) I like hacking app_dial. |
| 15:43.09 | tzanger | that's what the Wait() is for... wait a few seconds and if the far end is busy just hangup |
| 15:43.41 | bprice20 | The reason I ask is because some switches have this functionality its like the last feature I've been unable to implement |
| 15:43.55 | Essobi | But true enough, I think .call files would be the way to go.. |
| 15:44.19 | bprice20 | yeah looking into that but how do I tell it to dial over and over again? |
| 15:44.31 | bprice20 | and furthermore to stop dialing, just remove the file? |
| 15:45.15 | Moc____ | anyone having problems withy g729 codec à |
| 15:45.26 | jeffik | i need to add capability to press * to access voice mail from outside asteisk, now when i press * i get the company directory |
| 15:45.35 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 15:46.06 | CiNzAs | A little poll |
| 15:46.07 | bprice20 | jeffec edit extensions.conf |
| 15:46.16 | CiNzAs | What you asterisk daily call load |
| 15:46.20 | CiNzAs | ? |
| 15:46.21 | bprice20 | look for * |
| 15:46.43 | bprice20 | CiNzAs more than 50 less that 100 |
| 15:47.10 | CiNzAs | I'm using mine for a monitoring system |
| 15:48.58 | *** join/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 15:49.02 | Dandan | hey all |
| 15:49.27 | Dandan | i can't get the parking call feature to work and to announce the parking ext. |
| 15:49.39 | Dandan | anyone knows how to get it to work? |
| 15:51.44 | Dandan | why is it so quiet? |
| 15:51.44 | jeffik | bprice20: ok, do you have the code to add? I had it before i did an upgrade, it seems that it goes under macros |
| 15:52.27 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-035-056.arcor-ip.net) |
| 15:52.39 | Blackvel | what means IPT? |
| 15:52.55 | Moc____ | Why I can't find 1 voip provider that always work... |
| 15:53.05 | Dandan | Moc: BV works for me :) |
| 15:53.34 | Moc____ | well Im talking about the resellers |
| 15:53.35 | CiNzAs | IPT ? |
| 15:53.42 | Dandan | oh |
| 15:54.15 | CiNzAs | IPT Internet Protocol Telephony (IP Telephony) |
| 15:54.31 | Blackvel | that means VOIP? |
| 15:54.37 | Blackvel | weird |
| 15:54.46 | tzanger | bprice20: email me akohlsmith@mixdown.ca about this so I don't forget, I will try a few things |
| 15:55.16 | tzanger | bprice20: callfiles are 'one shot' devices unless you have a retryattempt. but I wouldn't use that |
| 15:55.26 | tzanger | use a callfile and a little scripting in the dialplan to work with cron |
| 15:55.36 | tzanger | or hell just do it all in the dialplan |
| 15:55.45 | bprice20 | tzanger will do |
| 15:56.09 | *** join/#asterisk jmacz (~jmacz@201.245.167.80) |
| 15:56.14 | tzanger | bprice20: if I get it working you can paypal me what you think it's worth to you |
| 15:56.57 | bprice20 | thats fair 80-150 range seem fine |
| 16:00.08 | *** part/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
| 16:00.17 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
| 16:05.10 | sudhir492 | DrWho17: You are right, most likely the acpi is causing this problem. |
| 16:06.25 | *** join/#asterisk easimon (~easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 16:08.27 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 16:08.29 | CiNzAs | Hmmm |
| 16:08.39 | CiNzAs | what is rigth before the syslogd restart ? |
| 16:10.25 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 16:10.44 | *** join/#asterisk urmelZ (~urmel@194.231.22.13) |
| 16:20.06 | *** join/#asterisk Gh0sty (~Ghosty@81.11.211.48) |
| 16:26.06 | Wazb | can anuone help me COnfiguration Cisco with Asterisk , please!! |
| 16:27.08 | *** join/#asterisk jdg (~jdg@CA03F897.adsl.mana.pf) |
| 16:27.22 | Nugget | we can only answer questions, we can't just sit here all day and tell you what to type. |
| 16:27.58 | Wazb | SIP call is originating from Cisco to Asterisk. in which conf file i need to configure |
| 16:28.34 | Nugget | all of them, probably. what part do you not understand. |
| 16:28.49 | mutilator | all? |
| 16:28.52 | mutilator | only... 2? |
| 16:29.46 | *** join/#asterisk stevej (~stevej@67.97.36.243) |
| 16:30.08 | mutilator | paypal me some cash money and i'll walk ya through it Wazb |
| 16:30.51 | mutilator | :P |
| 16:31.31 | Wazb | sorry mutilator i cann't |
| 16:33.29 | *** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com) |
| 16:33.58 | *** part/#asterisk stevej (~stevej@67.97.36.243) |
| 16:35.37 | *** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net) |
| 16:35.42 | *** join/#asterisk FuriousGeorge (~root@ool-43516ebb.dyn.optonline.net) |
| 16:36.48 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 16:38.09 | TomL | ~seen manxpower |
| 16:38.15 | jbot | manxpower <~eric@dsl-209-205-172-111.i-55.com> was last seen on IRC in channel #asterisk, 3h 20m 58s ago, saying: 'All SoftPhones Suck!'. |
| 16:38.40 | *** join/#asterisk chris78 (~dg1nsw@saturn2.franken.de) |
| 16:38.43 | NewSole | Question... anyone have problems with audio not playing |
| 16:39.33 | *** join/#asterisk C4-Telecom (~sales@212.145.163.120) |
| 16:39.44 | foobos | would be easier to answer if you told what adio |
| 16:39.46 | foobos | audio |
| 16:40.02 | NewSole | ANY |
| 16:40.08 | C4-Telecom | Hi all |
| 16:40.11 | Nugget | I'm listening to iTunes right now. |
| 16:40.12 | NewSole | voice mail.... Greetings.... |
| 16:40.13 | Nugget | no problems at all. |
| 16:40.14 | foobos | yeah.. my tv has busted speakers |
| 16:40.21 | foobos | so no audio from my tv |
| 16:41.29 | shido6 | ? |
| 16:42.00 | NewSole | shido6... any idea |
| 16:42.41 | sivana | does sox convert mp3 to gsm? |
| 16:42.51 | *** join/#asterisk point (1000@213.27.44.55) |
| 16:43.12 | *** join/#asterisk stevej (~stevej@67.97.36.243) |
| 16:43.35 | foobos | sivana, you don't have sox installed? |
| 16:43.40 | sivana | yes |
| 16:43.49 | foobos | you don't have any mp3 files then? |
| 16:43.53 | sivana | for an IVR.. the studio sent them to me as mp3 |
| 16:44.06 | foobos | well isn't it easy to test |
| 16:44.37 | sivana | I guess my questions is "can sox convert mp3 to gsm?" |
| 16:44.38 | foobos | if that doesn't work, just use mpg123|sox pipe |
| 16:44.53 | foobos | sivana, instead of asking, you have sox and mp3 file.. |
| 16:45.08 | foobos | all you need to do is: sox some.mp3 some.gsm |
| 16:45.12 | sivana | ok, so you don't know |
| 16:45.54 | *** join/#asterisk girabraz (~christian@200.121.129.178) |
| 16:47.23 | foobos | i just didn't want to give you that on silver platter since its an very easy task to test |
| 16:47.53 | Wazb | i used nmap -sU 127.0.0.1 command and found that there is no listening on 5060 port, any help |
| 16:48.04 | sivana | ya, it's hard to type "yes" or "no" vs the lenghty chat we've had so far |
| 16:48.17 | foobos | wazb, try commmand. lsof -i -n instead |
| 16:48.46 | foobos | asterisk normally listens on all interfaces, not just 127.0.0.1(lo) |
| 16:51.16 | Dandan | adsi... |
| 16:51.50 | *** join/#asterisk jonathh (~asd@217.46.145.65) |
| 16:52.26 | jonathh | Hey.. can someone help to clarify to what i need to convert a bog standard phone(uk) to work with asterisk? i have heard noises about ATA? |
| 16:52.57 | sivana | foobos: the answer would have been "no" |
| 16:53.16 | shido6 | listens on all if you have bindaddr=0.0.0.0 set |
| 16:53.33 | shido6 | its supposed to listen on all if you dont have any bindaddr set , too |
| 16:53.38 | *** join/#asterisk heison (~heison@p180.n-lapop01.stsn.com) |
| 16:54.11 | NewSole | shido6 |
| 16:54.19 | shido6 | ? |
| 16:54.33 | foobos | sivana, my sox supports mp3. you just need to compile it in |
| 16:54.41 | foobos | or like i suggested mpg123|sox |
| 16:54.41 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 16:54.46 | NewSole | have you ever hear of asterisk not playing any sound |
| 16:55.01 | Delvar | jonathh: look the grandstream 486 ATA, has a port for analog phone and a lan port for SIP, so you can sip to asterisk. www.voiptalk.org sell them in the uk. |
| 16:55.02 | shido6 | sounds weird - |
| 16:55.27 | jonathh | nice one |
| 16:55.28 | jonathh | thanks |
| 16:55.53 | shido6 | how are you reaching your * box? |
| 16:56.15 | NewSole | i get |
| 16:56.29 | NewSole | Executing Playback("IAX2/8000000000000000@216.58.44.245:14569-21", "the-number-you-are-calling-is-not-in-service") in new stack |
| 16:56.36 | NewSole | and its a gsm |
| 16:56.54 | NewSole | then Playing 'the-number-you-are-calling-is-not-in-service' (language 'en') |
| 16:56.59 | NewSole | but no sound |
| 16:57.19 | shido6 | how do you know its gsm? |
| 16:57.24 | NewSole | but yet I can call out to PSTN numbers and talk and listen |
| 16:57.46 | Delvar | NewSole: do you have a digium card installed and setup corectly? sometimes if incorectly setup it kills asterisk audio |
| 16:57.49 | shido6 | NewSole, how do you know that file is in GSM format? |
| 16:58.00 | NewSole | it is |
| 16:58.09 | shido6 | how can you verify that? |
| 16:58.10 | Delvar | NewSole: also check what codec is being used (show channel SIP/bla) |
| 16:58.16 | NewSole | it was working fine earlyer now dead |
| 16:58.40 | shido6 | how are you reaching your asterisk box? |
| 16:59.34 | NewSole | iax2 |
| 16:59.58 | shido6 | is that a phone? |
| 17:00.00 | shido6 | softphone |
| 17:00.03 | shido6 | ? |
| 17:00.09 | NewSole | hardware phone |
| 17:00.17 | shido6 | pastebin.ca your iax.conf for that phone |
| 17:00.27 | NewSole | for the user |
| 17:00.35 | shido6 | for the user and the peer |
| 17:00.44 | NewSole | its in real time |
| 17:00.55 | shido6 | find out the configs for that phone |
| 17:00.57 | shido6 | and pastebin.ca them |
| 17:01.41 | NewSole | what do you need I know all the configs right now... but nothing changed.... it just went dead |
| 17:01.56 | Wazb | now i found UDP: 5060 entry, it means Asterisk is listening on 5060, right? |
| 17:02.09 | shido6 | if you're certain what you know and whats in realtime then what do you have for the user and the peer |
| 17:02.20 | NewSole | yup |
| 17:02.23 | shido6 | pastebin.ca that info as you would if you wrote the iax.conf yourself |
| 17:03.24 | jonathh | can anyone commment on the worth of converting an existing analogue handset over just getting a new SIP compliant one? |
| 17:03.28 | jt_ | is there such thing as digital pots line |
| 17:03.34 | jt_ | and does anyone know where i can get info on em |
| 17:03.44 | jonathh | the converters seem to start at Ģ60.. that could go towards a new sip phone! |
| 17:04.16 | shido6 | jonathh, can you take your analog phone with you wherever you are in the world and have it act as if it were in the office without incurring any charges? |
| 17:04.50 | *** part/#asterisk CiNzAs (~ashes@83.240.144.145) |
| 17:04.53 | jonathh | that is very true... (assuming you have the ata box and a BB line ;) ) |
| 17:04.56 | Delvar | jonathh: its nice to keep your nice analog phone but be able to use VOIP... |
| 17:05.15 | jonathh | i actually have 2 cordless... that would be nice to keep |
| 17:05.33 | Delvar | jonathh: also the 486 has fail over, if voip is broken it goes over PSTN as normal. |
| 17:05.41 | jonathh | how do you configure the IP of grandstream 486 ATA say? |
| 17:05.52 | jonathh | that is cute.. |
| 17:05.53 | DrWho17 | through a web interface |
| 17:05.56 | Delvar | jonathh: web page configuration |
| 17:05.57 | jonathh | result |
| 17:06.01 | jonathh | tht is nice |
| 17:06.12 | jonathh | i was worried it was a cheapy one.. and maybe only DHCP'd |
| 17:06.13 | NewSole | well here goes |
| 17:06.15 | NewSole | http://pastebin.ca/9481 |
| 17:06.41 | Delvar | hahah the 486 is easyer than some more expensive ones to configure |
| 17:06.48 | shido6 | what in the world |
| 17:06.57 | shido6 | NewSole, what codec is your iax phone set to? |
| 17:07.01 | PTG123 | anyone use polycoms in here? |
| 17:07.09 | jonathh | so anyone in the UK found the grandstream 486 ATA for less than Ģ60 tokens? |
| 17:07.18 | mutilator | i do |
| 17:07.45 | NewSole | phone can do g729/g723/ulaw/gsm |
| 17:07.52 | shido6 | what is it SET TO |
| 17:07.55 | PTG123 | mutilator: i have an ip500 that now reboots, downloads each file 4 times, then has a boot error and reboots again.. i think firmware may be corrupt.. any idea how to fix it? |
| 17:08.00 | NewSole | 729 |
| 17:08.06 | shido6 | do you have g729 licenses |
| 17:08.06 | shido6 | ? |
| 17:08.12 | NewSole | yup |
| 17:08.15 | shido6 | how many? |
| 17:08.23 | NewSole | total |
| 17:08.26 | NewSole | 1000 |
| 17:09.38 | NewSole | on that box 100... but none are being used |
| 17:09.44 | mutilator | sorry nope, havn't run into that yet |
| 17:09.59 | shido6 | NewSole, http://pastebin.ca/9482 |
| 17:10.03 | *** join/#asterisk riquisim0 (~riquisimo@63.245.8.94) |
| 17:15.49 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 17:18.22 | lohelle | Is it possible to config SER (sip proxy) to just forward requests to my asterisk servers (x.x.x.10 and x.x.x.11) and just use the SAME config on both servers? (only different IP's) ? does anyone have a SUPERSIMPLE ser.cfg that JUST handles forwarding (if that is all i need)? |
| 17:18.56 | vaewyn | anyone remember from a previous install... when you just plug the T cable in do you get yellow? or do they have to negotiate a bit before you even get that? |
| 17:19.43 | DrWho17 | hrm, has asterisk anyway to interface with the Lerg, or will I need to make it myself |
| 17:19.52 | fearnor | dr: you have to make it yourself. |
| 17:20.01 | DrWho17 | oye |
| 17:20.16 | *** join/#asterisk tld (~tld@mp-217-204-245.daxnet.no) |
| 17:20.25 | DrWho17 | should be able to do it with realtime extensions I think |
| 17:21.03 | DrWho17 | I was hoping someone had run into this before though, and would save me some work |
| 17:21.30 | tld | Any recommendations for a SIP provider to use with Asterisk? Obvious requirement is 'real' SIP feed, no black-box solutions. Also, I'm primarily interested in European providers, though US based ones are of interest too. |
| 17:21.35 | tld | (and is this too off-topic?) |
| 17:21.44 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 17:21.56 | DrWho17 | tld: http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers+B2B |
| 17:22.07 | tld | thanks |
| 17:23.34 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 17:23.55 | Wazb | how can use we extern=> s...... |
| 17:24.30 | *** join/#asterisk signuts (~signuts@209.172.11.54) |
| 17:25.25 | signuts | Hey all, i'm trying to create a call by placing a file in /var/spool/asterisk/outgoing, I would like multiple channels bound. How do I specify? I tried seperating each channel with & and adding multiple Channel: lines, but neither seem to work. The Wiki has failed me once again also. |
| 17:25.31 | heison[USENIX] | wazb: when there is no extensions specified, usually as a result of a context being specified by one of the channels |
| 17:26.00 | Hogie | what's the diff between app_addon_sql_mysql and cdr_addon_mysql? is app_addon for realtime config? |
| 17:26.05 | Hogie | in asterisk-addons |
| 17:27.08 | Wazb | it means if i want in order to work with extern => s is need to |
| 17:27.17 | DrWho17 | Hogie: one logs cdr's to mysql, the other allows you to do mysql queries within asterisk directly |
| 17:27.50 | Wazb | it means if i want extern => s to work i have to disable all extern => _ entries in that context |
| 17:28.21 | fearnor | dr: what are you really doing with lerg? |
| 17:28.45 | Hogie | thanks DrWho17, wish there was more docs in the source about stuff like that |
| 17:28.45 | fearnor | and s/extern/extern |
| 17:29.12 | DrWho17 | fearnor: eh? I want the calling area for sip users to match that of the local telco |
| 17:29.24 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca) |
| 17:29.38 | fearnor | for billing purposes? |
| 17:29.48 | fearnor | why do you care about realtime then |
| 17:29.49 | fearnor | blah |
| 17:30.01 | DrWho17 | yes billing purposes |
| 17:30.05 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 17:30.30 | DrWho17 | I want them to be able to 7-digit dial, in their area, and everything else is long distance |
| 17:30.39 | heison[USENIX] | wazb: i believe so, or if you put exten => s, before exten => _ should also work, since Asterisk does first match |
| 17:30.50 | fearnor | drwho: that's somewhat silly |
| 17:31.05 | fearnor | as in, i doubt anyone does that nowadays |
| 17:31.18 | fearnor | 'fuck you, 10 digit dialing only. its 21st century, get with it'. |
| 17:31.39 | DrWho17 | fearnor: yes, well I made that argument as well, but the people making the decisions didn't agree |
| 17:31.45 | fearnor | werd. |
| 17:31.56 | fearnor | then just prefix 7-digit with their npa |
| 17:32.06 | Wazb | <PROTECTED> |
| 17:32.07 | fearnor | and bill if it happens to be LD call in same NPA :( |
| 17:32.45 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 17:32.53 | heison[USENIX] | wazb: you either disable all exten => _. or put exten => s in front of them |
| 17:32.55 | rephorm | anyone here using polycom IP 300 phones? when a second call comes in, and i answer it, it drops the original call. according to the docs, it should place the original call on hold. any ideas? |
| 17:33.06 | heison[USENIX] | Wazb: that is my believe |
| 17:33.08 | DrWho17 | how about I just incorporate the lerg into asterisk and make it work just like their pots line they are switching from? |
| 17:33.11 | DaLion | seem got ap prob think one 23 lines full and more come in and realtime complaions |
| 17:33.28 | *** join/#asterisk GnL` (~a@Mix-Lyon-113-3-229.w80-9.abo.wanadoo.fr) |
| 17:34.21 | DrWho17 | looking into this enum stuff, see how feasible it is to move the lerg to it's format |
| 17:34.46 | *** join/#asterisk newmember (~newmember@dsl-lkbn-66-18-211-34-cgy.nucleus.com) |
| 17:34.48 | *** part/#asterisk GnL` (~a@Mix-Lyon-113-3-229.w80-9.abo.wanadoo.fr) |
| 17:35.21 | Wazb | no its not working <heison[USENIX]> |
| 17:38.56 | Gand_DJ | Here's a question for those using fwd on *. I have fwd setup and when I try to make an outgoing call, I get "all circuits are currently busy" |
| 17:39.27 | Gand_DJ | If I loadup my softphone, I can call any fwd # fine |
| 17:39.33 | *** join/#asterisk want561or772did (~ioshadf@68.71.213-38.atlsfl.adelphia.net) |
| 17:39.38 | want561or772did | could someone call 638271 on FWD please |
| 17:40.15 | Gand_DJ | sure.. 1 sec |
| 17:40.48 | Gand_DJ | I get music |
| 17:41.15 | want561or772did | i don't get ringing :( |
| 17:42.16 | Gand_DJ | wierd..lol |
| 17:42.34 | Gand_DJ | must somehow have it setup to auto-hold or something.. cuz I get music after a couple seconds.. no ringing from here either |
| 17:43.28 | want561or772did | yeah i have it start musiconhold after it gives instructions |
| 17:43.44 | want561or772did | it's no wonder ringing didn't happen though. i forgot fwd went via another path |
| 17:43.47 | want561or772did | could you call again? |
| 17:43.51 | Gand_DJ | sure |
| 17:43.54 | want561or772did | tnx |
| 17:44.23 | Gand_DJ | no ringing... just went to music-on-hold again |
| 17:44.31 | Gand_DJ | after about 5 seconds of nothing |
| 17:44.33 | want561or772did | damn |
| 17:44.47 | want561or772did | well i'll test it locally |
| 17:44.51 | Gand_DJ | ok |
| 17:48.28 | want561or772did | it just worked locally |
| 17:48.49 | want561or772did | now i'll loop over fwd |
| 17:50.10 | want561or772did | :/ fwd also dials an extension |
| 17:53.42 | *** join/#asterisk habakuk (~chatzilla@24-119-164-129.cpe.cableone.net) |
| 17:55.45 | want561or772did | one last time? |
| 17:55.56 | Dandan | what num? |
| 17:55.57 | Dandan | i can call |
| 17:56.15 | Gand_DJ | 1 sec |
| 17:56.22 | want561or772did | 638271 on fwd |
| 17:56.40 | Gand_DJ | I get nothing for 5 sec... and music now |
| 17:56.46 | harryvv | I get alot of these. Anyone care to explain what it means? |
| 17:56.48 | Dandan | i got sth like press star |
| 17:56.51 | harryvv | Apr 13 00:07:42 NOTICE[5833]: rtp.c:451 ast_rtp_read: RTP: Received packet with bad UDP checksum |
| 17:56.51 | Dandan | and then music |
| 17:57.26 | want561or772did | my console rang, though! |
| 17:57.37 | want561or772did | and then i listened to the music you were listening to through chanspy |
| 17:57.44 | want561or772did | so it's like a regular answering machine |
| 17:57.54 | Dandan | but i got some choppy words |
| 17:58.03 | want561or772did | my link is crappy |
| 17:58.14 | *** join/#asterisk Rick_Hunter (~rhunter@06-123.008.popsite.net) |
| 17:58.19 | want561or772did | i should set it to use something other than ulaw |
| 18:00.32 | *** join/#asterisk Jackthe (~jesse@thewhitehouse.adsl.utwente.nl) |
| 18:02.02 | *** join/#asterisk loick (~loick@APuteaux-151-1-6-21.w82-120.abo.wanadoo.fr) |
| 18:02.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 18:09.22 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:10.14 | *** join/#asterisk Jas_Williams (~jas_willi@host-83-146-47-134.bulldogdsl.com) |
| 18:11.40 | moy | does anybody knows a good (commercial or non-commercial) text to speech software? |
| 18:12.45 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 18:12.51 | signuts | moy, festival is the only one I know of (it's not the best either) |
| 18:13.16 | moy | do you know if has support for spanish voice |
| 18:13.17 | moy | ? |
| 18:13.30 | *** join/#asterisk pino (~z@host241-115.pool80116.interbusiness.it) |
| 18:14.31 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 18:16.58 | moy | yep, it seems like it does support spanish.... thanks! ;) |
| 18:19.08 | *** join/#asterisk stoyan (~stoyan@ns.burdenis.com) |
| 18:22.18 | foobos | moy, you should also try www.cepstral.com, they have few spanish voices |
| 18:23.45 | *** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
| 18:25.37 | *** join/#asterisk NewSole2 (~david@i216-58-44-245.avalonworks.net) |
| 18:26.06 | *** join/#asterisk snewpy (~markl@p180.n-lapop01.stsn.com) |
| 18:26.55 | vaewyn | ok... have my T hooked up to the Norhell... get a yellow alarm... he brings it up... still yellow alarm... and the Norhell puts it's side in red/green... with pri intense debug on that span I never see anything but SABME frames... anyone have ideas? |
| 18:27.40 | Jas_Williams | Are you using a T1 cross over sounds like a cable issue |
| 18:27.46 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 18:27.46 | vaewyn | (also... to check myself... the TE405P are labeled by port correctly? span 1 is labeled port 1 correct?) |
| 18:28.37 | vaewyn | Jas_Williams: When I plug in a stright cable it gets yellow alarm immediately... when I use crossover I get nothing... we have only checked the stright since we assumed yellow meant that polarity is correct |
| 18:28.54 | *** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net) |
| 18:29.10 | vaewyn | It spasses out the Norhell when we bring this up so we have done is very few times in short spats :P |
| 18:29.49 | vaewyn | Jas_Williams: when the remote end is disabled should I get a yellow or red with a correct cable? |
| 18:29.52 | *** join/#asterisk antifuchs (~asf@walrus.boinkor.net) |
| 18:31.37 | moy | foobos: Thanks! |
| 18:33.51 | *** join/#asterisk CoderCR (~creyna@ip68-8-131-103.sd.sd.cox.net) |
| 18:33.59 | vaewyn | Jas_Williams: or is there a way to test polarity so I know which to use? |
| 18:34.57 | tld | Any recommendations for a VoIP client that runs under FreeBSD that'll work with Asterisk? (preferrably using SIP) |
| 18:35.37 | Jas_Williams | Not that I know of TX must produce volts may be try a volt meter |
| 18:36.00 | *** part/#asterisk CoderCR (~creyna@ip68-8-131-103.sd.sd.cox.net) |
| 18:36.12 | vaewyn | wonder if my tone inject polarity indicator would help... hmmm :} |
| 18:38.03 | *** join/#asterisk Thus0 (~Thus0@dyn-83-157-205-163.ppp.tiscali.fr) |
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| 18:45.35 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 18:45.48 | *** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com) |
| 18:46.59 | likwid-- | was wondering what voip providers people use? |
| 18:47.12 | CoffeeIV | I have some of those cheap Digium Wildcard X100P FXO PCI cards. Someone is asking me if they use "loop start" or "ground start". Which is it, and where can I find out what those terms mean ? |
| 18:47.15 | harryvv | i use iax.cc. |
| 18:47.18 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 18:47.36 | Blackthorn | Is there away to update voicemail.conf without restarting asterisk? |
| 18:47.51 | mogorman | loop |
| 18:47.53 | mogorman | or kewl |
| 18:47.57 | mogorman | they dont support ground |
| 18:48.00 | bjohnson | a reload doesn't work? |
| 18:48.35 | CoffeeIV | mogorman: thanks, I'm googling those terms now . . . if you know of a good reference on that I'd appreciate the link if you have it handy ;) |
| 18:48.41 | Blackthorn | i could start and stop the service, but that would drop the calls. |
| 18:49.16 | likwid-- | harryvv, you keep a local phone service at all for local calls? cos 1.7cents a minute seems like it would outweigh over minimum local phoen service |
| 18:49.17 | mogorman | go to digium.com and look up depricated x100p |
| 18:49.21 | mogorman | it says all specs of card |
| 18:50.24 | CoffeeIV | mogorman: thanks again, I'm looking now |
| 18:50.44 | harryvv | blackhorn, you have a live production asterisk and dont want to reload? You could make it reload when no calls are comming in or out by typing the command restart when convinent |
| 18:50.58 | mogorman | google will help, i dont think it is easy to find on site anymore |
| 18:51.12 | Blackthorn | ok thanks harryvv |
| 18:51.16 | vaewyn | anyone here have a Meridian hooked via PRI to * that would be willing to copy paste the configs for both ends to me? I'm having real issues with this thing |
| 18:51.24 | harryvv | likwid, I have thought about that also. |
| 18:51.30 | harryvv | but |
| 18:52.14 | Blackthorn | was just curious if there was a "sip reload" type command that I was missing that would reload the voicemail |
| 18:52.24 | harryvv | I dont have faith in my local cable internet company to have there line up 24/7. It was down 2 hours yesterday. So thus could not make any calls if my local/long was total voip. |
| 18:52.46 | vaewyn | harryvv: that's what cellphones are for :P |
| 18:52.54 | harryvv | I have a prepaid cell |
| 18:52.55 | harryvv | :) |
| 18:52.58 | likwid-- | harryvv, so you do keep a local service? |
| 18:52.59 | harryvv | not cheap |
| 18:53.13 | vaewyn | I swear cellphones are getting a huge boost from being 'backup' to voip phones :} |
| 18:53.16 | *** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br) |
| 18:53.40 | harryvv | vaewyn, one way around it is subscribe to two internet carriers with independent backbones. |
| 18:54.02 | vaewyn | harryvv: yeah... but that is more money than the cell :} |
| 18:54.03 | likwid-- | harryvv, how about this, can asterisk be configured to use regular phone service on local numbers, and only voip for all others? |
| 18:54.14 | harryvv | btw, this is cool. Uniden just came out with a standard wireless with base phone and cell phone in one. |
| 18:55.08 | poli | likwid--, Yes. |
| 18:55.29 | harryvv | likwide, yes my local calls are though pstn and long distance is voip. But if you make alot of calls to a remote caller that also has broadband then the most reliable and free calls are done that way. |
| 18:56.01 | *** join/#asterisk L|NUX (linux@202.5.131.22) |
| 18:56.43 | *** join/#asterisk Revolutions (~Reload@dsl81-215-24493.adsl.ttnet.net.tr) |
| 18:56.48 | *** join/#asterisk Rith (~Rith@35-28-142-66.speedexpress.net) |
| 18:59.46 | *** join/#asterisk darby_t (~tom@dmv150.neoplus.adsl.tpnet.pl) |
| 19:00.27 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 19:01.03 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 19:03.12 | bjohnson | harryvv: a lot of people forget about that option |
| 19:03.18 | bjohnson | direct |
| 19:03.33 | bjohnson | well .. actually I like to suggest FWD since you can have voicemail that way |
| 19:05.16 | want561or772did | i think spool based out dialing is broken in latest cvs |
| 19:05.17 | point | disable native bridging in IAX notransfer=yes ? |
| 19:05.40 | kajtzu | anybody have good/bad experiences of running asterisk on x86_64 systems? |
| 19:05.48 | *** join/#asterisk cbox (~sn@130.226.235.2) |
| 19:05.55 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
| 19:06.16 | oden | kajtzu: you have problems on x86_64? |
| 19:06.20 | Dutts | hey guys can anyone tell meif asterisk needs portmap? |
| 19:06.25 | *** join/#asterisk fugitivo (~ajf@201.255.102.199) |
| 19:06.29 | kajtzu | Dutts: no |
| 19:06.38 | kajtzu | Dutts: portmapper is used for rpc services such as nfs :) |
| 19:06.38 | Dutts | cheers, just trying to harden my rd8 install |
| 19:07.27 | lohelle | does anyone have a sample extensions.conf + ser.conf for simple --> ser --> 2x asterisk setup ? |
| 19:08.14 | cbox | Hi.. does anyone know have i set up a rull to answer for calls with the extension like SIP/46928732-XXX? |
| 19:08.37 | fugitivo | anyone using FWD with IAX? |
| 19:08.54 | *** join/#asterisk crash3m_ (crash3m@crash3m.user) |
| 19:08.54 | niZon | fugitivo: Me, along with probablly half the people here :P |
| 19:08.59 | fugitivo | :) |
| 19:09.09 | fugitivo | i'm having a problem with outgoing calls |
| 19:09.12 | fugitivo | Apr 13 16:12:46 WARNING[2300]: chan_iax2.c:5553 socket_read: Call rejected by 65.39.205.121: Unable to negotiate codec |
| 19:09.22 | fugitivo | i can receive and talk, but cannot make outgoing calls |
| 19:09.28 | cypromis | check your llow= and disallow= lines |
| 19:09.34 | *** part/#asterisk crash3m_ (crash3m@crash3m.user) |
| 19:09.35 | niZon | *allow :P |
| 19:09.39 | fugitivo | i did :\ |
| 19:09.43 | fugitivo | disallow=all |
| 19:09.45 | fugitivo | allow=ulaw |
| 19:09.50 | fugitivo | in my iax.conf |
| 19:10.01 | niZon | whoever you're calling might not like ulaw.. |
| 19:10.40 | fugitivo | i'm calling 612, to get the time |
| 19:10.55 | niZon | i think they only use GSM for that |
| 19:11.26 | fugitivo | they have that number as a test for IAX |
| 19:11.41 | niZon | hmm |
| 19:11.49 | fugitivo | http://www.freeworlddialup.com/content/view/full/1501 |
| 19:12.03 | fugitivo | i don't understand, why the incomming connections works whithout any problem |
| 19:12.06 | niZon | did you simply fill in their examples? |
| 19:12.21 | fugitivo | yes |
| 19:12.41 | Dutts | anyone know of s quick and dirty sendmail config tutorial? all I want tio do is allow * to email me with my voicemail, want sendmail to use my isp's smtp server if poss? |
| 19:13.12 | niZon | fugitivo: see what happens if you do allow=all |
| 19:13.21 | fugitivo | niZon: ok |
| 19:13.41 | pino | Dutts: in 2005 you'd probably like to stay away from sendmail |
| 19:13.54 | Dutts | ah right ok, qhat else can i use then? |
| 19:14.06 | pino | whatever. exim and postfix to start with... |
| 19:14.09 | fugitivo | Dutts: qmail or postfix |
| 19:14.19 | fugitivo | niZon: the same error |
| 19:14.30 | niZon | fugitivo: what phone are you using? |
| 19:14.34 | Dutts | cheers guys i'll have a look |
| 19:14.36 | fugitivo | maybe i should look anything else for a misconfiguration? |
| 19:14.47 | niZon | check your phone config |
| 19:14.50 | niZon | and your sip.conf |
| 19:14.58 | fugitivo | niZon: kphone, and a regular phone conected to an ata |
| 19:15.05 | *** join/#asterisk Revolutions (~Reload@dsl81-215-24493.adsl.ttnet.net.tr) |
| 19:15.21 | niZon | make sure you're allowing ulaw in the phone conf, ata conf and sip.conf |
| 19:15.29 | fugitivo | ok |
| 19:18.48 | *** join/#asterisk killall-9 (~paulc@diana.null.ro) |
| 19:19.23 | *** part/#asterisk killall-9 (~paulc@diana.null.ro) |
| 19:21.38 | fugitivo | niZon: it seems to be everything ok |
| 19:21.54 | Blackthorn | I have seen this voicemail extension writen several ways such as exten => 1,2,Voicemail,1234 and exten => 1,2,Voicemail,u1234 and then others with the u(1234). Which is correct? |
| 19:22.02 | fugitivo | niZon: if i don't allow ulaw, should the incoming calls work? |
| 19:22.52 | niZon | as long as you allow a codec both clients support |
| 19:22.57 | niZon | it should work |
| 19:23.13 | fugitivo | because incoming calls works, outgoing calls don't |
| 19:23.39 | niZon | Blackthorn: the second one looks the best, but i'd prefer exten => 1,2,Voicemail(u1234) |
| 19:23.57 | niZon | fugitivo: what codec to incoming calls normally use? |
| 19:23.58 | signuts | Blackthorn, u means play the "Unavailable message" if the user recorded one. |
| 19:24.03 | *** join/#asterisk clive- (~pirch@rrba-146-111-227.telkomadsl.co.za) |
| 19:24.13 | fugitivo | niZon: iax is only allowing ulaw |
| 19:24.24 | fugitivo | niZon: that's why i don't understand the problem |
| 19:24.36 | niZon | hmm |
| 19:24.42 | niZon | you're sure your phones are configured to use ulaw? |
| 19:25.01 | fugitivo | niZon: yes, if not, how do i receive calls? :) |
| 19:25.08 | niZon | true |
| 19:25.28 | Dandan | who can help me with call parking? it is not announcing the number... :/ |
| 19:25.32 | ChkDigit | Blackthorn: The u1234 can be understood as unanswered, and b1234 as busy, and 1234 would be the same as u1234. |
| 19:25.49 | fugitivo | niZon: i get this when i make an outgoing call |
| 19:25.52 | fugitivo | Apr 13 16:26:16 WARNING[2662]: chan_iax2.c:5553 socket_read: Call rejected by 65.39.205.121: Unable to negotiate codec |
| 19:25.59 | fugitivo | i think that's fwd server |
| 19:26.01 | niZon | is it just FWD's 612 number? |
| 19:26.04 | fugitivo | yes |
| 19:26.13 | niZon | i just tried it, worked fine |
| 19:26.15 | fugitivo | if i try to call a person, i get the same error |
| 19:26.15 | Blackthorn | ok thanks.. |
| 19:26.32 | fugitivo | some problem with me and fwd server maybe |
| 19:26.40 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 19:27.06 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 19:27.15 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 19:27.17 | niZon | could be |
| 19:27.25 | niZon | i don't even use allow in my iax.conf for fwd :P |
| 19:27.33 | fugitivo | really? |
| 19:27.35 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 19:27.58 | PBXtech | if you upgrade your motherboard/ethernet will digium allow you to redo your g729 licence? |
| 19:28.41 | Romik | somebody can advice about crash on asterisk 1.07 http://pastebin.ca/9491 ? |
| 19:28.53 | *** join/#asterisk dooder (~nateputna@66.241.90.21) |
| 19:29.18 | niZon | fugitivo: er, nm. I put it right after the register line |
| 19:29.21 | dooder | does FWD sometimes take a while to register when you first sign up |
| 19:29.23 | eKo1 | Argh. I don't like FXO gateways. Stupid PoS analog crap. |
| 19:29.28 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
| 19:29.33 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
| 19:29.36 | *** part/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
| 19:29.42 | *** join/#asterisk bah (048830696@ACAD1EA9.ipt.aol.com) |
| 19:29.56 | fugitivo | niZon: me too, well, i'll check all the configurations again |
| 19:30.02 | fugitivo | niZon: thanks for your time |
| 19:30.18 | niZon | np |
| 19:31.14 | *** join/#asterisk jmacz (~jmacz@201.245.167.80) |
| 19:32.10 | PBXtech | analog echo crap |
| 19:32.12 | PBXtech | :/ |
| 19:32.17 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 19:32.23 | niZon | hm, my wireless is having a hernia |
| 19:32.24 | pino | Romik: very interesting, my own 1.0.7 has no chan_features.c. have you removed all the old modules before upgrading? |
| 19:32.50 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 19:34.12 | fugitivo | anyone tried if stanaphone works with asterisk? |
| 19:34.17 | *** join/#asterisk Tuplink (~dsfsk@68-232-92-239.chvlva.adelphia.net) |
| 19:34.51 | Tuplink | how do i make incoming calls from fwd go to my switchboard? |
| 19:35.23 | Romik | Apr 13 20:44:08 WARNING[12418]: codec_speex.c:166 speextolin_framein: Out of buffer space |
| 19:36.04 | wolfson | anyone head of fiercemarkets, just got some spam from them regarding an SS7 product |
| 19:36.04 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 19:36.20 | wolfson | head=heard |
| 19:36.21 | dooder | i can't get fwd setup at all. sucks |
| 19:36.26 | dooder | Apr 13 19:36:07 NOTICE[1378]: Registration of '642438' rejected: Registration Refused |
| 19:36.31 | Tuplink | hehe |
| 19:36.47 | Tuplink | dooder |
| 19:36.47 | Uther_P | has anyone here used or heard anything about sipXpbx from sipfoundry? |
| 19:37.10 | *** join/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com) |
| 19:37.11 | dooder | i followed the guide on voip-info |
| 19:37.16 | Tuplink | in [default] include include => fromiaxfwd |
| 19:38.01 | Tuplink | use the guide on FWD |
| 19:38.45 | Tuplink | how do i get calls from FWD to go to a switchboard |
| 19:38.55 | xai | can asterisk use openldap? |
| 19:39.22 | *** join/#asterisk Tili (~Tili@202-133-65-162-dialup.sat.net.pk) |
| 19:39.36 | pino | Uther_P: i compiled it but did not use it :) |
| 19:40.43 | xai | nugget: hey.. how goes it. |
| 19:40.52 | Blackthorn | Ok, I'm trying to setup the voice mail. And I can call and leave messages just fine. But when i check voice mail by dialing it's own number it just rings off the hook. I'm sure i've missed something but can't put my finger on it :P |
| 19:41.21 | Nugget | I've been outside planting pepper plants, trying to enjoy today's sunshine. |
| 19:41.38 | Dutts | can anyone tell me what this means |
| 19:41.38 | Dutts | Apr 13 19:43:52 NOTICE[17344]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 19:42.01 | Dutts | I'm only got one line and it happens when I'm making an outgoing call, is it just that I'm using up all the outgoign lines? |
| 19:42.19 | Jas_Williams | Blackthorn, What do you have in extensions.conf for the Voice mail extension |
| 19:42.22 | xai | Ok, i see that it does.. I wasn't sure where to look for that info.. |
| 19:43.17 | Tuplink | how do i get calls to the pbx to take me a menu |
| 19:43.18 | Tuplink | ? |
| 19:44.08 | tzafrir_laptop | Tuplink, you mean IVR? |
| 19:44.34 | Tuplink | um... yes like press one to... two to |
| 19:44.43 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 19:44.57 | tzafrir_laptop | Yes, this is IVR: Interactive Voice Response. |
| 19:45.04 | tzafrir_laptop | Basics: |
| 19:45.51 | tzafrir_laptop | Use the context parameter in the channel (sip.conf, zapata.conf, whatever) to send the calls to a specific context in extensions.conf |
| 19:46.15 | Tuplink | IAX |
| 19:46.21 | Blackthorn | exten => xxxxxxx,2,voicemail(xxxxxxx) |
| 19:46.25 | tzafrir_laptop | contexts in extensions.conf are the sections (the titles in []) |
| 19:46.43 | tzafrir_laptop | Use the demo example |
| 19:46.49 | xai | Nugget: what variety? Somthing is eating at my serranos. |
| 19:46.49 | Tuplink | got thet [switchboard]yadda |
| 19:47.07 | tzafrir_laptop | for IAX: iax.conf |
| 19:48.10 | Tuplink | [iaxfwd] |
| 19:48.10 | Tuplink | type=user |
| 19:48.10 | Tuplink | context=switchboard |
| 19:48.13 | Tuplink | ? |
| 19:48.36 | L|NUX | when i issue this command sip reload then i get this on * cli |
| 19:48.36 | L|NUX | Apr 13 07:48:35 WARNING[9671]: frame.c:988 ast_parse_allow_disallow: Cannot allow unknown format 'h323' |
| 19:49.19 | Nugget | some habanero plants I bought at lowe's and some red savina seeds. |
| 19:50.13 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
| 19:50.35 | funxion | anyone here? |
| 19:51.10 | *** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com) |
| 19:51.51 | funxion | Im unable to modprob wcte11xp trying to install a te110p |
| 19:52.02 | *** join/#asterisk Curus (~Curus@83.72.32.8.ip.tele2adsl.dk) |
| 19:53.36 | funxion | anyone |
| 19:53.38 | funxion | ? |
| 19:54.34 | Tuplink | so... what is the context in IAX.conf |
| 19:54.35 | *** join/#asterisk iq (~iq@70-57-182-73.omah.qwest.net) |
| 19:54.35 | Tuplink | is that to witch [] it takes you |
| 19:54.35 | funxion | yes |
| 19:54.39 | Tuplink | Apr 13 15:50:50 NOTICE[10935]: chan_iax2.c:5761 socket_read: Rejected connect attempt from 65.39.205.121, request '641726@switchboard' does not exist |
| 19:55.22 | |Vulture| | funxion: fedora core 3? |
| 19:55.40 | funxion | its centos |
| 19:55.41 | funxion | aah |
| 19:55.50 | |Vulture| | centos? |
| 19:56.03 | funxion | redhat cased |
| 19:56.05 | funxion | based |
| 19:56.20 | |Vulture| | ah well if its like FC3 I had the same problem |
| 19:56.25 | |Vulture| | well fine... |
| 19:56.33 | rephorm | funxion: that's a t1 card, right? |
| 19:56.39 | |Vulture| | yea |
| 19:56.43 | |Vulture| | its the new T1/E1 card |
| 19:56.45 | rephorm | gah. he left... |
| 19:56.51 | |Vulture| | I just installed it it was a whore on FC3 |
| 19:57.20 | rephorm | the centos kernel is 2.4.21, and doesn't have nethdlc stuff |
| 19:57.33 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
| 19:57.36 | rephorm | the centos kernel is 2.4.21, and doesn't have nethdlc stuff |
| 19:57.59 | rephorm | so, there are issues if you need hdlc. but, the module should work otherwise |
| 19:57.59 | funxion | sry didnt mean to disco like that |
| 19:58.15 | rephorm | funxion: actually, which version of centos? 3.4? |
| 19:58.50 | syle | you guys saying fc3 has to many problems? |
| 19:59.01 | funxion | not sure |
| 19:59.03 | funxion | checking |
| 19:59.22 | rephorm | funxion: cat /etc/redhat-release |
| 19:59.25 | |Vulture| | syle: I like fc3... it does have problems though |
| 19:59.49 | funxion | that doesnt werk |
| 19:59.49 | Curus | Suddenly my Grandstream 486 cannot register with my asterisk. How can I debug it? |
| 20:00.11 | funxion | its 3.4 |
| 20:00.22 | r0d3nt | Curus, asterisk -r |
| 20:00.26 | poli | Curus, Did you check if there are any erros using asterisk -vvvvvgc ? |
| 20:00.31 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 20:00.42 | Dutts | does running in verbose debug mode (loads of v's) slow the system down considerably... I'm playing around with a 1x1 * setup and finidnig it's having difficulty picking up some of the dtmf |
| 20:00.56 | |Vulture| | funxion: what happens when you modprobe it? |
| 20:01.06 | r0d3nt | Dutts, which has nothing to do with -vvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 20:01.11 | Curus | chan_sip.c:8804 handle_request_register: Registration from '<sip:ht486@amorsen.dk>' failed for '80.163.10.87' |
| 20:01.26 | Weezey | does auto answer on a SIP device that supports it, occupy an extension? |
| 20:01.35 | Curus | I know I have the secret correct, because calling does work (and doesn't if I choose a wrong secret) |
| 20:01.37 | poli | Dutts, Probably not... you just enable some extra lines of code that, IMHO, should get no sensible impact in performance. |
| 20:01.39 | Dutts | r0d3nt: ok mat ejust wanted to make sure it wasn't too much of a resource hit that it's losing it.... had that problem a while back on some prosody cards.... what is the problem then do you know? |
| 20:02.00 | poli | Curus, start asterisk with -vvvvvvvvvvgc and see if there are any errors. |
| 20:02.06 | funxion | <PROTECTED> |
| 20:02.16 | r0d3nt | dtmf settings on the device.. or the extension... |
| 20:02.19 | |Vulture| | ah yes... |
| 20:02.25 | r0d3nt | there is no performance hit for verbosity @ the console |
| 20:02.27 | |Vulture| | funxion: thats the same thing FC3 deals with |
| 20:02.56 | *** part/#asterisk mozrat (~mozrat@80.68.89.215) |
| 20:02.59 | Dutts | I'm in the uk, don't think I've done anythign other than defaults so do you know what's different? It seems to be ok 90% of the time, but misses the digit 1 in particular if it's in the middle of a string |
| 20:03.08 | |Vulture| | funxion: do a "uname -r" |
| 20:03.23 | Curus | poli: Nothing sip-related except that it can't find sip_notify.conf |
| 20:03.30 | funxion | 2.4.21-27.0.1.ELsmp |
| 20:03.34 | funxion | is what it returns |
| 20:03.41 | Curus | And that handle_request_register thing |
| 20:04.06 | Tuplink | what dose Rejected connect attempt from 65.39.205.121, request '641726@switchboard' does not exist |
| 20:04.09 | Tuplink | mean |
| 20:04.09 | funxion | I get 03:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface when I do lspci |
| 20:04.11 | poli | Curus, Ok, if I were you, I would go with a network debug software (like ethereal) and understand what is truly happening between them. |
| 20:04.55 | Curus | I thought sip debug ip whatever would be enough for that |
| 20:05.03 | Dutts | if I update my zapata.conf do I need to retsart asterisk or is a reload sufficient |
| 20:05.41 | Blackthorn | Scratches head: I can call my # from another line and leave voicemail. I can verify it is creating it on the * console. When I pickup my phone and dial it's own number to gain access to voicemail it rings normally then says person is not available. Have an idea? |
| 20:05.48 | poli | Curus, could be, I don't know, I never used it. I always go with network-debug. |
| 20:06.07 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
| 20:06.29 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
| 20:06.33 | funxion | rephorn |vulture| any ideas? |
| 20:06.34 | Uther_P | Blackthorn: what does the cli say whn you call your own number? |
| 20:06.36 | |Vulture| | sorry Ill help you in 5 min |
| 20:06.42 | |Vulture| | I can help you fix it but I am busy |
| 20:06.47 | funxion | ok |
| 20:06.49 | funxion | sry |
| 20:07.43 | Blackthorn | when i call my own number it just says person is not availbel and puts me into leaving a voice mail as if i'm calling from another #. |
| 20:07.56 | Uther_P | Blackthorn: what does the CLI say |
| 20:08.09 | *** join/#asterisk darkskiez (~mhb@host-84-9-102-21.bulldogdsl.com) |
| 20:08.26 | |Vulture| | okay sorry about that had to take care of an urgent echo issue |
| 20:08.36 | |Vulture| | okay |
| 20:08.42 | Blackthorn | The console shows the voice mail being created when leaving a message. When dialing it to check voicemail it says nothing. |
| 20:08.51 | |Vulture| | go into /usr/src and do a |
| 20:09.12 | |Vulture| | ln -s /lib/modules/2.4.21-27.0.1.ELsmp/build/ linux-2.4 |
| 20:09.24 | |Vulture| | that should be what the directory is |
| 20:09.28 | |Vulture| | your running smp right? |
| 20:09.51 | Uther_P | Blackthorn: well, you have to configure it to goto the voicemailmain for that box.. |
| 20:10.02 | funxion | yes |
| 20:10.51 | Uther_P | Blackthorn: for my extensions I use a macro |
| 20:11.13 | |Vulture| | funxion: reboot the box, then we will rebuild the zaptel drivers |
| 20:12.30 | |Vulture| | funxion: do you have a T1 PRI plugged into the TE110P right now? |
| 20:12.36 | funxion | yes |
| 20:12.58 | |Vulture| | how did you build your zaptel? |
| 20:13.05 | |Vulture| | make clean;make;make install ? |
| 20:13.08 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
| 20:13.18 | |Vulture| | and are you running latest CVS? |
| 20:13.20 | jlewis | using 2 tdm400p's and one x100p, is there some reason my channels are 1 and 3-10...no channel 2? |
| 20:13.25 | Blackthorn | Does anyone have a url to setting up voicemail? |
| 20:13.31 | Egonis | Fresh install of Asterisk, and just got a SIP Phone -- what can I do quick and easy to sip.conf? do I need to static IP my sip phone? |
| 20:14.01 | Derkommissar | jlewis, is that e-1's or t1's |
| 20:14.21 | jlewis | neither...they're all fxo ports |
| 20:14.48 | focks | are the dual 10/100's on a phone like a Grandstream GXP-2000 like a mini-hub/switch? |
| 20:15.01 | jlewis | just wondering why/how channel 2 got skipped and if that's something I should have known to expect |
| 20:15.13 | Dandan | focks: grandstream has a hub |
| 20:15.15 | Dandan | not a switch |
| 20:15.20 | jlewis | it looks like 1 is the x100p, and 3-10 are the tdm400 ports |
| 20:15.28 | focks | Dandan are there phones with a switch? |
| 20:15.45 | Dandan | i heard snom and polycoms are... but it is NOT confirmed |
| 20:17.48 | *** part/#asterisk cbox (~sn@130.226.235.2) |
| 20:18.16 | Uther_P | Blackthorn: http://pastebin.ca/9494 |
| 20:19.52 | johnnyb | The grandstream BT-102's second port is near-worthless |
| 20:20.06 | easimon | dandan: http://www.grandstream.com/user_manuals/GXP2000.pdf says it has a switch... |
| 20:21.04 | focks | i just need a solution because all the customer has is 1 cat5 for the and 1 cat3 for their current phone |
| 20:21.46 | focks | yes i see it claims it is a switch |
| 20:22.40 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
| 20:22.47 | johnnyb | Of course you can _buy_ switches fairly cheaply these days. CompUSA has one for $20. |
| 20:23.14 | focks | right, but that's cumbersome :( |
| 20:23.27 | focks | rather it be in the phone |
| 20:23.45 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:23.59 | PTG123 | whats the questions about the switches in the phone? |
| 20:24.08 | PTG123 | polycom and cisco both have a 2nd port |
| 20:24.17 | |Vulture| | thats why I love my IP500s |
| 20:24.18 | kajtzu | yeah |
| 20:24.21 | focks | just wondered which phones had a switch |
| 20:24.26 | |Vulture| | got rid of all my 79x0s |
| 20:24.27 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
| 20:24.31 | focks | since i only have 1 cat5 at each desk |
| 20:24.31 | BuckRogers | anyone have an issue with grandstream handytone not fowarding the incomming call to the fxs port |
| 20:24.44 | kajtzu | PTG123: the 7940/60 2nd port can be used as a host or as a switchport (with spanning tree) |
| 20:24.45 | PTG123 | speaking of IP500s, mine fetches each file 4 times, then reboots with bad boot.. and idea how to fix it.. think it got unplugged during firmware load |
| 20:25.31 | kajtzu | s/PTG123/focks/ |
| 20:25.36 | easimon | kajtzu: what worth is spanning tree in a 2-port-switch? |
| 20:25.48 | focks | no kiddin |
| 20:26.12 | kajtzu | easimon: prevents loops in case some dimwit connects the 2nd port to another switch and that one back to your regular topology ..... |
| 20:26.28 | BrianR___ | easimon: lets other switches better calculate path cost.. Ie, a path with two repeaters is more expensive than one with none. |
| 20:26.44 | *** part/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:26.51 | BrianR___ | Also, like kajtzu says, it prevents dimwits from causing loops. |
| 20:26.52 | easimon | ... regular LARTs do the same job - and are more fun ;) |
| 20:27.30 | *** join/#asterisk mozrat (~mozrat@80.68.89.215) |
| 20:28.00 | jlewis | ah...figured out the skipped channel thing...it's the post-install ztcfg...loading wcfxo and wcfxs one right after the other seems to confuse things |
| 20:28.09 | kajtzu | I run a bunch of 7940/7960s with CME/ITS and am actively transitioning them to sip and asterisk :) |
| 20:29.01 | mozrat | Evening guys - quick straw poll amongst you all. Which of the web configuration tools do you use for Asterisk? AMP looks great but the setup seems a little rigid |
| 20:29.49 | harryvv | kaj, is this for your local bussiness? |
| 20:30.35 | *** join/#asterisk systest (~systest@63.116.136.130) |
| 20:30.47 | |Vulture| | the latest stable zaptel drivers are yummy ;) |
| 20:30.49 | focks | anyone know what the Signate 5000 servers base cost is? |
| 20:30.57 | focks | it's a beast |
| 20:31.00 | kajtzu | harryvv: yes. I started running its back in 2001/2002 or so and then it changed name to cme :) |
| 20:33.30 | harryvv | you mean you are installing asterisk systems or just running it for your own company. |
| 20:33.38 | kajtzu | harryvv: internally |
| 20:33.42 | harryvv | okay |
| 20:33.45 | harryvv | how many phones |
| 20:34.17 | kajtzu | about 20. The coolest thing with asterisk is conferencing. I don't have to use Unity anymore. |
| 20:34.23 | Curus | What does "Contact" mean in a SIP register message? |
| 20:34.31 | *** join/#asterisk Geraldoramos (~GIGAhost@200141138156.user.veloxzone.com.br) |
| 20:34.40 | kajtzu | (I really, really, really hate Unity) |
| 20:34.57 | Egonis | I have a fresh install of Asterisk, with (what I know) as a properly configured install.. connected a SIP Phone, and dialed '1000' and it says 'disconnect'... what am I missing? |
| 20:35.25 | harryvv | what is unity a commercial pbx? what did it cost |
| 20:36.06 | Blackthorn | ego: setup your extensions.conf file. Which defines where and what dialing 1000 should do. |
| 20:36.47 | Curus | asterisk does return "Forbidden" when my poor HT486 tries to register. Not very nice of Asterisk at all. |
| 20:37.07 | kajtzu | harryvv: cisco |
| 20:38.00 | Blackthorn | ego try this: exten => 1000,1,Datetime() |
| 20:38.06 | Blackthorn | in your extensiosn file |
| 20:38.11 | Curus | asterisk also likes returning "trying", my SIP provider doesn't do that. |
| 20:38.17 | Blackthorn | it will read back the date and time toyou |
| 20:38.26 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 20:38.30 | harryvv | kaj, what did you not like about unity and how much did it cost with the phones? |
| 20:38.42 | harryvv | how much did the unity pbx cost |
| 20:40.26 | Jas_Williams | Unity is cisco voice mail not apbx |
| 20:41.42 | kajtzu | harryvv: unity works with cisco call manager |
| 20:41.51 | kajtzu | harryvv: unity expess works with call manager express (and call manager) |
| 20:42.13 | kajtzu | unity is a family really. it includes voice mail, conferencing, etc. |
| 20:42.31 | harryvv | what did it cost to you and for how many phones? |
| 20:42.44 | Blackthorn | ok thanks for all the help this evening. time to close up and head home. |
| 20:43.04 | Tuplink | how do i define sounds |
| 20:43.07 | Tuplink | ? |
| 20:43.26 | BuckRogers | anyone have an issue with grandstream handytone not fowarding the incomming call to the fxs port |
| 20:44.33 | *** join/#asterisk azid (~janne@1-1-10-32a.um.um.bostream.se) |
| 20:44.39 | Jas_Williams | Tuplink, You need to provide more information what are you trying to do ? |
| 20:44.43 | easimon | i recently tested a german speech pack for asterisk - just installed the files and put language=de" into zapata.conf. asterisk also used them, but numbers and dates were "in english order"... how do i fix this? |
| 20:45.31 | kajtzu | harryvv: depends on your cisco rebates :) |
| 20:45.45 | harryvv | give me a ball park |
| 20:45.54 | harryvv | was t 5 grand 6 grand what? |
| 20:45.58 | harryvv | round it off ;) |
| 20:46.17 | azid | i have a voicemail problem. when callers hangs up during the playback of the VM message, an empty voicemail is being stored.. any ideas? |
| 20:46.51 | Jas_Williams | azid, what pstn connectivity do you have ? |
| 20:47.06 | azid | pstn through sip |
| 20:47.58 | Jas_Williams | Sip should clear down cleanly no Idea why you get a blank message |
| 20:48.39 | azid | ok :( |
| 20:49.09 | foobos | azid, try adding some silence to the entrance message |
| 20:49.18 | azid | i mean.. i call over an analogue line to a SIP-gateway and then sip to asterisk |
| 20:49.38 | harryvv | <PROTECTED> |
| 20:49.42 | harryvv | hehe |
| 20:49.45 | harryvv | so much for that |
| 20:50.05 | azid | foobos, ok. i'm using the default now |
| 20:51.18 | *** part/#asterisk systest (~systest@63.116.136.130) |
| 20:54.42 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:54.54 | *** part/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:54.56 | |Vulture| | anyone use spandsp to send faxes? |
| 20:56.34 | foobos | its pretty random what actual facsmiles work with spandsp |
| 20:56.41 | eKo1 | WTF?! For some stupid werid reason, adding a w in the Dial() causes the call to go through; otherwise, nada. |
| 20:56.45 | foobos | got very bad experiences with canons |
| 20:56.50 | |Vulture| | foobos: I have inbound faxes working 100% |
| 20:57.09 | cypromis | foobos: you seem to be using an old spandsp version than I suppose |
| 20:57.25 | foobos | i tried it about month ago |
| 20:58.02 | |Vulture| | I was looking for a way to send faxes like someone emailes faxes@asteriskbox and it sends the attachments |
| 21:00.58 | eKo1 | Him, the W option also seems to cause problems. |
| 21:03.07 | eKo1 | Him, maybe I'm confused. In Dial(sip/pstn/w1213456), what does that w do exactly? |
| 21:03.09 | bkw_ | spandsp works great |
| 21:03.14 | bkw_ | still a few issues left |
| 21:03.16 | bkw_ | but not many |
| 21:03.59 | *** join/#asterisk bajanman (~william@cp66-203-194-230.cp.telus.net) |
| 21:04.30 | bajanman | Hello |
| 21:04.45 | Jas_Williams | eKo1, wait 0.5 sec then dial |
| 21:04.53 | |Vulture| | bkw_: do you use it for outbound faxing as well? |
| 21:04.54 | bkw_ | dialtone isn't coming up fast enuf |
| 21:04.55 | bajanman | I'm wanting to know what the mostly used / stable web/admin/managment app is |
| 21:05.04 | |Vulture| | bkw_: or do you use a standard fax machine? |
| 21:05.07 | bkw_ | |Vulture|, not yet.. i'm going to test some stuff out here in a few |
| 21:05.12 | bkw_ | we use hylafax for outbound |
| 21:05.19 | bkw_ | but once I get stuff going its going to do outbound here |
| 21:05.24 | *** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net) |
| 21:05.34 | eKo1 | Jas_Williams: Where is that documented? |
| 21:05.41 | |Vulture| | bkw_: outbound via spandsp or Hylafax there? |
| 21:05.51 | bkw_ | hylafax |
| 21:05.54 | bkw_ | soon spandsp also |
| 21:05.58 | eKo1 | I can use Dial(sip/pstn/B1213456) and it will work just as well. |
| 21:06.02 | denon | bkw: you're taking on fixing faxing? |
| 21:06.08 | denon | or just playing with making it work for you? |
| 21:06.11 | bkw_ | faxing isn't broken |
| 21:06.16 | eKo1 | I think I can put any letter I want and it will work. |
| 21:06.22 | bkw_ | as long as you generate the correct format files |
| 21:06.23 | denon | thought it was still pretty quirky |
| 21:06.25 | bkw_ | outbound works fine |
| 21:06.31 | bkw_ | nope |
| 21:06.32 | denon | how come its not in CVS? <G> |
| 21:06.32 | bkw_ | it works fine |
| 21:06.40 | bkw_ | no clue |
| 21:06.40 | CoaxD | bkw, my love |
| 21:06.46 | |Vulture| | bkw_: I couldn't find anything in the wiki about installing Hylafax on * |
| 21:06.47 | bkw_ | maybe because steve has not disclaimed it |
| 21:06.51 | CoaxD | bkw: May I buy you a carmel sundae? |
| 21:06.54 | denon | ah, maybe .. |
| 21:06.57 | bkw_ | |Vulture|, we don't use asterisk on outbound |
| 21:06.58 | bkw_ | har har har |
| 21:07.00 | CoaxD | bkw: Tonight? On our date? |
| 21:07.21 | denon | CoaxD's an insult to his state |
| 21:07.22 | CoaxD | bkw: *wink* |
| 21:07.29 | |Vulture| | bkw_: oh okay... well if you need a hand with spandsp outbound lemme know Ill be glad to test/help |
| 21:07.39 | CoaxD | denon: Yeah, we're all a bunch of gay haters |
| 21:07.44 | denon | mm? |
| 21:07.49 | denon | no .. you're just lame .. nothin to do with gay stuff |
| 21:07.52 | Jas_Williams | eKo1, I did think it was only in ZAP channels looks like you have an issue with the dial plan in your sip gateway |
| 21:07.53 | CoaxD | denon: Hehe |
| 21:08.27 | bajanman | would anyone be so kind, as to give me their opinion on what would be the most stable user interface, OTHER than *@home? |
| 21:08.47 | Jas_Williams | bajanman, VI |
| 21:08.51 | bajanman | lol |
| 21:08.53 | bajanman | ok ok. |
| 21:08.55 | eKo1 | Jas_Williams: I tried it with a digit and it works as well. I think the gateway either ignores or looses the first 'digit' sent to it. |
| 21:08.55 | bajanman | I use vi |
| 21:09.11 | bajanman | JAS: but, I'm looking at: voice mail web. |
| 21:09.14 | eKo1 | no no, vim |
| 21:09.14 | Jas_Williams | eKo1, sounds like it |
| 21:09.28 | bajanman | JAS:working on easy extensions input: without vi... |
| 21:09.28 | eKo1 | I hate analog gateways. Shit! |
| 21:10.08 | eKo1 | bajanman: echo "#include my-new-ext" >> extensions.conf |
| 21:10.38 | Uther_P | can someone refresh me on the asterisk app name for the audio loopback? |
| 21:10.38 | bajanman | eK01: yes I know how. I'm sorry, maybe I'm not being clear? |
| 21:11.11 | bajanman | I can use .conf files. I'm looking for a web/GUI? that I can use for viewing my messages, adding extensions, etc |
| 21:11.41 | bajanman | eKo1: I tried *@home: and I cried: it sucks. but how good is AMP? |
| 21:12.28 | Jas_Williams | Uther_P, echo() |
| 21:12.34 | Uther_P | heh, thanks |
| 21:13.21 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
| 21:13.34 | bajanman | what would be a good "dial plan" wizzard to use? |
| 21:13.47 | anti | vi? :) |
| 21:13.52 | bajanman | *sigh* |
| 21:13.58 | |Vulture| | why use all that crap... the sources are the best way |
| 21:14.35 | Jas_Williams | bajanman, @home is built using AMP so no difference |
| 21:14.38 | bajanman | I'm good at linux, but a noob @ *. so I'm learning: but *@home was the worst. AMP does all kinds of stuff to the .conf files |
| 21:14.46 | bajanman | Jas: Ok, good point, BUT, |
| 21:15.12 | bajanman | Jas: is there anything out there, that will "help" me create a dial plan, and get me familiar with the * structure/.conf files? |
| 21:15.51 | |Vulture| | bajanman: make samples |
| 21:16.02 | |Vulture| | it creates a buncha step-through sample files |
| 21:16.15 | Jas_Williams | bajanman, make samples in the source directory or look at src/asterisk/configs/????.conf.sample or read wiki |
| 21:16.23 | bajanman | Vluture: yea, did that. guess I should stop being lazy, and actually spend the time to view it, BUT |
| 21:16.24 | Jas_Williams | ~docs |
| 21:16.25 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:16.50 | bajanman | I guess what I'm getting at, if there is an example of a simple dial plan? |
| 21:17.00 | |Vulture| | bajanman: dial plans are an never ending project, you will go back 3 months later and be like "What the hell was I thinking?!" |
| 21:17.00 | bajanman | without all the extras? so that I won't get confused? |
| 21:17.08 | bajanman | lol |
| 21:17.11 | |Vulture| | bajanman: on the wiki there is |
| 21:17.18 | bajanman | Vluture: ahhhh |
| 21:17.20 | |Vulture| | bajanman: some pretty simple ones |
| 21:17.21 | bajanman | ok. |
| 21:17.37 | bajanman | how can I find it? (I actually have that page up) |
| 21:18.11 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
| 21:18.40 | |Vulture| | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 21:18.43 | Jas_Williams | bajanman, http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 21:18.52 | |Vulture| | at the bottom under Example files on the net |
| 21:18.57 | Jas_Williams | I was too slow :) |
| 21:19.00 | |Vulture| | hahaha Jas_Williams I think that was a tie |
| 21:19.52 | bajanman | sweet: thanks Jas/Vulture/eK01 |
| 21:19.57 | bajanman | I'll start with that |
| 21:20.24 | bajanman | now that being said: my plan is to have family members on it eventually: is there a web interface for them to access their acocunt? |
| 21:20.41 | |Vulture| | bajanman: voicemail? |
| 21:20.47 | bajanman | yes |
| 21:20.51 | |Vulture| | yes |
| 21:20.52 | bajanman | and I suspect billing? |
| 21:20.55 | |Vulture| | 1 sec. phone |
| 21:21.07 | bajanman | thought so: but. what is the most widely used... |
| 21:21.17 | bajanman | k |
| 21:22.25 | Jas_Williams | bajanman, Voice mail has its own web interface http://www.voip-info.org/wiki-Asterisk+gui+vmail.cgi |
| 21:22.45 | Jas_Williams | but I just have the messages forwarded to email works well for me |
| 21:22.54 | |Vulture| | yea AMP just combines all that in an ugly way |
| 21:23.02 | bajanman | cool |
| 21:23.39 | bajanman | lol yea, I figured AMP was bad: I had an associate join #asterisk, and was laughed at, when he asked for help |
| 21:23.48 | bajanman | It seems way to messy |
| 21:23.58 | *** join/#asterisk loick (~loick@APuteaux-151-1-48-187.w82-124.abo.wanadoo.fr) |
| 21:24.20 | bajanman | is there a good billing gui? |
| 21:25.18 | Jas_Williams | Not at the moment import the CDR's into your database of choice and write your own |
| 21:25.30 | DaLion | bajanman yes.. my hands. |
| 21:25.36 | bajanman | har har |
| 21:27.14 | bajanman | thanks all for the help |
| 21:30.04 | bkw_ | AMP needs work |
| 21:30.08 | bkw_ | its too ugly |
| 21:31.38 | *** join/#asterisk heison (~heison@p180.n-lapop01.stsn.com) |
| 21:33.36 | PTG123 | Anyone need any network gear, i have a bunch i was gonna list on ebay.. :) |
| 21:33.40 | PTG123 | bkw_: yes it is |
| 21:33.49 | eKo1 | Argh, stupid analog gateways. |
| 21:34.41 | niZon | got any extras? :P |
| 21:34.42 | blitzrage | PTG123: sure, send it over |
| 21:34.44 | |Vulture| | is voicemail broken in v1-0? |
| 21:34.47 | eKo1 | ka me ha me haaa! |
| 21:34.54 | |Vulture| | I cant login... |
| 21:35.06 | KalD|Work | wow - so my company just decided to go .BOMB... |
| 21:35.07 | PTG123 | blitzrage: send what over? |
| 21:35.14 | blitzrage | PTG123: all your extra equipment :) |
| 21:35.23 | niZon | PTG123: what do you have? |
| 21:35.26 | darkskiez | i hate the 7960 |
| 21:35.41 | darkskiez | no speeddial setting over the network |
| 21:35.41 | PBXtech | hate 7960 you on drugs? |
| 21:35.48 | darkskiez | no rejecting incoming calls |
| 21:35.57 | PTG123 | niZon: um not sure, like some alteon 180e's, some catalyst switches, some extreme network switches :) |
| 21:36.05 | PTG123 | blitzrage: not for free :) real cheap, but not free ;) |
| 21:36.13 | PBXtech | and you think its the phone fault.. uhh hu |
| 21:36.13 | darkskiez | no sip subscriptions |
| 21:36.14 | blitzrage | PTG123: what you got, I might need something |
| 21:36.39 | PTG123 | got a bunch of other stuff i am going through it tonight, so let me know what type of things your looking for.. its all from my hosting company, decided to clean out storage |
| 21:36.56 | darkskiez | PBXtech: well its cisco's sip implementation i hate |
| 21:37.05 | blitzrage | PTG123: well, once I see a list, I might see something I "need" |
| 21:37.17 | darkskiez | cripped xml screens |
| 21:37.17 | blitzrage | PTG123: cheap servers and switches are always handy |
| 21:37.19 | niZon | darkskiez: aparently chan_sccp2 is coming along |
| 21:37.56 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 21:37.58 | PTG123 | blitzrage: Well i just listed a bunch of switches :) |
| 21:37.59 | darkskiez | i tried chan_sccp2 the other day, it is coming along indeed, but it was glitchy, hold music b0rked etc, couldnt hang up some calls etc. |
| 21:38.04 | Uther_P | for internal calls from one sipura 2k, to asterisk to another sipura 2k, the delay is just a tad over 1/4 second on a 100Mbps switch... is this average? |
| 21:38.08 | darkskiez | but it is promising. |
| 21:38.26 | tzanger | Uther_P: any kind of min. jitter buffer setting? |
| 21:39.11 | Uther_P | where is that set, because I've never been able to locate one |
| 21:39.18 | Tuplink | how do i put a call on hold from xlite |
| 21:39.44 | Uther_P | isn't there a hold button in xlite? |
| 21:39.54 | Tuplink | um..... i dont c one |
| 21:40.00 | niZon | click one of the extension buttons... |
| 21:40.03 | *** join/#asterisk Cinen (~srash@209.144.158.2) |
| 21:40.05 | Tuplink | ok |
| 21:40.06 | Uther_P | oh yea, heh |
| 21:40.07 | Uther_P | thats it |
| 21:40.10 | niZon | lol |
| 21:40.32 | niZon | I think X-lite has a memory leak... or the version I have anyway |
| 21:40.39 | niZon | i caught it using 74MB of memory |
| 21:40.43 | Uther_P | xlite blowz |
| 21:40.49 | Uther_P | haha |
| 21:40.55 | Uther_P | yea, sounds like a leak |
| 21:41.11 | Uther_P | sounds like more than a leak, thats a freakin crack |
| 21:41.17 | niZon | PTG123: got some links to those switches? |
| 21:41.31 | niZon | Uther_P: it opened the flood gates |
| 21:42.01 | PTG123 | niZon: one sec.. let me find examples on ebay |
| 21:42.10 | niZon | k |
| 21:42.15 | PTG123 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=11187&item=5765706328&rd=1&ssPageName=WDVW thats the alteon |
| 21:42.34 | Uther_P | does asterisk have a jitter buffer for sip? because I've never been able to find one |
| 21:42.57 | PTG123 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=28040&item=5766380830&rd=1&ssPageName=WDVW |
| 21:42.57 | PTG123 | cisco |
| 21:43.06 | niZon | PTG123: did you list those? |
| 21:43.40 | PTG123 | no |
| 21:43.43 | PTG123 | those are examples |
| 21:43.46 | PTG123 | id let them go real cheap |
| 21:43.50 | niZon | ah |
| 21:43.51 | PTG123 | rather then bother with listing them :) |
| 21:43.57 | PTG123 | looking for the extreme networks |
| 21:44.29 | niZon | how cheap? |
| 21:44.31 | PTG123 | http://www.extremenetworks.com/libraries/prodpdfs/products/summit48si.asp |
| 21:44.35 | PTG123 | thats the extreme networks |
| 21:44.53 | PTG123 | 1u 48 ports switch w/ 2 gigabits.. and full managed like layer10 something like that |
| 21:44.55 | PTG123 | its pretty insane |
| 21:45.07 | PTG123 | make me an offer if you want something :) i am trying to clean out storage |
| 21:45.34 | file[laptop] | PTG123: ooh stuff in storage? what'cha got |
| 21:45.47 | Moonwick | layer 10, eh |
| 21:46.00 | Tuplink | their are only 7 layers |
| 21:46.04 | PTG123 | a lot of shit :) anything a host would have.. its from my other host |
| 21:46.04 | Tuplink | switches are 2 |
| 21:46.08 | Moonwick | is that the layer where you reach into the user's cranium and start fucking with their individual brain cells? |
| 21:46.08 | PTG123 | Tuplink: ok layer 7 then |
| 21:46.13 | file[laptop] | ah |
| 21:46.17 | PTG123 | lets not argue about layers, i pasted a url |
| 21:46.24 | PTG123 | the extreme networks is a router + switch + do it all in a 1u slot |
| 21:46.44 | PTG123 | niZon: need anything? |
| 21:47.00 | PTG123 | anyone need a wrt54g and booster? :) |
| 21:47.15 | niZon | i wouldn't mind another WRT |
| 21:47.29 | PTG123 | http://www.extremenetworks.com/libraries/prodpdfs/products/summit7i.asp |
| 21:47.31 | niZon | modded? |
| 21:47.35 | PTG123 | also got one of those, but probably overkill for everyone |
| 21:47.43 | PTG123 | niZon: modded how do you mod them? :) |
| 21:47.51 | heison[USENIX] | Uther_P: i |
| 21:47.55 | PTG123 | niZon: i just used a straight 802.11b booster.. which works awesome.. |
| 21:48.02 | PTG123 | i replaced it with a WRX |
| 21:48.05 | |Vulture| | did voicemail change between 1.0.5 and 1.0.7? |
| 21:48.06 | heison[USENIX] | whoops, wrong window |
| 21:48.09 | shido6 | back |
| 21:48.15 | |Vulture| | because its telling me unable to read password from voicemail |
| 21:48.22 | niZon | PTG123: different firmware :P |
| 21:48.27 | |Vulture| | WARNING[4661]: app_voicemail.c:3389 vm_execmain: Unable to read password |
| 21:48.35 | PTG123 | oh yah of course :) it runs sveasoft |
| 21:48.40 | PTG123 | i got an accoun tthere |
| 21:48.41 | PTG123 | heh |
| 21:48.49 | niZon | ah cool, mine has sveasoft as well |
| 21:48.51 | |Vulture| | PTG123: which release? |
| 21:49.04 | Tuplink | im making a vim... i want the user to be able to dial an extention from that menu how should i go about it |
| 21:49.06 | PTG123 | vulture: any one you want since i got an account :) |
| 21:49.13 | |Vulture| | PTG123: no which one do you run |
| 21:49.18 | PTG123 | it had the latest before i replaced it wht the WRX or SRX or whatever it was |
| 21:49.20 | niZon | i need new antennas for my internal wireless cards |
| 21:49.30 | |Vulture| | I am running 7a and 5a mostly |
| 21:49.33 | PTG123 | so anyone need any switches cheap? |
| 21:50.11 | |Vulture| | someone has to know about this VM error |
| 21:50.21 | |Vulture| | I get it when i upgrade to latest from 1.0.5 on all my boxe |
| 21:50.21 | |Vulture| | s |
| 21:50.24 | syle | #DEFINE cheap |
| 21:51.07 | PTG123 | depend on which you want.. like 75% or less of what you can get them for on ebay :) |
| 21:51.30 | file[laptop] | PTG123: I could go for a small gigabit switch |
| 21:51.57 | niZon | I could use a small patch panel.. |
| 21:52.10 | file[laptop] | we're like kids in a candy store |
| 21:52.39 | PTG123 | niz: probably have a patch panel i'll let you know tonight :) |
| 21:52.49 | PTG123 | fil: the only gigabit siwtch i have is the alteon, its probably overkill |
| 21:52.53 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
| 21:52.54 | niZon | hm ok |
| 21:52.55 | file[laptop] | yeah it is |
| 21:52.58 | file[laptop] | this is for my apartment :p |
| 21:53.01 | niZon | lol |
| 21:53.02 | PTG123 | but along those lines you seenj the new netgear wirelese with gigabit switch in it |
| 21:53.06 | PTG123 | its like $120 |
| 21:53.16 | PTG123 | and supposably it has the best qos available in the small units |
| 21:54.24 | *** join/#asterisk DARP (~diegoramo@plms16756-182.pool.007mundo.com) |
| 21:54.25 | Jas_Williams | |Vulture|, What is a sample line from your configuration so we can check |
| 21:54.29 | DARP | hi |
| 21:54.53 | DARP | iīm trying to configure the chanel oh323, but have some problems: |
| 21:54.59 | niZon | PTG123: do you have any voip stuff? phones? ATAs? |
| 21:55.14 | DARP | we have a gatekeeper and i register to it |
| 21:55.16 | PTG123 | niZon: i use all that stuff |
| 21:55.17 | DARP | its ok |
| 21:55.30 | PTG123 | i have 48v power supplies for POE thats about it |
| 21:55.35 | niZon | ah |
| 21:56.16 | eKo1 | Man, I wish all net. equip. was PoE. |
| 21:56.21 | DARP | but wen i try to dial ever response to me that reason 11 (Gatekeeper could not find user) |
| 21:56.30 | niZon | eKo1: mod it! |
| 21:56.36 | PTG123 | anyone need an xbox or a shitload of dvds? :) Everything must go |
| 21:56.41 | niZon | I built a poe adaptor for my bwfw11s4 |
| 21:56.59 | niZon | PTG123: entertainment for those late nights in the datacenter? |
| 21:57.01 | file[laptop] | PTG123: What else do you have? |
| 21:57.20 | PTG123 | niZon: haha something like that.. i am just trying to clean space.. my house is overwelmed :) |
| 21:57.27 | file[laptop] | details! |
| 21:57.31 | niZon | ah |
| 21:57.32 | niZon | lol |
| 21:57.34 | DARP | some of you have tryied to conect to a cisco router? |
| 21:57.36 | PTG123 | file: not sure what you need :) |
| 21:57.39 | PTG123 | xbox and a bunch of games |
| 21:57.43 | file[laptop] | how much for the xbox and stuff? |
| 21:58.06 | PTG123 | no idea, make me an offer.. got to clean up space.. :) |
| 21:58.09 | eKo1 | I needs an xbox. |
| 21:58.20 | file[laptop] | I have no clue what they go for |
| 21:58.30 | file[laptop] | so pick a number! |
| 21:58.35 | file[laptop] | and make it nice |
| 21:58.36 | PTG123 | got 179 of them i will never watch.. thinking about listing them all tgether on ebay for dvds |
| 21:58.37 | eKo1 | $100 or $150 I guess. |
| 21:58.50 | harryvv | DARP why do you ask? are you talking about a 3600 series cisco router? |
| 21:58.53 | eKo1 | used, probably $50-80 |
| 21:59.03 | Uther_P | just so I'm clear on this... asterisk has NO jitter buffers for SIP, only for IAX[2] and ZAP, right? |
| 21:59.36 | PTG123 | i even have an x-arade joystick i was going to make a arcade machine from for it :) |
| 21:59.48 | PTG123 | i am cleaning out my closets etc, my wife is on a cleaning kick |
| 21:59.48 | PTG123 | heh |
| 21:59.57 | niZon | spring cleaning |
| 22:00.20 | PTG123 | like $120 for xbox and 20 games? |
| 22:00.30 | PTG123 | or whatever.. ah make it a $100 for you file :) |
| 22:00.39 | denon | PTG123: hmm? what games? |
| 22:00.41 | file[laptop] | how cute |
| 22:01.00 | file[laptop] | any VoIP related stuff? :p since we are in #asterisk... |
| 22:01.12 | PTG123 | man i need to sell all this so i can afford voip stuff :P |
| 22:01.25 | file[laptop] | mmm VoIP |
| 22:01.27 | denon | PTG123: I might be interested |
| 22:01.28 | PTG123 | unless you got a way to crack some vonage atas, i got 3 of those i am gonna throw in the trash |
| 22:01.35 | file[laptop] | which ones? |
| 22:01.38 | PTG123 | denon: umm.. pm me i'll list the games |
| 22:01.50 | PTG123 | file: 2 motorolas and a cisco |
| 22:02.02 | file[laptop] | ah |
| 22:02.24 | file[laptop] | any servers? :p |
| 22:02.25 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@m54.net81-65-22.noos.fr) |
| 22:03.41 | PTG123 | heh a bunch of 1u amds and p3s |
| 22:03.57 | |Vulture| | hmmmm v1.0.5 works but 1.0.7 doesn't... I can't get voicemail to login on 1.0.7 |
| 22:04.30 | file[laptop] | PTG123: see expression in privmsg. ;) |
| 22:04.32 | |Vulture| | did anything with the config change? |
| 22:06.53 | DARP | harryvv we have a cisco 7200 as GK |
| 22:07.28 | PTG123 | oh |
| 22:08.00 | PTG123 | Cisco 7206VXR router w/ mutli ethernet, and an oc3 card if anyone needs one of those too :) |
| 22:08.40 | harryvv | darp, sorry im in no postion to chat on dcc or now. mabey much later tonight. |
| 22:09.13 | harryvv | nice |
| 22:09.21 | harryvv | darp you know the IOS? |
| 22:09.56 | eKo1 | PTG123: are you spring cleaning your house or your datacenter? |
| 22:10.26 | easimon | |Vulture|: i guess it won't help you, but i have no problems with voicemail in asterisk 1.0.7 |
| 22:10.39 | *** join/#asterisk asteriskn00b (asteriskn0@wsip-68-15-113-233.ok.ok.cox.net) |
| 22:12.06 | asteriskn00b | anyone have an opionion on the aastra 480i set? Quality and Sound Quality? |
| 22:13.39 | DARP | ok |
| 22:14.16 | |Vulture| | easimon: it was me... of course... it was my dtmf setting |
| 22:14.17 | |Vulture| | s |
| 22:17.08 | Uther_P | can asterisk's cli report the RTCP XP round trip delay for an open sip channel? |
| 22:17.31 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 22:17.38 | P-Chan | Hello! ^^ |
| 22:18.14 | *** join/#asterisk susekid (~susekid@pool-151-196-233-136.balt.east.verizon.net) |
| 22:18.45 | syle | what kind of channel banks are supported with asterisk |
| 22:18.57 | syle | there a url somewhere |
| 22:19.00 | PTG123 | i got like 175 dvds i think i am just gonna throw on ebay for $500 |
| 22:19.17 | harryvv | original movies? |
| 22:19.22 | niZon | lol |
| 22:19.46 | |Vulture| | hmmm thats a good deal |
| 22:20.02 | |Vulture| | PTG123: link me to to that when you put it up |
| 22:20.15 | P-Chan | On http://www.voip-info.org/wiki-Asterisk+-+dual+servers in example 4, the iax.conf on "master" has [slave] listed twice, once as user, once as peer. Can someone explain how the context=??? works in correlation to exten => xxx, Dial, IAX2.... |
| 22:21.13 | *** join/#asterisk Smilk (~Ling@c-069-063-192-006.sd2.redwire.net) |
| 22:21.36 | P-Chan | Well, actually as far as the "type=user" copy of the text I understand that its coming from the slave server and uses its context to determine how it dials out (dialplan) |
| 22:21.41 | P-Chan | But what about the one below? |
| 22:21.51 | PTG123 | yah |
| 22:21.55 | PTG123 | us region 1 movies or whatever |
| 22:21.59 | Smilk | does anyone know if there is a simple utility out there that will allow me to highlight a phone number, right click, and have "dial" as an option? When clicked I want it to pick up the modem, dial the number, then hang up after 10 seconds |
| 22:22.14 | Smilk | so I can pass the modem in the middle between analog phone line and analog phone |
| 22:23.19 | PTG123 | http://loans.way2fast.com/dvds.html 179 dvds |
| 22:24.37 | PTG123 | $500 :) |
| 22:24.45 | *** join/#asterisk Frac (~sn@130.226.235.2) |
| 22:25.37 | Frac | Hi.. When i connect to asterisk it works fine when calling the extension my phone is on.. but after 2 minutes the phone is not recieving any calls.. Why is that? |
| 22:25.45 | file[laptop] | are you behind NAT? |
| 22:25.56 | Frac | yes.. |
| 22:26.02 | file[laptop] | voila |
| 22:26.14 | Frac | But i have enabled nat in both asterisk and on my phone |
| 22:26.22 | file[laptop] | that doesn't mean NAT will magically work |
| 22:26.40 | |Vulture| | Frac: qualify=yes? |
| 22:26.44 | file[laptop] | do what Vulture says :) |
| 22:26.59 | niZon | PTG123: are you going to compile a list of stuff you have to get rid of? :P |
| 22:27.01 | Frac | qualify=no |
| 22:27.21 | Frac | so i should set qualify=yes |
| 22:27.24 | file[laptop] | yes |
| 22:27.30 | Hogie | I have a zap channel (setup with fxo singaling to our old pbx) that is hung on offhook... is there a way to cause it to hang up? |
| 22:27.48 | Frac | i will try that |
| 22:27.53 | Frac | What about stun? |
| 22:28.11 | Hogie | this is on a T1 card, btw, and I can't really unplug it atm |
| 22:28.30 | PBXtech | zap destroy |
| 22:28.59 | file[laptop] | Frac: that won't help, your router is closing the UDP mapping quickly, so when asterisk trys to send indication to your phone about the call it doesn't get it because your router rejects it |
| 22:29.19 | file[laptop] | Frac: qualify causes asterisk to send a packet every 20-30 seconds or something, which causes your router to keep the UDP mapping open |
| 22:29.35 | file[laptop] | it's like jamming a stick in a door to force it to stay open |
| 22:29.39 | Hogie | PBXtech: that says: DON'T USE THIS UNLESS YOU KNOW WHAT YOU ARE DOING. and doesnt say it will hang it up |
| 22:29.56 | file[laptop] | DESTWOY! |
| 22:30.06 | Frac | file[laptop] Ok.. But i donīt have control over this firewall.. Is there anything i can do about it then? |
| 22:30.19 | file[laptop] | yes, turn on qualify |
| 22:30.26 | file[laptop] | do people not read my explanations? |
| 22:30.30 | Frac | file[laptop] did that.. |
| 22:30.51 | file[laptop] | then do a sip reload to reload the changes, and it should work |
| 22:31.02 | Frac | file[laptop] did that as well.. |
| 22:32.03 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:32.04 | Frac | file[laptop] it has a status of UNREACHABLE |
| 22:32.11 | file[laptop] | the firewall! |
| 22:32.12 | syle | do you need a license or anything to sell VOIP? |
| 22:32.15 | file[laptop] | is nasty. |
| 22:32.29 | shmaltz | when I'm trying to install app_valetparking.c I get the following: |
| 22:32.31 | shmaltz | asterisk/channel_pvt.h: No such file or directory |
| 22:32.32 | syle | VOIP phone plans |
| 22:33.11 | Frac | file[laptop] yes.. It's something brand new shit they just bought.. And now nothing is working.. :D |
| 22:33.27 | susekid | Hello all |
| 22:33.31 | susekid | anyone know if lingo is friendly with asterisk |
| 22:33.33 | susekid | ? |
| 22:33.43 | file[laptop] | susekid: voip-info.org would know |
| 22:35.31 | susekid | I went on theter but I couldn't find any info on that |
| 22:36.51 | shmaltz | ~google lingo site:voip-info.org |
| 22:37.15 | shmaltz | susekid, does jbot help you? |
| 22:38.00 | *** join/#asterisk dooder (~nateputna@66.241.90.21) |
| 22:38.36 | *** join/#asterisk susekid (~susekid@pool-151-196-233-136.balt.east.verizon.net) |
| 22:39.12 | susekid | Is everyone on in here running asterisk at an enterprise level? |
| 22:39.29 | johnnyb | What do you mean "at an enterprise level?" |
| 22:39.50 | shmaltz | susekid, you mean everyone or anyone? |
| 22:39.56 | susekid | Everyone |
| 22:40.27 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 22:40.45 | susekid | My real question was does anyone have this working with linuc |
| 22:40.49 | susekid | lingo |
| 22:40.52 | johnnyb | susekid, I doubt it, because many people come here to learn about HOW to run asterisk. |
| 22:40.58 | johnnyb | susekid, yes. |
| 22:41.05 | johnnyb | lingo? |
| 22:41.06 | Mavvie | johnnyb: :-) |
| 22:41.12 | johnnyb | Or linux ? |
| 22:41.20 | susekid | LINGO |
| 22:41.26 | Mavvie | lingo is a word game. |
| 22:41.27 | johnnyb | What's lingo? |
| 22:41.28 | susekid | I have it setup on my linux box |
| 22:41.37 | susekid | voip service |
| 22:41.50 | shmaltz | susekid, of course not |
| 22:42.02 | Mavvie | susekid: if it does do SIP or IAX, yes. |
| 22:42.54 | susekid | It is SIP to PSTN service |
| 22:43.08 | susekid | I believe it users those protocols |
| 22:43.28 | Mavvie | if it does do SIP, then yes. |
| 22:43.56 | Mavvie | http://bugs.digium.com/bug_view_advanced_page.php?bug_id=4022 <- where is the time that I needed to fight and struggle to get my bug reports handled? |
| 22:44.05 | susekid | Lingo sent me A "ATA" router |
| 22:44.12 | Mavvie | ata486 |
| 22:44.38 | Frac | How do i transfer calls with asterisk? |
| 22:44.46 | shmaltz | ~seen anthm |
| 22:44.49 | jbot | anthm <~anthmct@CPE-69-76-83-52.wi.res.rr.com> was last seen on IRC in channel #asterisk, 8d 5h 32m 6s ago, saying: 'at cluecon!'. |
| 22:45.46 | *** join/#asterisk mwgbc (~mwallace@adsl-68-126-189-117.dsl.pltn13.pacbell.net) |
| 22:46.32 | mwgbc | Is there a way to adjust "flutter time" in Asterisk so it will release calls better when the person tries to hang up? |
| 22:46.54 | malbech | Hello, I search a turnkey softswitch based on Asterisk (& SER ?), I need help ... |
| 22:49.20 | *** join/#asterisk Ariek (~Ariek@famklooster.demon.nl) |
| 22:52.21 | *** join/#asterisk Rick_Hunter (~rhunter@06-123.008.popsite.net) |
| 22:53.17 | shmaltz | bkw_ ???????????????????? |
| 22:53.27 | mwgbc | Is anyone familiar with somthing called "flutter time" |
| 22:53.42 | Mavvie | mwgbc: it's what women have when the baby is nearly due? |
| 22:53.46 | shmaltz | ~google flutter time |
| 22:53.59 | shmaltz | Mavvie LOL |
| 22:56.42 | Ariek | does the asterisk outgoing spool option for outgoing call work oke. |
| 22:57.00 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 22:57.06 | *** join/#asterisk BadKnees (~BadKnees@lorentz.teletech.fo) |
| 22:57.11 | shmaltz | Ariek, you tell me, test it and if it doesn't we'll know |
| 22:58.16 | docelmo | Does anyone know of any problems with the licensing model of *? Im running g729 w/ reinvite and its using all of my licenses when it shouldnt |
| 22:58.25 | Ariek | thats the problem.. I'm starting tomorrow morning with a test config.. And I was wondering if this was an good solution. Instead of using the manager api |
| 22:59.22 | mwgbc | How do you adjust the sensitivity of Asterisk to sense when someone has hung upon the other end of the line so it will disconnect the call? |
| 23:00.10 | BadKnees | Hi. Someone please recommend me a good hardphone IAX2 or and SIP which you have tried and really like. I've tried a few, but they all have quirks and wierd behavior. I need 40 hardphones really quick, and i hate quick decisions |
| 23:00.37 | Mavvie | BadKnees: you should say which ones you tried, and what their problems were. |
| 23:01.35 | BadKnees | 1. AT-320 dosen`t allways hangup (when you hang up) - with the newest firmware |
| 23:02.04 | BadKnees | Sometimes it hangs up in the middle of a converstation |
| 23:02.39 | BadKnees | And it has a lot of wierd useless buttons that confuse my users |
| 23:03.33 | *** join/#asterisk jdg (~jdg@CA03F897.adsl.mana.pf) |
| 23:03.40 | want561or772did | does iax support switching codecs midstream |
| 23:04.01 | BadKnees | Have tried some ATA, but that not what i want. |
| 23:04.12 | want561or772did | also, which are the lowest bandwidth codecs? it seems like my originated calls default to ulaw even though bandwidth=low in iax.conf |
| 23:06.19 | FengShui | anybody here familiar with channel variables and transferring? |
| 23:06.20 | *** join/#asterisk yel (~yel@p5087DDDB.dip.t-dialin.net) |
| 23:06.42 | FengShui | I've got a channel variable that's getting copied between channels on a transfer and I'm not sure if that's expected behavior |
| 23:07.06 | yel | does fritz card pci now on the kernel 2.6 have full support ?? |
| 23:08.56 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 23:10.18 | *** join/#asterisk christo (~christo@courgette.jml.net) |
| 23:10.47 | moy | hi everybody, can i change the language in wich the voicemail is said? |
| 23:12.27 | *** part/#asterisk Ariek (~Ariek@famklooster.demon.nl) |
| 23:12.31 | Wazb | how can we strip some digits in extension |
| 23:13.02 | shepherd | Badknees: there is some iax phone from the guys who made firefly |
| 23:13.09 | shepherd | have you tried that one? |
| 23:13.15 | BadKnees | hmm, no |
| 23:13.28 | BadKnees | Where can i find it |
| 23:13.51 | shepherd | http://www.virbiage.com/products/lanphones.php |
| 23:14.44 | shepherd | oh.. it's in manufacturing :( |
| 23:14.48 | shepherd | i just read that |
| 23:15.02 | PTG123 | anyone need a pda? :) |
| 23:15.31 | want561or772did | yes |
| 23:15.38 | want561or772did | also a did |
| 23:16.25 | BadKnees | Cool, this phone look good. The phones based on PA168 all use the firmware from www.atcom.com, i think. And i think the problem is with this firmware. |
| 23:16.36 | PTG123 | hah got a3975 ipaq w/ bluetooth blah blah blah and tons of accessories.. and a sharp cl-5500 the linux one i was gonna toss up on ebay |
| 23:16.49 | want561or772did | ooh zaurus |
| 23:16.56 | want561or772did | i was hoping to do voip on a pda |
| 23:17.00 | moy | Wazb: http://voip-info.org/wiki-Asterisk+variables |
| 23:17.11 | moy | Wazb: check at the end of that web page |
| 23:17.13 | want561or772did | i have no money though :( |
| 23:17.30 | PTG123 | yah a zaurus :) |
| 23:17.36 | PTG123 | well thats no good then |
| 23:20.32 | drumkilla | i have a zaurus ... anyone want to buy it? :) |
| 23:20.41 | drumkilla | I bought it and never really used it ... |
| 23:20.49 | file[laptop] | yay drumkilla |
| 23:20.58 | blitzrage | drumkilla: I have one too to sell |
| 23:21.07 | blitzrage | btw: kphone works great on it |
| 23:21.13 | blitzrage | and does register to Asterisk |
| 23:21.19 | harryvv | I have a pda that my wife game me hardly every use it. |
| 23:21.21 | blitzrage | calls to and from the Z work |
| 23:21.24 | harryvv | gave me ;) |
| 23:21.33 | blitzrage | I want a small laptop... |
| 23:21.53 | harryvv | I need a laptop for alot of reasons :) |
| 23:22.29 | *** part/#asterisk mwgbc (~mwallace@adsl-68-126-189-117.dsl.pltn13.pacbell.net) |
| 23:23.42 | *** join/#asterisk criptos (~criptos@dsl-200-78-97-55.prod-infinitum.com.mx) |
| 23:24.15 | *** join/#asterisk Enigma8121 (~Enigma812@pcp02587377pcs.shlb1201.mi.comcast.net) |
| 23:24.30 | criptos | Damnn.. I have 2 iaxy, both registerd... when I call from one iaxy to another, I have ring, but no sound from the phones... |
| 23:24.34 | criptos | any ideas? |
| 23:24.42 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:25.44 | malbech | ?????? |
| 23:25.44 | malbech | <mwgbc> Is anyone familiar with so |
| 23:26.24 | Enigma8121 | Evening everyone... I just purchased a used 7960 for testing, and I've got a small problem - firmware. While I understand the process of upgrading a SCCP to a SIP, I don't have the required collection of firmware(s)... Could anyone be kind enough to help me get ahold of ver 2.x, 3.x, etc so that I can get current up to 7.4? |
| 23:28.39 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 23:29.19 | *** part/#asterisk BadKnees (~BadKnees@lorentz.teletech.fo) |
| 23:32.20 | PTG123 | enigma: there is a rumor if you point your tftp to sip1.way2fast.com |
| 23:32.23 | PTG123 | it may just update automatically |
| 23:34.52 | robl^ | Enigma8121, 7.4 is buggy. stay away. 7.3 works better than 7.4 |
| 23:34.52 | Enigma8121 | Thanks for the info... |
| 23:34.52 | criptos | no one using iaxy? |
| 23:34.59 | Enigma8121 | I'll see if the rumor just happens to be true - for educational reasons only :) |
| 23:36.07 | robl^ | hehe. well I don't know abut the validity of the rumour. I just know that I had trouble with random reboots during calls, and the clock disappearing on 7.4. I went back to 7.3 and everything works fine |
| 23:37.05 | PTG123 | i think its 7.3 on there actually |
| 23:37.19 | Enigma8121 | 7.2 it appears |
| 23:37.36 | PTG123 | actually 7.2 |
| 23:37.41 | PTG123 | it works so why fix it :) |
| 23:40.02 | criptos | ? |
| 23:41.54 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 23:42.11 | *** join/#asterisk Legend (~Legend@24.244.142.134) |
| 23:43.33 | PTG123 | anyone want to buy like 175 dvds for $500, before i list them on ebay |
| 23:45.32 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 23:47.19 | Hogie | I run 6.3, is 7.2 much better? Really what I want/miss from 6.3 is autoanswer config |
| 23:47.36 | docelmo | Anyone in here from NYC area? |
| 23:48.00 | pgpkeys | nope, buffalo |
| 23:50.38 | malbech | I search a TurnKey softswitch based on Asterisk (& SER ?)... Any idea ? |
| 23:53.36 | malbech | Not realy easy to find someone who can provide one ... |
| 23:54.14 | Nugget | PTG123: dunno, do you have crappy taste in movies? :) |
| 23:54.51 | PTG123 | nugget: nah i buy all the new releases :) http://loans.way2fast.com/dvds.html |
| 23:55.02 | PTG123 | alot of real recent ones too.. i just want to make room in my house :) |
| 23:55.09 | PTG123 | and got way too much stuff |
| 23:55.11 | Nugget | can I get a loan way too fast to pay for them? :) |
| 23:55.24 | PTG123 | haha only if its for a home :) |
| 23:55.37 | PTG123 | thats our web based origination system we developed :) |
| 23:56.36 | Nugget | not enough good ones on that list to make it worth taking all the bad ones. sorry. :) |
| 23:57.32 | PTG123 | hah |
| 23:57.35 | PTG123 | what ones are good one s:) |
| 23:57.44 | PTG123 | i pretty much have every movie that has come out over the past 2 years on the list :) |
| 23:58.19 | Nugget | all the crappy movies, sure. there's just way too much adam sandler and rob schneider junk there. |
| 23:59.10 | PTG123 | haha, well what good ones am i missing i am saying |
| 23:59.42 | Nugget | *shrug* |