| 00:10.07 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 00:11.03 | SeRi | well... |
| 00:11.24 | *** join/#asterisk rpluto (~rpluto@a95-92-60-159.cpe.netcabo.pt) |
| 00:11.34 | *** part/#asterisk rpluto (~rpluto@a95-92-60-159.cpe.netcabo.pt) |
| 00:11.49 | *** join/#asterisk rpluto (~rpluto@a95-92-60-159.cpe.netcabo.pt) |
| 00:13.43 | SeRi | p3nguin: you in? |
| 00:16.25 | TheCops | complete idle :) |
| 00:19.01 | SeRi | lol |
| 00:20.42 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 00:24.13 | *** join/#asterisk bintut (~bintut@cm61.sigma15.maxonline.com.sg) |
| 00:37.33 | scubes13 | having issue every now and then where user making outbound call cannot hear a ring tone, gets silence…. |
| 00:37.47 | scubes13 | on the other end, the recipient is picking up and hears nothing as well.... |
| 00:38.00 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 00:38.04 | scubes13 | guessing it may be firewall related, but not sure where/how to begin to track down |
| 00:38.17 | scubes13 | 11 internal extensions on client network |
| 00:38.23 | scubes13 | pbx is hosted "cloud" |
| 00:39.24 | xpot-mobile | running ver. 1.8.7.1 on CentOS 5.7, when a user dials 555 for spy, sometimes the channel will lock. I am unable to perform a channel request hangup and the only way to close the channel is to stop services. Is this a known bug? any ideas how to resolve this? Appears to only happen with 555 spy extension. |
| 00:39.48 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 00:42.26 | thebitguru | scubes13: I am using SIP and the server is on the same network, behind a firewall. I am pretty sure that my issue is related to the firewall because things worked OK without the firewall |
| 00:43.18 | scubes13 | thebitguru - ok… ours is hosted externally… our issue may be firewall as well, but guessing probably not the same… :( |
| 00:43.35 | thebitguru | what firewall are you using? |
| 00:43.45 | scubes13 | we are using pfsense 2.0 |
| 00:44.01 | scubes13 | with sipproxyd |
| 00:44.03 | thebitguru | on a custom box or some appliance? |
| 00:44.12 | scubes13 | on a custom box |
| 00:44.25 | thebitguru | I see. |
| 00:44.27 | SeRi | scubes13: I use pfsense 2.0 Relase and I have an asterisk box behind it |
| 00:44.54 | SeRi | Did you enable AON? |
| 00:45.00 | SeRi | scubes13: ^^ |
| 00:45.08 | scubes13 | our asterisk box is not behind the firewall - it is hosted from a datacenter with public ip… our phones are behind the firewall |
| 00:45.36 | scubes13 | I had it enabled, but removed when I set sipproxyd |
| 00:45.36 | SeRi | ok are you using siproxd? |
| 00:45.36 | SeRi | Ok |
| 00:45.36 | scubes13 | yes, siproxd (sorry) |
| 00:46.11 | scubes13 | SeRi - should we be using both? |
| 00:46.48 | SeRi | not really. If you are using siproxd you should be ok. Though the discribed behavior sounds like a nat issue |
| 00:47.10 | SeRi | Try enabling AON and see how it works out for you. |
| 00:47.38 | scubes13 | just enable it? anything else really needed beyond just that? |
| 00:47.49 | SeRi | Not really. |
| 00:48.04 | SeRi | AON is the onlything you have to enable when using SIP behind pfsense |
| 00:48.29 | SeRi | because pfsense rewites the port and sip does not like that |
| 00:48.44 | SeRi | s/rewites/rewrites/ |
| 00:49.10 | scubes13 | so need to enable "Static Port"? |
| 00:49.21 | SeRi | infobot: had a loong week end I am sure.... |
| 00:49.33 | SeRi | scubes13: Yes |
| 00:50.33 | *** join/#asterisk troyt (~troyt@2001:5c0:1000:b::a06b) |
| 00:51.36 | scubes13 | anything I might also look in logs for that would help point to NAT issue? |
| 00:52.15 | SeRi | sip set debug on and core set verbose 3 |
| 00:52.35 | SeRi | use those two when the issue happen again to capture it. |
| 00:52.42 | SeRi | *IF* |
| 00:52.47 | SeRi | :) |
| 00:53.15 | scubes13 | ok |
| 00:53.53 | scubes13 | verbose was set to 10 originally |
| 00:54.03 | scubes13 | sip debug wasnt on |
| 00:55.21 | SeRi | you shouldnt need 10. 3 should sufice |
| 00:55.45 | SeRi | come back with a PB of the info *IF* it happens again |
| 00:55.59 | scubes13 | def will do |
| 00:56.02 | scubes13 | thanks so much! |
| 00:57.39 | SeRi | your welcome |
| 00:57.41 | SeRi | cya |
| 00:57.57 | scubes13 | with 10 extensions, and no active calls… should I see a lot of sip traffic at the moment? |
| 00:59.10 | *** join/#asterisk mducharme-work (~nothing@206.188.121.4) |
| 01:00.46 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-xjaapnzolluuodaz) |
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| 01:01.41 | SeRi | scubes13: define "a lot of traffic" |
| 01:02.05 | SeRi | If the phones are not been used you should not see anything going on. |
| 01:02.32 | scubes13 | well, looks like a bunch of SIP Timers… stopping retransmision…. Destroying SIP dialog… Really destroying SIP dialog.... |
| 01:02.36 | SeRi | scubes13: core show channels |
| 01:02.50 | SeRi | scubes13: Thats sip debug |
| 01:02.57 | SeRi | and thats all normal information |
| 01:03.09 | scubes13 | ok, kewl deal… |
| 01:03.22 | scubes13 | core channels showed none active :) |
| 01:03.26 | SeRi | core show channels will show you active calls and thats real sip traffic |
| 01:03.37 | scubes13 | gotya |
| 01:03.51 | SeRi | The rest is registration/validation/etc.... |
| 01:04.05 | SeRi | you wont see it unless you have logger configure for it |
| 01:05.00 | scubes13 | ok, was just curious more so if I was seeing a lot of that b/c maybe my nat was doing something.. like causing the phones to consistently reregister or something |
| 01:05.09 | scubes13 | totally out of his depth |
| 01:05.37 | SeRi | :) |
| 01:05.59 | scubes13 | am trying to learn though ;) |
| 01:06.22 | scubes13 | for now, I will leave it be and have the users let me know when they start having probs next |
| 01:06.26 | scubes13 | *IF* |
| 01:06.29 | scubes13 | ;) |
| 01:06.34 | scubes13 | thanks again! |
| 01:06.40 | SeRi | :P |
| 01:06.47 | SeRi | Take care and good luck |
| 01:06.52 | SeRi | your welcome |
| 01:12.00 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 01:13.04 | s[X] | woot Seri, got my email from freenum |
| 01:16.56 | SeRi | s[X]: sick! |
| 01:16.59 | SeRi | :) |
| 01:17.10 | SeRi | very nice |
| 01:17.24 | s[X] | just trying to work out how to set it up in freepbx |
| 01:17.34 | SeRi | :/ |
| 01:20.02 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 01:20.32 | SeRi | does not know freepbx |
| 01:24.32 | *** part/#asterisk honree (~honree@net2.icemans.co.uk) |
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| 01:34.19 | hesco_home | Evening all: I'm seeking recommendations for an affordable ITSP for BC Canada DIDs and call termination primarily in the Southern Interior of BC. Any guidance? |
| 01:47.12 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:47.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:51.40 | p3nguin | FreePBX? Why would anyone use FreePBX? Especially for something so trivial as configuring Asterisk. |
| 01:52.18 | SeRi | p3nguin: !!!!!!!!!! |
| 01:52.28 | SeRi | I got my psu :) |
| 01:53.05 | SeRi | p3nguin: did the script at least give you some ideas? |
| 01:53.09 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 02:10.02 | *** part/#asterisk rpluto (~rpluto@a95-92-60-159.cpe.netcabo.pt) |
| 02:11.44 | WIMPy | Hmm. Is it really that hard to copy a few bytes of samples from a TE407P that a bigger number of used channels generates IRQ misses? |
| 02:22.37 | *** join/#asterisk master_of_master (~master_of@p57B54102.dip.t-dialin.net) |
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| 03:06.11 | dijib | sup bitches? |
| 03:11.55 | Kobaz | yeap |
| 03:14.06 | SeRi | :/ |
| 03:24.49 | *** join/#asterisk LostyJai (~blah@202.171.190.130) |
| 03:24.50 | LostyJai | hey |
| 03:24.52 | LostyJai | when a channel is active, it does not necessarily mean it's calling in/out, right? |
| 03:26.13 | p3nguin | Calling in or out is a human concept. All asterisk knows is that there is a channel in use. |
| 03:26.42 | LostyJai | sometimes we're getting "all circuits are busy" |
| 03:26.46 | LostyJai | just wasn't too sure why |
| 03:27.02 | p3nguin | How many active channels do you have? |
| 03:27.11 | LostyJai | 10/100 atm |
| 03:27.22 | LostyJai | now 12 |
| 03:27.25 | p3nguin | 10 out of 100 available? |
| 03:27.27 | dijib | i so need a wakup call |
| 03:27.36 | LostyJai | do you mean from my sip provider |
| 03:27.41 | LostyJai | i mean VOIP provider |
| 03:27.47 | LostyJai | or my asterisk server? |
| 03:27.51 | p3nguin | I just mean how many channels are active. |
| 03:27.56 | dijib | p3 what do you think about my inability to run custom .so's |
| 03:27.58 | LostyJai | says 12 now |
| 03:28.12 | p3nguin | What are those 12 channels doing? |
| 03:28.22 | p3nguin | Calls between phones and ITSP? |
| 03:28.43 | LostyJai | some on calls |
| 03:29.22 | p3nguin | 12 channels is probably 6 calls. |
| 03:29.31 | LostyJai | how can i tell what the others are doing |
| 03:29.33 | LostyJai | sip show channels ? |
| 03:29.42 | p3nguin | core show channels will show channels/calls. |
| 03:30.03 | LostyJai | 9 active channel, 6 active calls |
| 03:30.53 | p3nguin | What are the circumstances surrounding the all circuits busy message? |
| 03:31.04 | LostyJai | let me see |
| 03:31.06 | p3nguin | Calls out to the PSTN via ITSP? |
| 03:31.25 | WIMPy | And wehere does the message come from? |
| 03:31.53 | p3nguin | Is it a voice message or a message printed on the console? |
| 03:31.59 | p3nguin | NEED DETAILS |
| 03:32.43 | SeRi | waz up p3nguin |
| 03:32.46 | LostyJai | yep i know |
| 03:32.51 | LostyJai | checking now >< |
| 03:33.04 | LostyJai | it's when registering with the provider |
| 03:33.23 | LostyJai | [Dec 6 09:54:48] VERBOSE[21357] logger.c: [Dec 6 09:54:48] -- SIP/PROVIDER-08afb090 is circuit-busy |
| 03:34.38 | LostyJai | http://pastebin.com/jfyyjuBb |
| 03:34.59 | LostyJai | http://pastebin.com/raw.php?i=jfyyjuBb |
| 03:35.23 | p3nguin | When you have the message, check "sip show registry" to see if you are registered. |
| 03:35.33 | LostyJai | ok cheers |
| 03:35.58 | p3nguin | My ITSP requires that I am registered before I can send calls in to them. |
| 03:36.05 | p3nguin | Several behave this way. |
| 03:36.24 | LostyJai | doesn't it show in the asterisk log if you are no longer registered? |
| 03:36.52 | p3nguin | If you have configured logger to show it, maybe. |
| 03:38.03 | WIMPy | It can't know if you're registered. It can only tell you when it fails to register. |
| 03:38.38 | LostyJai | yeah i think it disconnected |
| 03:38.43 | p3nguin | And maybe if you become unregistered? |
| 03:38.48 | LostyJai | because registration time was 2minutes ago |
| 03:38.54 | SeRi | :/ |
| 03:38.58 | WIMPy | But the only way to find out what's going on it to enable sip debug and search for the message. |
| 03:39.00 | p3nguin | If it says "Registered" then it's fine. |
| 03:39.14 | LostyJai | refresh is 105 |
| 03:39.19 | LostyJai | does it reconnect every 105 seconds? |
| 03:39.30 | p3nguin | Reconnect, no. Reregister, yes. |
| 03:39.45 | LostyJai | "reg time" just updated |
| 03:39.47 | LostyJai | that's normal? |
| 03:39.51 | p3nguin | Yes. |
| 03:39.56 | p3nguin | (2139.00) <p3nguin> If it says "Registered" then it's fine. |
| 03:39.56 | LostyJai | okay |
| 03:40.17 | p3nguin | So that's probably not the issue. |
| 03:40.30 | LostyJai | alright i'll keep an eye out |
| 03:40.38 | p3nguin | How many channels did they allow you? |
| 03:40.42 | LostyJai | i think 15 |
| 03:41.15 | p3nguin | Any chance you're using all 15 already when the message appears? |
| 03:41.25 | LostyJai | i doubt it |
| 03:41.42 | LostyJai | wait wait.. |
| 03:41.49 | LostyJai | that limit.... is that active CALLS or active CHANNELS? |
| 03:42.27 | p3nguin | When you say limit, are you talking about the number of channels allowed by the ITSP? |
| 03:43.37 | p3nguin | If so, it's channels between your PBX and theirs. That's potentially 30 active channels on your PBX and 15 calls involving the ITSP. |
| 03:43.39 | SeRi | p3nguin: I configured an ivr for my brother today. My first setup :) I converted the wav file to ulaw using sox. I am proud of my self :P |
| 03:43.58 | p3nguin | You could have just used "file convert" in the asterisk cli. |
| 03:44.08 | p3nguin | Much easier. |
| 03:44.38 | p3nguin | file convert myfile.wav myfile.ulaw |
| 03:44.45 | SeRi | o well. At least I got it done. |
| 03:45.09 | SeRi | Now I know I can do it on the cli |
| 03:46.45 | p3nguin | I'm so tired of aches and pains. |
| 03:47.11 | SeRi | I can tell. You been quite.... |
| 03:47.15 | p3nguin | I've had the worst headache all day... on top of the horrible back pain. |
| 03:47.24 | SeRi | That sucks. |
| 03:47.25 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:47.39 | p3nguin | I didn't even start work until around 3 pm. |
| 03:48.01 | SeRi | :( |
| 03:48.10 | ChannelZ | Perfect time to go to the range! |
| 03:48.25 | SeRi | I been in pain but not as bad.... You got me beat by a long run. |
| 03:48.31 | p3nguin | I took Tylenol for my head as soon as I got up, and it made me sick to my stomach. |
| 03:48.51 | p3nguin | So then I had to wait for that to wear off. |
| 03:49.15 | p3nguin | I finally got around to eating lunch about 4:30. |
| 03:49.35 | p3nguin | I'm falling apart. I sure hope it's temporary. |
| 03:50.24 | dijib | sounds like fun |
| 03:50.36 | p3nguin | The good news is that I hardly notice the pain in my side that I thought was my liver. |
| 03:50.52 | dijib | tylonal can damage the liver |
| 03:51.02 | p3nguin | I don't take THAT much of it. |
| 03:51.06 | SeRi | p3nguin: It will all pass. |
| 03:51.07 | dijib | especially in combination with alcohol |
| 03:51.17 | p3nguin | I've has half my allowance of it for the day. |
| 03:51.18 | dijib | excersise |
| 03:51.23 | dijib | stop eating fast food |
| 03:51.38 | p3nguin | It's only fast because I cook quickly. |
| 03:51.43 | SeRi | hahaha |
| 03:52.35 | p3nguin | I only have fast foods once every couple of weeks at the most. |
| 03:53.03 | SeRi | only fast food I eat here and there is pizza. |
| 03:53.26 | p3nguin | Pizza is all the food groups. |
| 03:53.37 | SeRi | :) |
| 03:54.11 | SeRi | Pizza=nom nom nom! |
| 03:54.45 | p3nguin | I usually don't get any vegetables on my pizza, though. I don't like mushrooms, and I'm not a fan of onions that much. So I guess pizza for me is most of the food groups. |
| 03:54.58 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:55.42 | p3nguin | Tomato sauce and green peppers are fruits, cheese is dairy, the crust is bread/grain, sausage/pepperoni is meat... |
| 03:55.45 | p3nguin | What am I missing? |
| 03:55.59 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 03:56.33 | SeRi | to eat! |
| 03:56.36 | SeRi | nom nom nom |
| 03:56.44 | p3nguin | :> |
| 03:56.56 | SeRi | lol |
| 03:57.08 | p3nguin | :] |
| 03:57.34 | SeRi | lol |
| 03:57.38 | SeRi | now Ia m hungry. |
| 03:57.46 | p3nguin | Maybe beer would help. Beer is healthy, right? |
| 03:57.52 | p3nguin | Steak in a can? |
| 03:57.59 | p3nguin | or bottle, as the case may be. |
| 03:58.02 | SeRi | hells yea |
| 03:58.06 | SeRi | lol |
| 03:58.18 | SeRi | gulp gulp whiskey |
| 03:59.24 | p3nguin | I don't drink hard liquor too much. I have the occasional craving for some whiskey and cola or vodka with something. |
| 03:59.58 | p3nguin | Someone I really wouldn't mind having is some egg nog with Southern Comfort. |
| 04:00.35 | SeRi | ouch souther comfort |
| 04:00.39 | SeRi | memory's..... |
| 04:00.51 | p3nguin | It is delicious in egg nog. |
| 04:01.06 | SeRi | never tried.... I am curious |
| 04:01.28 | p3nguin | Southern Comfort brand egg nog gave me the idea to try it. |
| 04:01.55 | SeRi | lol |
| 04:01.56 | SeRi | nic |
| 04:02.00 | SeRi | nice* |
| 04:02.34 | p3nguin | I checked the carton to see if it had alcohol included, and it didn't, but said mix with SoCo. |
| 04:02.44 | p3nguin | So I tried, and it was the best. |
| 04:02.51 | SeRi | nice ! |
| 04:04.10 | *** join/#asterisk turtlefence (~tsmart@110.76.135.10) |
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| 04:13.55 | p3nguin | http://latino.foxnews.com/latino/news/2011/12/05/woman-reportedly-tries-to-cut-off-husbands-penis/ |
| 04:16.25 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
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| 04:21.47 | SeRi | p3nguin: damn! |
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| 05:22.52 | SeRi | p3nguin: you around? |
| 05:23.00 | p3nguin | YAY! |
| 05:23.24 | SeRi | lol |
| 05:23.45 | SeRi | ok one sec. I have a PB. |
| 05:25.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 05:25.58 | SeRi | well never mind. |
| 05:26.01 | SeRi | I fixed it |
| 05:26.02 | SeRi | lol |
| 05:26.38 | SeRi | It was the intercom. |
| 05:26.41 | SeRi | I broke it. |
| 05:26.46 | SeRi | and than fixed it again lol |
| 05:27.54 | SeRi | experimenting. |
| 05:28.02 | SeRi | sorry to bother. |
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| 06:14.31 | SeRi | working on a script to make my AstLinux take over in case of my primary astrisk system fails to respond via AMI. |
| 06:14.52 | SeRi | brb |
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| 07:37.34 | ollii | gmornin |
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| 08:28.46 | hajekd | Is it possible to display call price during call? Anyone was trying that with Asterisk? |
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| 08:53.20 | IsUp | hello |
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| 08:55.24 | IsUp | i have strange error messages in my log: http://pastebin.com/kQH8HiBm |
| 08:55.43 | ollii | hajekd: https://wiki.asterisk.org/wiki/display/AST/Advice+of+Charge |
| 08:55.56 | ollii | never tried it on my own |
| 08:56.34 | ollii | IsUp: you should use latest 1.4er version...your .26 is some kind of old |
| 08:56.59 | IsUp | ollii: my system was working stable since 10 months. nothing changed. |
| 08:57.47 | ollii | Memory Allocation Failure...maybe theres a problem with your memory? |
| 08:58.35 | hajekd | ollii: Yep, thanks - curious if anyone tried that on sip channel.... |
| 09:08.37 | IsUp | hajekd: if you have AOC announce (voice) file, you may use Playback with noanswer option. and put Progress before Playback. |
| 09:08.48 | IsUp | hajekd: thats how i handle on SS7 channels. |
| 09:15.01 | *** join/#asterisk oej (~olle@fw1.gml.g.icnet.infracom.se) |
| 09:17.48 | Chainsaw | IsUp: You're running out of RAM. Add more. |
| 09:18.32 | IsUp | Chainsaw: 4 GB RAM and 160 mb free right now, are you sure its a RAM problem? |
| 09:18.51 | Chainsaw | IsUp: Yes. It is wanting to allocate *memory* and it can't. |
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| 09:19.09 | Chainsaw | IsUp: What scenario other then "all memory already allocated" can you think of that would cause that? |
| 09:19.25 | Chainsaw | IsUp: You have two options. |
| 09:19.36 | Chainsaw | IsUp: You can upgrade your ancient software in the hope that there is a memory leak in older versions that has since been fixed. |
| 09:19.50 | Chainsaw | IsUp: Or you can stubbornly stay on it, leak and all, and add more RAM so that it hurts you less. |
| 09:20.04 | IsUp | ChanServ: i found the scenario. i am using cdr_tds, and my SQL server was down. so probably CDR buffer is caused that deadlock. |
| 09:20.06 | Chainsaw | doesn't care either way, but something will have to change |
| 09:20.38 | Chainsaw | Involving ChanServ in it now. Serious business. |
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| 09:20.42 | IsUp | :P |
| 09:20.55 | IsUp | Chainsaw: thanks for your advice but this is my production server and as i said, it was working stable 10+ months. |
| 09:21.10 | IsUp | Chainsaw: probably cdr_tds caused that. because i see too many "Failed to connect SQL server" errors |
| 09:21.32 | Chainsaw | IsUp: Yes. And the moment your SQL server is down you run out of RAM. |
| 09:21.39 | aberrios | I'm having an issue loading res_cepstral. Keeps complaining it cant find the swift library, but I've pointed ld.conf.so to the correct dir and run ldconfig... anything I've missed? |
| 09:21.48 | Chainsaw | IsUp: If you don't handle errors they will stack up and block your way. Yes. |
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| 09:35.52 | krotos | hi all :) |
| 09:36.26 | aberrios | lo |
| 09:48.55 | IsUp | i have 3.2GB core file in my /tmp, what does it mean? i know my asterisk crashed somehow but why file is 3.2GB? |
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| 10:36.00 | hajekd | Guys, what are you using for Outlook TAPI integration with Asterisk? |
| 10:36.12 | hajekd | Activa does not seem to be realiable...:( |
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| 10:56.04 | ollii | IsUp: core dump (more properly a memory dump or storage dump) |
| 10:56.17 | ollii | hajekd: outcall,phonesuite, estos procall |
| 10:56.24 | ollii | ordered by prize ;) |
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| 11:00.31 | IsUp | gotta go, thanks for help everyone |
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| 11:50.32 | mac|gyver | Hi all, I have a problem when hanging up an incoming call: Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/030XXXXXXX-00000002' in macro 'hangupcall' |
| 11:55.02 | singler | and your problem is? |
| 11:55.23 | fprior | mac|gyver: can you post macro-hangupcall on pastebin ? |
| 11:55.54 | mac|gyver | the problem is that after hanging up the call, I can't call inbound again, I get the voicemail of my provider |
| 11:56.57 | mac|gyver | fprior: http://pastebin.com/e2HXabZ9 |
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| 12:18.07 | NetRipper | hi, i have a question regarding outbound calls (no incoming calls at all) from a service using the AGI application (using asterisk-java library). I want to initiate multiple outgoing calls at the same time, but my voip provider only allows me to have one outstanding call at a time (subscription limitation). Is there a way to queue outgoing calls so that the next call is made after the first one hangs up? |
| 12:30.29 | mac|gyver | so after a fresh start, I can make an incoming call, when I hangup it does the "exited non-zero" message. Then when I call again this is logged: [Dec 6 13:26:58] NOTICE[6204] chan_sip.c: Call from '' to extension '307370602' rejected because extension not found in context 'default'. |
| 12:37.26 | singler | mac|gyver: exited non-zero is not related to your problem |
| 12:37.44 | singler | where does your "normal" calls land? |
| 12:38.08 | singler | I mean, which context |
| 12:38.36 | singler | because that call entered default context, and required extension (307370602) was not found there |
| 12:41.50 | mac|gyver | calls that do work use the same extension, not sure what context they use |
| 12:43.11 | singler | in verbose output you can see |
| 12:43.29 | mac|gyver | goes from from-pstn to from-did-direct |
| 12:45.25 | singler | ok, so that another call does not get matched to your sip config, check if provider uses same IP/credentials for failed call |
| 12:46.31 | mac|gyver | why would call #2 be different? (not saying that I disagree, just trying to understand) |
| 12:46.54 | singler | missconfiguration/load balancing |
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| 12:48.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:55.08 | mac|gyver | singler: how can I find out what IP/credentials the provider uses? |
| 12:55.31 | singler | is the system idle? |
| 12:55.37 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 12:55.45 | mac|gyver | yes |
| 12:56.26 | singler | I think you can use "sip set debug on" to enable sip debug, I personally use tcpdump and then analyze with wireshark :) |
| 12:56.39 | mac|gyver | that's a console command right |
| 12:57.29 | singler | "sip set debug on" is asterisk console command |
| 12:57.39 | mac|gyver | ok |
| 12:57.42 | leifmadsen | likes tshark as well |
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| 13:18.27 | jkroon | irroot, or anybody else able to assist with T.38? |
| 13:18.46 | jkroon | How do I go about debugging this: [Dec 6 15:17:29] ERROR[5052]: res_fax.c:1558 receivefax_t38_init: error reading frame while generating CED tone on SIP/ac1-0000002d / [Dec 6 15:17:29] ERROR[5052]: res_fax.c:1885 receivefax_exec: error initializing channel 'SIP/ac1-0000002d' in T.38 mode |
| 13:19.09 | gordonjcp | afternoon |
| 13:19.25 | gordonjcp | is there a way to specify a dialplan on the Grandstream Budgetone BT100 family? |
| 13:19.45 | gordonjcp | I know they're old, cheap and crap as a Chinese motorbike, but they're what I have lying around the workshop |
| 13:23.07 | [TK]D-Fender | gordonjcp, No. |
| 13:25.19 | gordonjcp | heh |
| 13:25.29 | gordonjcp | man, these things really *suck* ;-) |
| 13:25.44 | gordonjcp | ah well, good enough for rock'n'roll |
| 13:25.48 | jkroon | gordonjcp, yes they do, but you CAN create dialplan code from them on asterisk side. |
| 13:26.34 | mirelab | does anyone know if CDR can be recorded for failed calls and if it's possible to record it before Hangup ? :P |
| 13:26.47 | [TK]D-Fender | gordonjcp, There's a reason they're known as BarbieTones. |
| 13:27.16 | leifmadsen | gordonjcp: their great phones for smashing at the end of a rock concert |
| 13:27.21 | leifmadsen | s/their/they're/g |
| 13:27.23 | [TK]D-Fender | mirelab, Yes, there is a config option for that. Read up on the samples... |
| 13:28.30 | gordonjcp | leifmadsen: :-) |
| 13:28.37 | gordonjcp | jkroon: I'm not that worried about it tbh |
| 13:28.37 | [TK]D-Fender | leifmadsen, http://stopthecap.com/wp-content/uploads/2010/05/itt.jpg <--- when you absolutely, positively need to clock the McFuck outta someone, accept no substitutes. |
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| 13:28.52 | gordonjcp | I've been called upon to pull a SIP-over-wireless-link demo out of my arse at zero notice |
| 13:28.56 | leifmadsen | [TK]D-Fender: pfffft, I have the rotary version of that phone :) |
| 13:29.11 | gordonjcp | it just so happens that py personal laptop runs Ubuntu so installing asterisk is a piece of piss |
| 13:29.21 | [TK]D-Fender | leifmadsen, I've had both, but come on... touch-tone at least... |
| 13:29.28 | gordonjcp | and there are four Budgetone 102s kicking around in the Great Big Heap of Shite in the store |
| 13:30.41 | leifmadsen | [TK]D-Fender: pfffft! you crazy kids and your buttons! you're probably all too fat now to get your fingers in the holes in the dial! In my day, if you couldn't use the dial, we starved you for 3 weeks and you were better off for it! |
| 13:31.22 | jkroon | ok, so no help on the t38? |
| 13:31.39 | jkroon | any ideas where to start looking for what could be wrong in the above? |
| 13:31.46 | singler | leifmadsen: you do not need to insert your finger fully into dial, to dial it :P |
| 13:31.57 | leifmadsen | singler: what do you know?! |
| 13:32.08 | leifmadsen | has no idea and isn't nearly as old as he is trying to sound |
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| 13:32.51 | singler | I know, maybe I am not old, but I am from "less advanced" country, I did use phone with a dial some days ago ;) |
| 13:33.29 | leifmadsen | I haven't used a pulse dial phone in years now.... |
| 13:33.39 | leifmadsen | maybe since I was a kid |
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| 13:34.35 | [TK]D-Fender | leifmadsen, Last time I used pulse dial was when the keypad on a phone I was using was dead so I dialed with the hook-switch :) |
| 13:35.05 | leifmadsen | :) |
| 13:35.13 | ollii | leifmadsen: one of our customers wanted his old "grandma" phone on his new * pbx ... so we took a grandstream |
| 13:35.16 | ollii | and boom bay |
| 13:35.18 | ollii | *baby |
| 13:35.19 | leifmadsen | [TK]D-Fender: my cell phone doesn't have a hookswitch :) |
| 13:35.38 | leifmadsen | ollii: ya, I'm thinking about hooking up an ATA and turning on pulse dialing :) |
| 13:35.48 | leifmadsen | then putting my rotary dial phone somewhere in the house |
| 13:36.21 | ollii | its working ... pickup is a bit tricky...*8 ;) |
| 13:37.56 | leifmadsen | heh |
| 13:38.11 | leifmadsen | ollii: change it to *1? |
| 13:38.22 | leifmadsen | at least only 1 long dial |
| 13:38.24 | singler | star is difficult, not digits |
| 13:38.29 | leifmadsen | oh right |
| 13:38.39 | leifmadsen | technically you don't need to use * or # |
| 13:38.52 | leifmadsen | just use 01 or something? |
| 13:39.27 | ollii | "666" |
| 13:39.30 | ollii | evil magic |
| 13:39.31 | ollii | ;) |
| 13:39.32 | leifmadsen | :D |
| 13:39.43 | leifmadsen | 6 is still a relatively long dial :) |
| 13:40.02 | singler | use 0! :) |
| 13:40.21 | singler | like "000" :) |
| 13:40.35 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 13:40.40 | [TK]D-Fender | leifmadsen, You need ---> http://www.thinkgeek.com/electronics/cell-phone/7830/ |
| 13:41.27 | *** join/#asterisk gpearson (~gspearson@fw1.niesc.k12.in.us) |
| 13:43.08 | gpearson | Anyone familiar with Earthlink Business and configuring their T1 Voice lines in with Asterisk and Digium TE121 Card. Currently T1 is connected to an AdTrans Box to provide dialtone to Analog PBX |
| 13:43.39 | leifmadsen | [TK]D-Fender: too funny |
| 13:44.25 | [TK]D-Fender | gpearson, Couldn't find a specifications sheet for the service you're paying them for, or get them to just tell you? |
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| 13:48.14 | gpearson | [TK]D-Dender: What I have been told is our T1 is has Signaling type of ESFB8ZS and configured for "loop". |
| 13:49.38 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 13:50.53 | [TK]D-Fender | gpearson, "loop star" GAH... analog over digital... Get those chumps at the telco to get you real DID's at PRI signaling |
| 13:51.00 | [TK]D-Fender | start* |
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| 13:51.14 | mirelab | [TK]D-Fender: Thanks :) |
| 13:51.42 | [TK]D-Fender | gpearson, span => 1,1,0,B8ZS, ESF |
| 13:51.59 | [TK]D-Fender | gpearson, fxsls=1-24 |
| 13:52.10 | [TK]D-Fender | gpearson, mod for your ec of choice) |
| 13:53.01 | [TK]D-Fender | gpearson, those are the key systems.conf bits. fxs_ls is what to use in chan_dahdi.conf and looks just like any other analog card aside from that. |
| 13:54.01 | leifmadsen | wow, I don't think I've ever seen loopstart analog over a digital circuit before |
| 13:54.04 | leifmadsen | that's just madness |
| 13:56.04 | jkroon | looks at that and wonders HOW you would even begin to physically build a loopstart using a digital channel |
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| 14:13.01 | jkroon | leifmadsen, gpearson - just bumped into this - the provider could be referring to CAS: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800e2560.shtml |
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| 14:43.12 | [TK]D-Fender | leifmadsen, Improvement! ---> http://www.chipchick.com/2011/12/off-the-hook.html?utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+ChipChick+%28Chip+Chick%29 |
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| 14:45.05 | mac|gyver | singler: about the problem I had (still have actually), when I enable sip debugging I can't call the first time (I get the "goodbye" voice). If I don't enable debugging, I can call once, and it fails the second time (with "goodbye" too) |
| 14:46.12 | ollii | maybe today someone could answer my question... ;) (* 1.4 and 1.8) if im doing "queue show QUEUENAME" on * cli...is that avg holdtime a persistent value stored in astdb or somewhere in memory? |
| 14:46.16 | singler | I think debugging is not at fault here, enable it and try a few times |
| 14:46.28 | mac|gyver | ok |
| 14:46.31 | singler | or use tcpdump/tshark to capture packets |
| 14:46.36 | mac|gyver | yeah I might try that |
| 14:46.49 | leifmadsen | [TK]D-Fender: thanks, christmas idea sent to wife ;) |
| 14:47.06 | jkroon | mac|gyver, that sounds more like a dialplan issue IMHO. unless the goodbye is generated on your phone. |
| 14:47.09 | leifmadsen | ollii: I do not believe so |
| 14:47.30 | mac|gyver | jkroon: it's the goodbye from asterisk, it's also logged |
| 14:47.43 | jkroon | then it's more than likely dialplan, not sip. |
| 14:48.06 | singler | jkroon: some calls from provider hit default dialplan instead of from-pstn |
| 14:48.07 | ollii | so it might be only stored in memory? |
| 14:48.10 | mac|gyver | earlier I got the provider voicemail, I skipped the /DID from the registration string |
| 14:48.28 | jkroon | singler, ah ok, no then your sip peers matching isn't happening properly. |
| 14:48.42 | jkroon | if you don't spec /did it defaults to /s |
| 14:48.44 | singler | it is mac|gyver's |
| 14:50.13 | mac|gyver | hmm |
| 14:50.15 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:50.53 | *** join/#asterisk Ulrar (~Ulrar@2a01:e0b:1:136:62eb:69ff:fe8f:18a0) |
| 14:51.36 | Ulrar | Hi, is there a signal for C agi when the channel is hang up ? In perl I think it's a SIGHUP signal, is there something like that in C ? |
| 14:53.52 | ollii | leifmadsen: what do you mean? stored in memory or in astdb? |
| 14:54.28 | leifmadsen | <ollii> ...is that avg holdtime a persistent value stored in astdb or somewhere in memory? |
| 14:54.31 | leifmadsen | <leifmadsen> ollii: I do not believe so |
| 14:54.46 | leifmadsen | (i.e. not stored in the DB) |
| 14:54.52 | ollii | ah okay, thanks |
| 14:55.50 | WIMPy | Ulrar: DIGHUP is an OS thing, not a language thing, so yes. |
| 14:56.13 | Ulrar | I didn't knew |
| 14:56.15 | Ulrar | I'll try, thanks |
| 14:59.21 | mac|gyver | it's not that the hangup issue causes it to be in use or something? |
| 15:01.27 | Ulrar | Well, in perl it looks like the SIGHUP signal is received when the channel is hanged up |
| 15:01.37 | Ulrar | And that's what I need |
| 15:01.43 | mac|gyver | Ulrar: oh sorry, it wasn't a response to you :) |
| 15:01.50 | Ulrar | Ho ^^' |
| 15:03.50 | WIMPy | Yes, it's the HangUP signal. |
| 15:04.09 | WIMPy | So it makes sense for Asterisk to use that. |
| 15:05.47 | mac|gyver | jkroon: any idea how to debug that? |
| 15:09.28 | [TK]D-Fender | mac|gyver, SIP DEBUG will show you what the call is matching and loking for. |
| 15:10.23 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:11.51 | jkroon | mac|gyver, with great difficulty, sip debug is a good start. |
| 15:11.59 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 15:12.16 | jkroon | it takes some time and avoid looking into your eyelids. simple typos have been known to cause hours of frustration. |
| 15:18.01 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 15:18.14 | voipeng | is there anyway to place a test call to multiple phone numbers from the cli? |
| 15:19.04 | jkroon | channel originate? |
| 15:19.09 | voipeng | hmm? |
| 15:20.31 | jkroon | help channel originate |
| 15:21.36 | voipeng | http://www.voip-info.org/wiki/view/Asterisk+cli+originate |
| 15:21.36 | voipeng | ah |
| 15:23.22 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:24.05 | voipeng | wish i could just enter the number haha |
| 15:25.11 | jkroon | you can - if you script it :p |
| 15:25.11 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 15:25.19 | mac|gyver | jkroon: ah ha! destination IP changes |
| 15:25.24 | mac|gyver | eh |
| 15:25.26 | mac|gyver | source IP |
| 15:26.01 | jkroon | then you need multiple definitions would be in your future, one for each possible source. |
| 15:26.18 | jkroon | or change your default sip context :p |
| 15:26.34 | jkroon | and rely on anonymous calls (heavy insecure - take precautions) |
| 15:30.39 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:33.50 | mirelab | does anyone know how to change status of SIP/<exten>@<IP> member of queue, That is the phone that rings when caller joines queue |
| 15:34.19 | leifmadsen | change status? |
| 15:34.21 | leifmadsen | like, pause? |
| 15:34.22 | mirelab | but member status is always "not in use " |
| 15:34.29 | leifmadsen | what version of asterisk? |
| 15:34.35 | leifmadsen | you probably need callcounter=yes in sip.conf |
| 15:34.35 | mirelab | like from not in use to busy |
| 15:34.39 | mirelab | 1.8.7 |
| 15:34.49 | leifmadsen | ya, callcounter=yes in sip.conf |
| 15:34.51 | leifmadsen | should be all you ened |
| 15:35.22 | leifmadsen | if it's dynamic and the peer isn't configured in sip.conf, then you might need to define the state_interface |
| 15:35.24 | mirelab | thx Laif :) |
| 15:35.26 | *** join/#asterisk d00gster (~dt@77.30.126.165) |
| 15:35.59 | mirelab | Leif* |
| 15:37.08 | voipeng | anyone have an example i could use? |
| 15:37.22 | voipeng | for scripting the calling out function |
| 15:38.10 | mac|gyver | jkroon: think it's working now, thanks :) |
| 15:38.24 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 15:38.30 | dddh | hi |
| 15:38.43 | dddh | I guess I should have joined #skype |
| 15:38.45 | dddh | anyway |
| 15:39.57 | dddh | are there free skype<->sip solutions? |
| 15:41.23 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:42.05 | [TK]D-Fender | dddh, http://www.google.ca/#hl=en&cp=12&gs_id=1w&xhr=t&q=skype+to+sip&pf=p&sclient=psy-ab&biw=1600&bih=927&source=hp&pbx=1&oq=skype+to+sip&aq=0&aqi=g4&aql=&gs_sm=&gs_upl=&bav=on.2,or.r_gc.r_pw.r_cp.,cf.osb&fp=f607f49c1d4beb95 |
| 15:42.23 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:42.42 | jkroon | ok, if asterisk for a peer has T.38 MaxDtgrm: 122 - why would it put this in the SDP: a=T38FaxMaxDatagram:1393 ?? |
| 15:44.07 | leifmadsen | voipeng: it's pretty straight forward and wouldn't be any different than a regular dialout... |
| 15:44.17 | dddh | [TK]D-Fender: If I understand correctly "Skype Connect" is not a gateway |
| 15:44.50 | dddh | wants to call skype from sip |
| 15:45.02 | voipeng | i dont typical do much configuration on the asterisk side, the voiceaxis software typically takes care of it for me |
| 15:45.16 | leifmadsen | voipeng: for example, I just did this yesterday from the CLI: channel originate SIP/MyVoipProvider/<destination_number> extension astley@Rickroll |
| 15:45.50 | leifmadsen | voipeng: and the dialplan just looks like: exten => astley,1,Playback(/home/lmadsen/RickAstley) |
| 15:46.28 | leifmadsen | the 'astley' extension could just be a pattern match that does a Dial() to some peer |
| 15:46.44 | leifmadsen | so you could use your subroutine that converts the extension number to the SIP device you're calling |
| 15:47.26 | leifmadsen | anyways, I have to bike to the bank now, and it snowed out, so time to get bundled up |
| 15:47.32 | voipeng | ok, so first i make the extension |
| 15:47.38 | voipeng | then the dial function |
| 15:47.43 | voipeng | then the channel command? |
| 15:50.27 | dddh | I guess it means there are no working sip->skype solutions? |
| 15:50.46 | ollii | let it snow, let it snom, let it snom |
| 15:50.55 | ollii | pay attention...snow is whore |
| 15:52.20 | gordonjcp | looove snow |
| 15:52.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:57.10 | mirelab | leifmadsen: I've set up sip trunk for that phone with callcounter=yes and now the status is changed from (Not in use) to (Ringing) but not changed after Hangup :( |
| 15:57.49 | mirelab | leifmadsen: i know this is not a dynamic member this way :/ |
| 15:58.00 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:00.21 | SeRi | leifmadsen: That was me that add it you to my google+ last night |
| 16:00.32 | *** join/#asterisk clintc (~clintc@n128-227-125-126.xlate.ufl.edu) |
| 16:16.48 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
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| 16:32.39 | Ulrar | Mh, in my callback function for the SIGHUP signal, I do a "GET VARIABLE HANGUPCAUSE" and asterisk say that : "200 result=-1 endpos=32748" |
| 16:33.02 | Ulrar | Is that normal ? It looks like a play function or somethink like that |
| 16:33.10 | Ulrar | something* |
| 16:35.19 | *** join/#asterisk adeel|work (~adeel@unassigned-220.80.183.216.net.blink.ca) |
| 16:36.23 | adeel|work | anyone know of a tool that can generate a call periodically? i was considering sipp or sipsack |
| 16:37.16 | Qwell | adeel|work: Asterisk? |
| 16:37.57 | adeel|work | well the call must be routed through my * box, preferably from a typical UAC |
| 16:38.08 | adeel|work | i guess i could use another * box to do it |
| 16:38.10 | Qwell | Asterisk is a UAC. |
| 16:38.13 | adeel|work | i'm aware |
| 16:38.22 | Qwell | So then why don't you generate the call in Asterisk? |
| 16:38.42 | Qwell | I don't understand why you would need a separate box to do it. |
| 16:38.44 | adeel|work | because i have an SBC in front of it, and i'd like to test out the entire dialplan |
| 16:38.52 | adeel|work | and my entire network for that matter |
| 16:40.42 | adeel|work | plus, i'd like to raise an alert if the call fails and script out a few different events that should occur, without modifying the production dial plan |
| 16:41.37 | jkroon | UDPTL asked to send 59 bytes of IFP when far end only prepared to accept 54 bytes; data loss will occur.You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. |
| 16:41.42 | jkroon | any idea how to actually do that? |
| 16:43.26 | Qwell | jkroon: The error message tells you how. |
| 16:43.36 | Qwell | see sip.conf.sample |
| 16:49.57 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:50.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:50.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:52.41 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:53.47 | *** join/#asterisk pdtpatrick_ (~pdtpatric@ip72-211-207-15.oc.oc.cox.net) |
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| 16:58.22 | *** join/#asterisk hacim (~micah@debian/developer/micah) |
| 16:58.52 | jkroon | Qwell, actually it's already set to t38_udptl=maxdatagram=122,redundancy |
| 16:59.37 | jkroon | (which happens to be the value it advertizes) |
| 17:00.01 | jkroon | now the question is, once I know that it is only prepared to accept 54 - how do I calculate what I should set maxdatagram to? |
| 17:00.04 | hacim | i want to setup asterisk to call out to a specific number, I've purchased termination from callcentric, but am a little confused what I need to do to set this up |
| 17:02.37 | SeRi | ~book |
| 17:02.38 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:02.43 | SeRi | hacim: ^^ |
| 17:03.26 | SeRi | It has good iformation and examples to get you setup. |
| 17:06.36 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
| 17:11.23 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:11.56 | hacim | [Dec 6 09:10:30] WARNING[24981]: chan_sip.c:23482 set_insecure_flags: Unknown insecure mode 'very' on line 1173 |
| 17:12.05 | hacim | ipkall recommends that |
| 17:13.09 | jkroon | hacim, insecure=invite == better |
| 17:13.30 | pdtpatrick_ | Question .. im trying to see the actual SIP invite that goes out when i make a call, how can i see that please? |
| 17:13.38 | jkroon | sip set debug peer ???? |
| 17:13.46 | hacim | jkroon: i'll try that, dunno what ipkall supports |
| 17:18.01 | SeRi | pdtpatrick_: sup set debug on |
| 17:18.08 | SeRi | s/sup/sip/ |
| 17:18.14 | pdtpatrick_ | much appreciated :) |
| 17:18.32 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:18.39 | SeRi | good luck. |
| 17:20.44 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
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| 17:32.58 | hacim | how can I pass more than one sound to Data: in a .call file? |
| 17:34.05 | jkroon | I'm assuming you are using Application along with that? |
| 17:34.38 | hacim | jkroon: i'm just dropping a .call file in /var/spool/asterisk/outgoing |
| 17:34.54 | [TK]D-Fender | hacim, "core show application Playback" |
| 17:35.15 | [TK]D-Fender | hacim, data is whatever the application takes. read its instructions |
| 17:36.41 | hacim | hm, i thought that I could do: Data: tt-monkeysintro&tt-monkeys |
| 17:37.07 | jkroon | you can |
| 17:37.09 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 17:37.39 | hacim | oh duh, tpo :) |
| 17:37.41 | hacim | typo |
| 17:37.59 | Kobaz | Playback(path/to/sound1&path/to/sound2&...) |
| 17:38.13 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:39.19 | l2trace99 | is there any way to set context by inbound ip for sip ? |
| 17:40.00 | [TK]D-Fender | l2trace99, Yes. Make a peer for it |
| 17:41.46 | l2trace99 | I already have one |
| 17:42.29 | l2trace99 | I am looking to connect 2 servers but need to route differently |
| 17:43.08 | *** join/#asterisk irroot (~gregory@41.49.16.67) |
| 17:43.23 | l2trace99 | based on their inbound ip |
| 17:45.29 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:46.32 | [TK]D-Fender | And that's what peers do |
| 17:47.13 | l2trace99 | then how do I specify the host entry in the sip.conf ? |
| 17:47.20 | [TK]D-Fender | host=IP |
| 17:47.30 | l2trace99 | that is the remote ip |
| 17:48.09 | [TK]D-Fender | Well if you're referring to which IP on your server they came in on look at "core show function CHANNEL" |
| 17:48.21 | l2trace99 | i am looking to specify context based on inbound connection |
| 17:48.43 | l2trace99 | so would have to do in the dialplan |
| 17:48.45 | l2trace99 | ? |
| 17:49.00 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:49.22 | [TK]D-Fender | yes |
| 17:49.31 | l2trace99 | ok |
| 17:49.53 | l2trace99 | I was just looking to see if there was a way to do it from within the sip.conf |
| 17:50.49 | [TK]D-Fender | nope |
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| 18:01.01 | hacim | is there way to provide a message on error when an outgoing call doesn't work? |
| 18:04.38 | paulc | hacim: Sure - use the DIALSTATUS variable (I think) and jump to a Playback |
| 18:06.16 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 18:06.42 | hacim | paulc: hm. have an example? i'm using a .call file |
| 18:07.21 | paulc | hacim: So you call party A via a call file, then want to call party B, but play an error message if the outbound call fails? |
| 18:08.15 | hacim | paulc: i call party A via a call file (basically I am needing to call this number once per month to keep it alive) and if it fails I want to send an email or other notification |
| 18:08.23 | hacim | because if the call doesn't work in one month, the number goes away |
| 18:08.59 | hacim | essentially, I need some kind of notification that the call did not work |
| 18:09.28 | paulc | hacim: Hmm.. You could do it via AMI - originate the call then watch events waiting for the result of that call |
| 18:10.08 | paulc | Or just set yourself a recurring calendar appointment and pick up the phone once a month - might be easier/more efficient in terms of time ;-) |
| 18:10.14 | hacim | i can't log via syslog or send an email on error or something? |
| 18:11.04 | paulc | Sure.. get your call file to dial a Local/ channel, then use DIALSTATUS result to spawn a shell/system command to send you an email (for success or failure, so you know) |
| 18:12.58 | hacim | hm i wonder if I can plug that into nagios |
| 18:13.12 | SeRi | hacim: Yes you can |
| 18:13.23 | SeRi | I monitor with nagios. open channels |
| 18:13.35 | [TK]D-Fender | hacim, dial a local channel and put your logic in there. |
| 18:13.47 | [TK]D-Fender | AMI is serious overkill |
| 18:13.59 | SeRi | along with other things.... |
| 18:14.17 | hacim | ok, i'll look into how to dial a local channel |
| 18:16.31 | SeRi | [TK]D-Fender: My new polycom is on the way :P |
| 18:22.00 | [TK]D-Fender | SeRi, Which one are you getting now? |
| 18:24.13 | SeRi | [TK]D-Fender: os is the same I bid on last week. the 321. |
| 18:24.23 | SeRi | s/os/oh/ |
| 18:24.49 | SeRi | should be here by friday. |
| 18:25.05 | [TK]D-Fender | SeRi, Cool, you'll be able to give a run at the latest firmware then |
| 18:25.17 | SeRi | Yes sr :) |
| 18:25.32 | [TK]D-Fender | SeRi, Prepare for culture shock... |
| 18:25.42 | SeRi | lol |
| 18:25.48 | [TK]D-Fender | 3.3 diverged quite a bit... 4.0 looks even moreso |
| 18:25.54 | SeRi | is ready with matches and speer |
| 18:26.07 | SeRi | wow. |
| 18:26.18 | SeRi | exciting! |
| 18:26.48 | SeRi | [TK]D-Fender: I was reading that I should do combined the first time flashing and than move to split.... does it really matter? |
| 18:27.53 | mac|gyver | one last problem.. I have my Outbound CallerID set on the sip trunk, but external phones receiving calls show "Blocked" |
| 18:28.31 | [TK]D-Fender | SeRi, not sure what you mean there.. |
| 18:28.59 | [TK]D-Fender | mac|gyver, Show us the complete call in a pastebin |
| 18:28.59 | [TK]D-Fender | ~pb |
| 18:29.00 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 18:29.01 | [TK]D-Fender | ^^^ |
| 18:29.28 | mac|gyver | [TK]D-Fender: just the default verbose logging? or sip debug? |
| 18:30.06 | SeRi | [TK]D-Fender: I read that when the phone has 3.x firmware on it and you are fixing to upgrade to 4.0 for the first time to use the combine firmware. once flashed than you can use the split conf files. It was at some blog.... |
| 18:30.52 | [TK]D-Fender | mac|gyver, Both so you can see what's really going on |
| 18:31.24 | [TK]D-Fender | SeRi, I always dump the raw samples into a folder, and wipe it from scratch with stock, then start modding. |
| 18:31.46 | SeRi | [TK]D-Fender: noted. Thanks. |
| 18:32.16 | *** join/#asterisk b0ot (~Jinxed---@147.177.57.101) |
| 18:32.50 | b0ot | I have a device that is not registering that it supports DTMF signalling... could I have this device register with asterisk and then have asterisk tell that it supports DTMF signaling? |
| 18:39.39 | [TK]D-Fender | b0ot, Registration has nothing to do with how its calls will process |
| 18:39.49 | [TK]D-Fender | b0ot, the mde you set in you peer is what * will use |
| 18:40.37 | b0ot | [TK]D-Fender, I'm not sure what you mean, all I can tell is that my device doesn't seem to send the correct rtpmap setting and the other device doesn't think it can support DTMF... |
| 18:42.18 | [TK]D-Fender | b0ot, What I'm saying is the "registration" has nothing to do with what gets negotiated. |
| 18:42.41 | [TK]D-Fender | b0ot, set your mode in the peer. that's all there is to do |
| 18:43.47 | b0ot | Where is the peer? |
| 18:45.44 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
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| 18:46.42 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 18:48.06 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 18:48.28 | voipeng | if g729 show licenses isnt an accepted command... i guess its safe to say its not installeD? |
| 18:48.52 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 18:49.04 | SeRi | voipeng: If I am not mistaken you have to buy the license from digium |
| 18:49.18 | voipeng | yea |
| 18:49.21 | voipeng | you are right |
| 18:49.36 | voipeng | but i want to check out if its already installed, i thought it was g729 show licenses |
| 18:49.38 | mac|gyver | [TK]D-Fender: http://pastebin.com/jHqzEdiC the 030 number should be the outgoing CID, 06 is the destination |
| 18:49.47 | mac|gyver | so this is asterisk -> mobile phone |
| 18:51.40 | SeRi | show g729 |
| 18:51.47 | SeRi | voipeng: ^^ |
| 18:52.03 | voipeng | ? |
| 18:54.17 | [TK]D-Fender | b0ot, sip.conf |
| 18:54.23 | SeRi | voipeng: whet v of * you have? |
| 18:54.31 | dijib | nt working today SeRi? |
| 18:54.35 | SeRi | s/whet/what/ |
| 18:54.36 | *** join/#asterisk Srini (~Srini@219.91.201.74) |
| 18:54.45 | Srini | Hi room |
| 18:54.48 | SeRi | dijib: I been sick with shingles |
| 18:54.58 | voipeng | 1.4.29 |
| 18:54.59 | [TK]D-Fender | voipeng, If you don't have the commands then the module doesn't exist or isn't loaded |
| 18:55.02 | dijib | wernt you telling me about htat the other day>?/ |
| 18:55.03 | voipeng | k |
| 18:55.12 | dijib | saying that i needn't be stressed about anything |
| 18:55.42 | Srini | I am a newbie trying to configure the digium te220 for the first time for inbound calls - can someone help me with a pointer to an easy step by step document? |
| 18:56.02 | Srini | Google did not help much :( |
| 18:56.09 | SeRi | dijib: Yes... and I got it again. |
| 18:56.25 | SeRi | I went to work yesterday but I could not bare the pain |
| 18:56.47 | dijib | damn ive never had it, must be a warm weather thing |
| 18:56.56 | SeRi | warm? |
| 18:57.03 | SeRi | is 38F right now |
| 18:57.08 | SeRi | fucking cold as shit |
| 18:57.18 | dijib | sissy |
| 18:57.19 | dijib | lol |
| 18:57.39 | irroot | its 20:57 here 21c frogs croaking in the yard .... |
| 18:57.58 | SeRi | must be nice irroot!!! |
| 18:58.13 | dijib | australia? cookoo burrows? |
| 18:58.17 | irroot | doors open windows open |
| 18:58.29 | SeRi | irroot: enjoy it :) |
| 18:58.32 | irroot | is in johannesburg south africa |
| 18:58.33 | SeRi | beer? |
| 18:58.45 | SeRi | irroot: ^^ |
| 18:58.47 | irroot | maybe a whisky |
| 18:58.50 | dijib | im drinking beer this weekend |
| 18:58.54 | SeRi | ahhhh indeed |
| 18:59.08 | SeRi | You speko my language now :) |
| 18:59.17 | SeRi | and whiskey are best friends |
| 19:00.05 | irroot | drinks like a dog ages never gets drunk as in dog beers i only have 2 |
| 19:01.10 | dijib | <PROTECTED> |
| 19:01.11 | dijib | <PROTECTED> |
| 19:01.13 | dijib | <PROTECTED> |
| 19:01.16 | dijib | <PROTECTED> |
| 19:02.06 | SeRi | irroot: lol |
| 19:04.48 | Srini | Hmmm no luck for me? |
| 19:05.31 | irroot | Srini you go to the digium site and get the pdf's there ?? |
| 19:05.34 | irroot | ~thebook |
| 19:05.34 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:05.45 | irroot | srini that will help too |
| 19:06.25 | irroot | im in M$ AD hell |
| 19:07.02 | tuxxie | I am trying to key to create a key based on the callerid with a value of 1. Set(DB(gn1num/${CALLERID(num)})=1) |
| 19:07.39 | dijib | Srini, do you have to card installed properly? see it in lsmod? |
| 19:08.05 | tuxxie | but this is not working. i get a error of ignoring entry 'DB(gn1num/9106206507=1' with no '=' |
| 19:08.18 | Srini | dijib: yes I can see |
| 19:08.36 | Srini | It is identified as wct4xxp |
| 19:09.01 | tuxxie | what am i missing? |
| 19:10.09 | dijib | Set(DB(gn1num/${CALLERID(num)})=1); |
| 19:10.16 | dijib | tuxxie: |
| 19:10.32 | tuxxie | Thanks! |
| 19:10.51 | Srini | dijib: Is it okay if I could explian my problem? |
| 19:11.27 | dijib | youve have installed asterisk? configured iax.conf/sip.conf @ dialplan.conf ? |
| 19:11.35 | dijib | explain Srini |
| 19:12.01 | dijib | tuxxie: i dont see whats wrong with you line. |
| 19:13.11 | dijib | alternatively you could do System(/usr/sbin/asterisk -rx "database put gn1num ${CALLERID(num)} 1" |
| 19:13.14 | dijib | i believe |
| 19:13.34 | SeRi | ~ask |
| 19:13.34 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:13.44 | SeRi | Srini: ^^ |
| 19:13.50 | tuxxie | dijib: i'll try than |
| 19:13.55 | Srini | dijib: I am on a PRI (E1) line connected to Span2 of the card. I have installed all the driver modules as explained in the Digium User manual. The module is visible in lsmod as wct4xxp. dahdi_tool still show RED BLU/RED alarm. I am completely new to Inbound setup using asterisk and digium card, I have had successfull trials using asterisk for Outbound (SIP) calling. Now I am not sure about |
| 19:13.55 | Srini | he steps to follow in order to have inbound calls on my asterisk server... |
| 19:14.46 | SeRi | dijib: I fail to see how your context was different from his. |
| 19:14.48 | Srini | dijib: I am not sure is my configuration is erroneous or the connection... trying with limited dahdi_ commands... lights on the card are still read and blinking |
| 19:14.53 | dijib | lets see your sip.conf |
| 19:15.06 | dijib | ~pb |
| 19:15.07 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:15.38 | dijib | make sure you change your passwords |
| 19:15.40 | dijib | btw |
| 19:15.42 | Srini | dijib: I am asking for help in the room pointer to the 'steps' to follow |
| 19:15.44 | irroot | Srini if you see a Blue alarm may need to reset NT please pb the config have you set the jumper to E1 on the card do you need CRC4 ?? |
| 19:16.13 | dijib | i have not ever touched digium hardware |
| 19:16.45 | SeRi | And thst why I dont offer to help on things I have no clue about :P |
| 19:17.21 | irroot | has no clue about anything <- my std disclaimer im sticking to it |
| 19:17.29 | SeRi | sits back and watch those who have experience... I just learn :) |
| 19:17.40 | SeRi | irroot: lol |
| 19:17.44 | dijib | totally SeRi |
| 19:18.01 | dijib | i like it irroot |
| 19:18.23 | irroot | i use E1 quite a bit so may know something about it :P |
| 19:18.40 | tuxxie | dijib: is there a way i can list all keys in gun1num? |
| 19:19.06 | SeRi | I am sure that has nothing to do with sip.... |
| 19:19.23 | SeRi | tuxxie: database show gun1num |
| 19:19.55 | irroot | i think this may be a bit exesive 75%+ of the servers capacity is taken up with torrents |
| 19:20.07 | tuxxie | SeRi: Thanks |
| 19:20.16 | SeRi | irroot: way to much :/ |
| 19:20.22 | SeRi | tuxxie: your welcome |
| 19:20.22 | citywok | irroot: good lord, that's a lot of keyboard drivers |
| 19:20.44 | Srini | irroot: The jumpers were open (default set to T1), I have closed them to set it to E1 |
| 19:21.05 | Srini | irroot: which conf to pb? system.conf? |
| 19:21.23 | mac|gyver | If anyone could have a look at this log: http://pastebin.com/jHqzEdiC Outgoing calls don't show the caller ID. The 030 number should be the outgoing CID, 06 is the destination |
| 19:21.42 | irroot | Srini you will need to sit in the naughty corner :P yeah system.conf |
| 19:22.19 | citywok | mac|gyver: i don't know if (number) works... i always use (num) |
| 19:22.40 | citywok | oh, that's freepbx so it must be valid :p |
| 19:22.46 | Srini | irroot: :) |
| 19:22.57 | mac|gyver | citywok: you'd hope so :P |
| 19:23.21 | [TK]D-Fender | mac|gyver, back. in your PB we can see that your CID is in the invite |
| 19:23.26 | irroot | citywok have lots of test files for drivers too some for audio drivers and some for video even some for HD video |
| 19:23.38 | mac|gyver | [TK]D-Fender: so... something wrong at the provider most likely? |
| 19:23.51 | [TK]D-Fender | mac|gyver, To: <sip:0612345678@sip.xs4all.nl> Contact: <sip:0301234567@172.20.6.6> |
| 19:24.04 | [TK]D-Fender | mac|gyver, Indeed they may block CID overall |
| 19:24.09 | [TK]D-Fender | mac|gyver, You should ask them |
| 19:24.12 | citywok | mac|gyver: on line 182 of the PB i see set(callerid(all) Sander <0301> |
| 19:24.22 | mac|gyver | there's a setting for that in their interface, I disabled that I think |
| 19:24.34 | mac|gyver | citywok: hmmm |
| 19:24.40 | mac|gyver | 0301 |
| 19:24.49 | citywok | looks like it's doing the right thing (i got tired of typing, the whole thing was there) |
| 19:24.50 | mac|gyver | oh right |
| 19:25.01 | mac|gyver | good |
| 19:25.05 | [TK]D-Fender | mac|gyver, [Dec 6 19:43:34] VERBOSE[17783] pbx.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/124-00000024", "1?Set(CALLERID(all)=Sander <0301234567>)") in new stack |
| 19:25.20 | mac|gyver | yeah |
| 19:25.22 | [TK]D-Fender | mac|gyver, this is the line that takes effect and you can see that info in the "from:" |
| 19:25.36 | mac|gyver | check |
| 19:26.23 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:26.58 | mac|gyver | ok I got it working.. |
| 19:27.05 | Srini | irroot: http://pastebin.com/YwyAvkN5 |
| 19:27.06 | mac|gyver | blames the crappy crappy interface at the provider |
| 19:27.23 | mac|gyver | they've hidden the save/reload button really really well |
| 19:27.56 | [TK]D-Fender | mac|gyver, If it took you this little time to find that exact option you suspected anyway ... I'd blame myself :p |
| 19:28.10 | irroot | Srini seems good |
| 19:28.29 | irroot | dahdi_cfg completes ?? lsdahdi ?? |
| 19:29.02 | irroot | if the red alarm stays remove crc4 flag |
| 19:29.04 | mac|gyver | [TK]D-Fender: uh.. yeah I did feel quite stupid :) |
| 19:29.13 | irroot | if you not sure it must be used or not |
| 19:30.37 | Srini | irroot: Yes it completes |
| 19:30.47 | irroot | Srini you can combine the dchan bchan and echocan lines to have 1 of each |
| 19:31.24 | [TK]D-Fender | mac|gyver, Minor brain-fart. I'm sure you'll grow past it :) |
| 19:32.04 | SeRi | [TK]D-Fender: is it possible to have two * side by side with the same config but with the itsp reg comented out in one of them? |
| 19:32.10 | Srini | irroot: :( if you don't mind... little more detail please |
| 19:32.27 | Srini | irroot: Newbie |
| 19:32.34 | citywok | SeRi: sure, just edit sip.conf in the other one |
| 19:33.00 | irroot | bchan = 1-15,17-46,48-62 |
| 19:33.12 | citywok | we use a floating IP for the primary * box, and the secondary box registers to the ITSP as well, but the ITSP sends calls directly to the floating IP so it works pretty well |
| 19:33.22 | SeRi | citywok: Thanks. I figure so. I just wanted to make sure. I puting up a fail over system |
| 19:33.43 | citywok | :) |
| 19:33.51 | mac|gyver | [TK]D-Fender: I just don't hope it's my last brain-fart :-) |
| 19:34.20 | irroot | Srini if you have the channels loaded you can go on to configure dahdi.conf in /etc/asterisk |
| 19:36.20 | Srini | irroot: Well in my case I have the dahdi.conf is in /etc/modprobe.d/dahdi.conf and it looks like an empty file... |
| 19:37.06 | Srini | irroot: dahdi_tool shows 31 channels total, 31 configured and 0 Active |
| 19:37.16 | Srini | irroot: On span 2 I mean |
| 19:37.38 | irroot | Srini mmm that is a modprobe conf file dont think you must mess with that |
| 19:38.08 | irroot | Srini ok so both spans are there then and all channels are in span 2 this is good |
| 19:38.23 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 19:38.41 | Srini | irroot: So 'somehow' the /etc/asterisk/dahdi.conf missing in my case? Mysterious! |
| 19:38.43 | irroot | Srini /etc/asterisk/chan_dahdi.conf |
| 19:38.51 | Srini | irroot: Got it |
| 19:39.10 | dijib | just readinw hat i missed |
| 19:39.17 | dijib | lookin good srini |
| 19:39.41 | Srini | dijib: :) |
| 19:40.06 | SeRi | ok I got my failover server online |
| 19:40.10 | Srini | irroot: All lines are commented... is that normal? |
| 19:41.29 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 19:43.32 | irroot | Srini yeah need to config it for your purpose |
| 19:43.58 | irroot | i have a script that builds them for me so not much help did the script long time back |
| 19:44.25 | Srini | irroot: Is it shareable? |
| 19:44.51 | Srini | irroot: Sorry if the request is not appropriate |
| 19:45.30 | irroot | Srini its out there but will hurt your head more than help it uses the perl bits to go through the dahdi cards and config them |
| 19:47.03 | irroot | Srini msg me a email addie ill send you a 2 port config from a customer |
| 19:47.10 | Srini | irroot: Ok, let me put it this way, I have to configure an inbound trunk so that I can recieve call from PRI then push them on to the extensions as necessary |
| 19:47.32 | Srini | irroot: Sending my email id in pvt. hope people will not find it offending.... |
| 19:47.34 | dym | How can i verbosely log everything that occours on my asterisk CLI to a logfile? I want call executions and stuff like that to be logged for later analysis. |
| 19:47.40 | SeRi | ok all of my phones have been switched to register to hostname instead of IP. |
| 19:49.26 | *** join/#asterisk dmz (~dmz@67.216.138.246.pool.hargray.net) |
| 19:50.24 | *** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net) |
| 19:50.41 | *** part/#asterisk serafie (~erin@nat/digium/x-xqprwmuvnbcethue) |
| 19:51.29 | *** join/#asterisk serafie (~erin@nat/digium/x-xqprwmuvnbcethue) |
| 19:53.47 | irroot | Srini dispatched remove the one group there 2 groups one per PRI or leave it in as backup perhaps |
| 20:02.08 | dijib | k game time for me |
| 20:15.42 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:18.18 | SeRi | p3nguin: you around? |
| 20:31.44 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:33.35 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 20:33.40 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:33.41 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:34.12 | SeRi | hates flexlm |
| 20:34.36 | libryder | is there some sort of data available for geographic routing based on npa/nxx? we have data from these guys http://www.npanxxsource.com/ but cellphones dont' always have a number that is issued to a wire center in their area |
| 20:37.35 | _Corey_ | libryder: How does their data differ from what you can download from the NANPA site directly for free? |
| 20:39.42 | libryder | _Corey_: where is that data available on nanpa? |
| 20:40.31 | _Corey_ | It's been a while since I've downloaded it myself, but I think it may be http://www.nanpa.com/reports/reports_npa.html |
| 20:41.12 | _Corey_ | Actually, it may be this one: http://www.nanpa.com/reports/reports_cocodes_assign.html |
| 20:52.27 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 20:53.08 | sawgood | exten => h,1,NoOp(Hangupcause${HANGUPCAUSE}) |
| 20:53.22 | sawgood | with the above statement, is it possible to issue two commands on one line? |
| 20:53.29 | SeRi | libryder: this people charge you for information that is free? Is there a difference from the data they are providing you? |
| 20:53.33 | *** join/#asterisk cmendes0101 (~nn@pool-173-58-50-52.lsanca.fios.verizon.net) |
| 20:53.52 | sawgood | I would like to see the current context name printed out in the NoOp as well as the hangupcause |
| 20:56.19 | *** part/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:56.26 | *** part/#asterisk Srini (~Srini@219.91.201.74) |
| 20:56.53 | sawgood | -- Executing [h@ITSP1-incoming:1] NoOp("SIP/ITSP1-00015e5b", "Hangupcause16") in new stack |
| 20:57.12 | sawgood | this is what the command oututs ... and I would like more information in the output |
| 20:59.29 | *** join/#asterisk henrikjott (~hj@c83-249-248-93.bredband.comhem.se) |
| 21:00.23 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 21:01.02 | henrikjott | Hi all! I´m having a problem with asterisk picking up .call-files too late. As i understand * should process them right away but in some cases we´re experiencing waits for up to a minute. Does anybody know anything about this? |
| 21:02.08 | henrikjott | btw im running asterisk 1.4.33.1 |
| 21:02.09 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 21:02.15 | Qwell | Why? |
| 21:03.00 | Qwell | 22-Jun-2010 17:55 |
| 21:03.07 | Qwell | 18 months old... |
| 21:03.33 | luckman212 | anyone know if it's possible to interrupt SayDigits() ? i have some dialplan that reads out long strings of numbers, and sometimes I really just want to skip to the next step. I tried pressing # to skip but, doesn't seem to do anything |
| 21:04.45 | henrikjott | @Qwell: I know, but this is for a customer of mine and in a while we will upgrade it.. But this problem has appeared recently! Worked fine before.. |
| 21:05.46 | Qwell | henrikjott: Attempting to solve an issue in a version that old is a waste of your time and ours. |
| 21:08.12 | cmendes0101 | luckman212: I think AGI has escape digits for "say digits" but not sure about in the dialplan |
| 21:08.54 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-smijupntwtkoslsf) |
| 21:09.47 | luckman212 | cmendes0101: going to try to convert SayDigits() to Background() |
| 21:09.50 | henrikjott | @Qwell: I guess you are right.. Im using asterisk to auto-dial out for a telemarketing service. As i said im using .call-files. but which way would be the best for auto-dialing? agi? |
| 21:10.07 | Qwell | call files are fine |
| 21:11.04 | [TK]D-Fender | henrikjott, Check your clock. call files placed in the future wait till their date comes due |
| 21:11.32 | libryder | SeRi: the data they provided us has wire center data with lat, lng, zipcode, city, switch name |
| 21:11.46 | libryder | i couldn't find that anywhere else |
| 21:11.56 | *** join/#asterisk [T]ank (~Tank@206.71.78.180) |
| 21:12.45 | [T]ank | i have a polycom soundpoint IP 4000 conference room phone. When i dial another extension and let it ring for a few seconds, then hang up... the phone that i dialed continues to ring. I cannot for the life of me figure out how to correct that. Any ideas? |
| 21:13.01 | cmendes0101 | luckman212: if you use background wont you have to have something waitexten or a wait to make it not continue on? |
| 21:13.21 | [T]ank | its only that phone that does it. i have lynksys phones everywhere else and they behave like i would expect |
| 21:13.23 | luckman212 | the next step is Read() so I think that might work? |
| 21:13.49 | cmendes0101 | oh ok that would probably be fine, just watch our for the timeout i guess |
| 21:13.58 | [TK]D-Fender | [T]ank, Go look at the call. |
| 21:14.22 | [T]ank | when i debug, i dont see anything... its like the phone is not sending the hangup signal |
| 21:15.23 | [T]ank | i wondered if it was an internal dial pattern, but everything i try makes no difference. |
| 21:15.23 | [TK]D-Fender | [T]ank, Well it either is or isn't... if you don't see it in *... well.. it's just not there. What is the phone running? |
| 21:16.04 | [T]ank | 3.1.0.0147 |
| 21:18.23 | [TK]D-Fender | [T]ank, See if you can upgrade it. If the phone shows every sign of hanging up but no packet to back it up then I'd first suspect it's a bug |
| 21:18.52 | cmendes0101 | For NorthAmerica, is there a NXXNXXX number in any area code that will do a directory listing or something? Had some directory listing charges and I think that was mentioned to me but I can't find what the number would be. |
| 21:21.19 | SeRi | libryder: Interesting. |
| 21:21.23 | [T]ank | K, thanks for the input. I will give an upgrade a try. |
| 21:21.29 | henrikjott | [TK]D-Fender: I have thought about that, but as im generating the files on the same system as * is running i thought the time would be in sync.. am i right? |
| 21:22.42 | [TK]D-Fender | henrikjott, I haven't heard that you've looked for these files in the folder you're supposed to put them in, don't see how you're doing it, no CLI to match. Right now I have no idea what's really going on. |
| 21:24.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:26.15 | [TK]D-Fender | Checkout time, bbiab |
| 21:28.04 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 21:29.21 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 21:43.00 | *** join/#asterisk eja (~user@173.219.119.226) |
| 21:43.19 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-iuicmlgagqovmqxd) |
| 21:44.10 | eja | does a sip or iax2 peer show up as OK if qualify is turned off? or will it always say unmonitored? |
| 21:44.15 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:6854:67ca:f327:363a) |
| 21:45.40 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-103-236.w86-208.abo.wanadoo.fr) |
| 21:48.52 | navaismo | eja: unmonitored |
| 21:49.53 | *** join/#asterisk kannan (kann@14.96.201.145) |
| 21:50.44 | kannan | hello, I am unable to use the # key as a termination / timeout for an IVR. i use a Backgound(file) followed by a WaitExten(5) |
| 21:51.15 | kannan | when i read back the respinse DTMF it includes the # also, instead of it returning only the response DTMF digits |
| 21:51.37 | eja | so what's the downside of turning qualify on all voip trunks? they won't register if the latency goes over 2s? |
| 21:51.44 | WIMPy | Use Read or use non-overlapping extensions. |
| 21:52.06 | WIMPy | eja: More traffic. |
| 21:52.14 | kannan | WIMPy , thanks |
| 21:52.20 | WIMPy | The maximum latency can be configured. |
| 21:52.44 | kannan | WIMPy , so the BackGround / Wait Exten will not use # as a termination key? |
| 21:52.56 | eja | how much more traffic? default interval is 60s for OK and 10s for unreachable? |
| 21:52.57 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:53.07 | SeRi | eja: you can control how qualify is used. |
| 21:53.18 | WIMPy | kannan: No. They wait for an extension and they can include #. |
| 21:53.32 | WIMPy | eja: yes |
| 21:53.54 | kannan | WIMPy , thanks again |
| 21:54.03 | SeRi | eja: http://pastebin.com/nJqsZrvK |
| 21:54.22 | SeRi | I tone down the qualify |
| 21:54.42 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 21:55.40 | eja | SeRi: have you checked how much bandwidth those settings end up using? |
| 21:55.41 | cj | is there any way to authN to google talk without having the cleartext password in gtalk.conf ? |
| 21:56.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:58.06 | SeRi | eja: with my settings not much at all. The frequency is 480 and and there is a gap in betwen each qualify |
| 21:58.53 | SeRi | eja: what are you using qualify for? |
| 22:02.55 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 22:03.39 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 22:04.09 | eja | i'd like the peers to show up as OK with the latency instead of unmonitored |
| 22:04.35 | eja | or unreachable if they are |
| 22:05.49 | SeRi | eja I see. well my settings so far worked ok for me. Although qualify purpose is dieffernt it can also be used that way. |
| 22:06.10 | SeRi | s/dieffernt/different/ |
| 22:08.55 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 22:09.04 | eja | but there's no real disadvantage to turning qualify on other than a small bandwidth hit? |
| 22:19.47 | WIMPy | And a little bit of CPU and RAM, off course. |
| 22:24.48 | SeRi | WIMPy: I have not seen a hit on ram or cpu on my embbeded system using qualify. |
| 22:25.35 | WIMPy | Add more peers :-) |
| 22:26.18 | SeRi | :P |
| 22:32.35 | eja | not too many peers per server i'd like to qualify. maybe 10 on each. |
| 22:40.01 | *** join/#asterisk netman (netman@100.169.76.188.dynamic.jazztel.es) |
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| 22:40.44 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mlhscgzwiugeeljy) |
| 22:40.44 | SeRi | I have about 15. |
| 22:41.14 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-jgbnymfskclzegoe) |
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| 22:56.56 | *** mode/#asterisk [+o angler] by ChanServ |
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| 23:18.49 | *** join/#asterisk tmrhmdv (~tmrhmdv@ool-4575afcd.dyn.optonline.net) |
| 23:19.30 | tmrhmdv | Hi, folks! |
| 23:19.40 | tmrhmdv | How can I disable anonymous SIP calls? |
| 23:20.04 | tmrhmdv | Actually, enable it |
| 23:20.23 | tmrhmdv | but Hangup to unidentified callers |
| 23:21.45 | tmrhmdv | is it this code? http://pastebin.com/4SJPjYtk |
| 23:22.42 | WIMPy | Send them to a context that does whatever you want. |
| 23:23.57 | tmrhmdv | Yep, thanks. Found it. |
| 23:24.02 | tmrhmdv | ALLOW_SIP_ANON = yes |
| 23:24.16 | WIMPy | What is that? |
| 23:24.24 | tmrhmdv | and then send them to where ever I wish |
| 23:24.48 | tmrhmdv | Umm, I found that in FreePBX generated extensions.conf. Is it true? |
| 23:24.51 | p3nguin | That's not a valid asterisk setting. |
| 23:24.55 | p3nguin | ~freepbx |
| 23:24.55 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 23:24.57 | tmrhmdv | Oh |
| 23:25.16 | p3nguin | allowguest=yes or no |
| 23:25.29 | tmrhmdv | My question isn't about FreePBX, I just used that thinking it was my solution |
| 23:25.42 | tmrhmdv | Oh, thanks p3nguin |
| 23:26.00 | p3nguin | The context that you configure in the general section determines where anonymous calls go. |
| 23:26.42 | tmrhmdv | Thank you, works! :) |
| 23:29.14 | SeRi | waz up p3nguin |
| 23:30.21 | SeRi | p3nguin: how you feeling today? |
| 23:49.02 | SeRi | p3nguin: I been having issues with having a failover system come up besides my main asterisk system |
| 23:49.20 | SeRi | when the system comes up after a few min callcentric drops off line |
| 23:49.31 | SeRi | the regsitration on the fail over system is disable |
| 23:51.19 | p3nguin | How are you dealing with failover? Does callcentric have a failover configuration like voipms, or do you have to deal with it completely on your end? |
| 23:51.54 | SeRi | p3nguin: This is a system fail over. meaning if my alix fails than I have my arch take over. |
| 23:52.09 | p3nguin | I really don't want to repeat the last question. |
| 23:52.45 | SeRi | p3nguin: callcentric does have fail over like voip.ms where if the peer is off line it send the cllas to where evr you would like |
| 23:52.56 | SeRi | s/cllas/calls/ |
| 23:52.57 | p3nguin | Is that the part you're having issues with? |
| 23:53.09 | SeRi | No. |
| 23:53.52 | SeRi | The part I am having issues is when I start asterisk in my second system after a few minutes callcentric drops off line on my main syste, |
| 23:54.30 | p3nguin | Are you registering both systems to the same account? |
| 23:54.43 | n3hxs | So now you get VOCP voice over cracked phone |
| 23:54.43 | SeRi | No sr. I have registration disable in one system. |
| 23:55.04 | n3hxs | oops wrong window. |
| 23:55.12 | p3nguin | That doesn't even make sense. If there is no registration, cc won't know the second system exists. |
| 23:55.35 | SeRi | I know. it puzzles me. I do have a peer set on the second system for callcentric |
| 23:55.47 | p3nguin | That can't affect the other system. |
| 23:56.08 | p3nguin | Only a registration will tell cc that the second system exists. |
| 23:56.15 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 23:56.21 | SeRi | Indeed |
| 23:56.25 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 23:56.41 | SeRi | ah! |
| 23:56.53 | SeRi | well looks like an issue on there end! |
| 23:57.10 | SeRi | It just drop off line and the second system is off line |
| 23:58.24 | SeRi | so looks like an issue on their end. |
