| 00:02.23 | paulc | anyone got a copy of Allison Smith's funny "you're not the next caller in line" audio knocking around? not-next.gsm I think it was called? |
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| 00:37.50 | hipitihop | I have been happily running astersk 1.6.2 ubuntu package for many months and although I'm happy to edit dialplans and the like, it is a bit much to ask the same of my martner. Can I just add FreePBX on top ? other recommendations ? |
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| 02:23.40 | jmmills | ^ re: my comment before res_config_curl and or #exec includes should do the trick |
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| 03:21.25 | kcamk | can someone help me with a problem where my outgoing calls disconnect after 30 secs to some numbers |
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| 03:53.28 | golikwid|mac | what error is it giving when it disconnects |
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| 04:05.12 | drfreeze | kcamk: what type of trunk? PRI, analog, sip? |
| 04:08.29 | AliRezaTaleghani | i am going to learn AGI, |
| 04:08.38 | AliRezaTaleghani | which scripting language will be better, to work with mysql? |
| 04:09.06 | AliRezaTaleghani | should i go toward the perl? vs python? or state on bash ?:-/ |
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| 05:29.11 | ChannelZ | AliRezaTaleghani: Any one you want that can read stdin and write stdout |
| 05:32.56 | Alton35 | ok, goofy question time, I am trying to telnet to asterisk on port 5038 and can't get into the account I defined in manager.conf, |
| 05:32.57 | Nugget | telnet is eeeeeeevil! |
| 05:33.13 | Alton35 | I keep restarting asterisk, assuming it's picking up the changes in manager.conf |
| 05:33.28 | Alton35 | Still can't get logged in any which way. Where to look? |
| 05:34.05 | Alton35 | The log files and console are unhelpful. |
| 05:34.33 | ChannelZ | are you telnetting from the same box? Is that port blocked? Is it listening on the interface you think it is? |
| 05:34.54 | Alton35 | yeah, it answers, just keeps saying: |
| 05:35.09 | Alton35 | Response: Error |
| 05:35.09 | Alton35 | Message: Authentication failed |
| 05:35.40 | ChannelZ | How are you trying to login? |
| 05:35.50 | Alton35 | telnet 127.0.0.1 5038 |
| 05:36.02 | Alton35 | lemme paste 3 or 4 lines, um |
| 05:36.23 | Alton35 | [callouts] |
| 05:36.23 | Alton35 | secret = secretx11 |
| 05:36.23 | Alton35 | allow=127.0.0.1/255.255.255.255 |
| 05:36.23 | Alton35 | write=originate |
| 05:36.51 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 05:36.52 | Alton35 | I tried taking out the spaces in the secret= line, but this is a faithful copy of the [mark] section above. |
| 05:36.55 | ChannelZ | so you're doing Action: login |
| 05:37.01 | ChannelZ | Username: callouts |
| 05:37.08 | Alton35 | right |
| 05:37.08 | ChannelZ | Secret: secretx11 |
| 05:37.15 | Alton35 | right, then 2 returns |
| 05:37.49 | Alton35 | I assume there's not where this manager.conf file is commented out or anything, just can't see the problem. |
| 05:39.18 | ChannelZ | 'manager show users' shows your 'callout' user? |
| 05:39.45 | Alton35 | callouts, yes |
| 05:39.55 | Alton35 | never used that before, but yup, it's there |
| 05:40.26 | Alton35 | verbose 10 and debug 10 don't show anything more than the failure to authenticate thing |
| 05:40.29 | ChannelZ | well then all I can guess is either you have a hidden character or something somewhere or you're not typing the auth right |
| 05:41.09 | ChannelZ | Perhaps your allow= line is what is holding it up, because you're not really coming *from* localhost |
| 05:41.27 | Alton35 | let me see |
| 05:41.38 | Alton35 | I guess I could always use the external ip, it's firewalled |
| 05:42.24 | Alton35 | well, you'll be amused to hear, |
| 05:43.18 | Alton35 | after I moved the ;deny= and ;read= commented-out lines out of the middle of the 4 lines I left, now it seems to work |
| 05:43.18 | Alton35 | yikes |
| 05:43.18 | Alton35 | oh well, you have helped |
| 05:43.18 | Alton35 | let me try to place a call, I'm sure more questions will ensue |
| 05:45.25 | Alton35 | I did spend more than 1/2 hour trying this every which way, so you really did help. |
| 05:47.50 | ChannelZ | happy you got it working |
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| 06:00.35 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 06:11.34 | EmleyMoor | I have now got my N97 connected directly to Asterisk, but have a couple of odd things going on: |
| 06:12.28 | EmleyMoor | (a) the phone is taking a long time to start ringing when a call is sent to it - 15 or so seconds after the other phones start ringing |
| 06:13.03 | EmleyMoor | (b) the phone is sending back many 415 (Unsupported Media Type) messages |
| 06:13.17 | EmleyMoor | Is there anything I can do about either of those? |
| 06:14.08 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 06:15.59 | ChannelZ | so you're dialing it and some other phones with & ? |
| 06:16.13 | EmleyMoor | Yes |
| 06:16.48 | ChannelZ | hmm well the delay sounds like the phone just isn't paying attention |
| 06:17.02 | EmleyMoor | Even when dialed on its own there is about 15 seconds between the "Called" in asterisk and the phone ringing |
| 06:17.27 | ChannelZ | Have you turned on sip debug? Is it retransmitting the invite several times? |
| 06:17.54 | EmleyMoor | I'll see... hold on |
| 06:20.20 | EmleyMoor | Looks as though it's not |
| 06:20.40 | ChannelZ | So it's sending an invite and then getting a response? |
| 06:21.08 | EmleyMoor | Yes |
| 06:21.18 | ChannelZ | And what's the response? |
| 06:22.14 | EmleyMoor | 180 Ringing |
| 06:22.28 | ChannelZ | but it doesn't |
| 06:22.52 | ChannelZ | Well that would be something with the device then, not asterisk. |
| 06:23.25 | EmleyMoor | So, the best thing to do would be to take account of it |
| 06:24.40 | EmleyMoor | Right - no great problem. What about the 415s? |
| 06:25.00 | ChannelZ | ? |
| 06:26.40 | EmleyMoor | The phone seems to be responding 415 Unsupported Media Type in response to asterisk's NOTIFY |
| 06:28.54 | ChannelZ | I meant I don't know. seems like an odd response to notify (which is usually just Asterisk giving message waiting indication) |
| 06:29.20 | EmleyMoor | Hmmm... OK, not a great problem either |
| 06:36.12 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
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| 07:01.58 | gamedna | is there a way to unregister a sip trunk from the CLI? |
| 07:02.07 | gamedna | or force it to re-register |
| 07:02.31 | gamedna | or is restart the only way? |
| 07:02.49 | *** join/#asterisk joobie (~joobie@CPE-121-219-40-191.lnse1.lon.bigpond.net.au) |
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| 07:13.22 | hipitihop | is it possible to time a call duration and interject and playback something which both parties would hear ? |
| 07:16.51 | Alton35 | one party for sure, but can't remember if it's possible for both parties |
| 07:27.54 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 07:29.10 | gamedna | Alton35: how do you do it for one party? |
| 07:36.20 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 07:39.18 | Alton35 | look at the options for the dial() statement |
| 07:39.25 | Alton35 | L I think |
| 07:46.02 | hipitihop | would it be possible to setup time quotas to particular number |
| 07:47.45 | ChannelZ | I've never used them but 1.6.2 anyway has a whole shitload of options for limits and playing sounds to both sides |
| 07:49.41 | ChannelZ | gamedna: sip unregister to make Asterisk forget about a peer, but it's up to that peer to re-register. |
| 07:50.56 | gamedna | can i sip unregister and then do a restart? |
| 07:51.13 | gamedna | hmm |
| 07:51.19 | gamedna | not sure if that will work |
| 07:51.38 | ChannelZ | yes but that still won't necessarily force a peer to re-register |
| 07:51.49 | gamedna | outbound call to a peer wont work either. |
| 07:52.39 | ChannelZ | registration is sort of a one-way thing.. the whole point is so that a remote peer can inform Asterisk (or a proxy, etc) what IP it's at. |
| 07:52.55 | gamedna | true |
| 07:53.22 | gamedna | probably need to setup a test |
| 07:53.31 | gamedna | and see what happens when i drop the peer |
| 07:53.58 | ChannelZ | The peer generally has registration timeouts whereby it will re-register every so often |
| 07:54.46 | gamedna | right, but for some reason this one peer is not disconnecting |
| 07:55.01 | gamedna | the registration stays, and for some reason its not routing calls propertly |
| 07:55.20 | gamedna | its almost like its stale |
| 07:55.59 | gamedna | probably buggy device |
| 07:57.50 | hipitihop | anyone here using FreePBX on ubuntu 9.04 and asterisk ? |
| 07:58.21 | fenrus | i would recommend #freepbx |
| 07:59.33 | ChannelZ | If the device is changing its IP or something but not re-registering, that's the device's fault |
| 07:59.47 | hipitihop | fenrus, what distro are you using ? |
| 07:59.53 | gamedna | thought that was the case, but the device is behind NAT and Static |
| 08:00.03 | gamedna | and there are other devices that are there too |
| 08:00.12 | ChannelZ | 'unregistering' it on the Asterisk side isn't going to cause anything interesting to happen besides for Asterisk to forget its last known IP |
| 08:00.13 | gamedna | none of the other devices have a problem |
| 08:00.43 | ChannelZ | So what happens, does it just become unreachable and/or its unable to make calls to asterisk? |
| 08:00.50 | gamedna | both |
| 08:00.53 | gamedna | cant recieve or send |
| 08:01.13 | gamedna | but it works whenever i powercycle the phone or restart asterisk |
| 08:01.14 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 08:01.20 | ChannelZ | so perhaps the firewall on one end or the other is unmapping the ports. |
| 08:01.23 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 08:01.32 | ChannelZ | Are you using qualify for the peer? |
| 08:01.40 | gamedna | let me check |
| 08:02.16 | ChannelZ | sip show peers should show you under 'status' |
| 08:02.57 | fenrus | hipitihop, i'm using debian |
| 08:03.19 | gamedna | its always been OK ~30ms for the status |
| 08:04.53 | gamedna | qualify is not set |
| 08:05.01 | gamedna | so its whatever the defaults are |
| 08:05.14 | ChannelZ | it probably is globally if you're getting 'OK (xx ms)' for it |
| 08:05.37 | ChannelZ | How long does it take before it becomes unreachable if you do nothing (make no calls, send it no calls) |
| 08:06.00 | gamedna | according to sip show settings.. Qualify: 0 |
| 08:06.24 | gamedna | hmmm |
| 08:06.29 | gamedna | that i am not sure of |
| 08:06.58 | gamedna | i thought it would go down overnight |
| 08:07.06 | gamedna | but it does sometimes and other times it does not |
| 08:07.14 | gamedna | will lock up during the day |
| 08:07.14 | gamedna | etc. |
| 08:07.21 | gamedna | i have a replacement inbound |
| 08:07.36 | gamedna | just has not showed up yet |
| 08:07.43 | gamedna | polycom 550 |
| 08:07.49 | gamedna | the only one that has a problem really |
| 08:08.04 | gamedna | other polycom 550s dont have an issue |
| 08:08.29 | ChannelZ | in the same location? |
| 08:08.40 | gamedna | yes |
| 08:08.59 | ChannelZ | Are they using different ports then? |
| 08:09.07 | gamedna | yes. |
| 08:09.37 | gamedna | swapped netowrk wires, factory reset, firmware reinstalled, changed configs |
| 08:09.56 | gamedna | even swapped ports on the network switch |
| 08:10.02 | gamedna | v. kooky |
| 08:11.44 | gamedna | sounds like its a bug w/ the phone |
| 08:15.29 | ChannelZ | yah dunno |
| 08:17.31 | gamedna | ChannelZ: thanks for the help. |
| 08:18.40 | ChannelZ | or lack thereof :) |
| 08:19.27 | gamedna | ChannelZ: qualify was a big hint as to what was probably going on |
| 08:20.11 | gamedna | will try qualify=yes on monday |
| 08:20.14 | ChannelZ | not if all the other phones are the same, behind the same firewall, and setup the same |
| 08:20.37 | gamedna | ChannelZ: supposedly the same... |
| 08:20.47 | gamedna | could be a id10T error |
| 08:20.50 | gamedna | on my part |
| 08:20.51 | ChannelZ | Do a 'sip show peer xxxx' and see what the Qualify Freq shows up as |
| 08:22.08 | gamedna | dont see it |
| 08:22.20 | ChannelZ | what version of asterisk? |
| 08:22.27 | gamedna | 1.4 |
| 08:22.45 | ChannelZ | ah. Maybe that's a 1.6 thing (or it's called something different in 1.4) |
| 08:22.57 | gamedna | im checking on a 1.6 |
| 08:22.58 | gamedna | right now |
| 08:23.07 | ChannelZ | its near the end, like 6th from last |
| 08:23.12 | gamedna | shows up on my 1.6 |
| 08:23.12 | gamedna | Qualify Freq : 60000 ms |
| 08:23.15 | gamedna | but not the 1.4 |
| 08:23.21 | gamedna | hmm, maybe upgrade? |
| 08:23.22 | gamedna | haha |
| 08:23.35 | ChannelZ | that's default. |
| 08:23.38 | gamedna | been working rock solid for over a year |
| 08:23.43 | gamedna | no config changes |
| 08:23.47 | ChannelZ | It's not that 1.4 doesn't have qualify, just that it doesn't show those settings that way |
| 08:24.02 | gamedna | yea, aware of that... |
| 08:24.32 | gamedna | on both of my servers quality=0 |
| 08:24.34 | ChannelZ | You might set it's qualify down a little, to like 30000 to make sure the NAT firewall isn't losing the port mapping |
| 08:24.36 | gamedna | according to sip show settings |
| 08:24.52 | gamedna | could be my isp |
| 08:24.53 | gamedna | hmm |
| 08:25.03 | gamedna | i remember reading that some ISPs are blocking voip |
| 08:25.17 | gamedna | wonder if its a default 5060 port |
| 08:25.42 | ChannelZ | But I'd think that'd be a flat out block, not 'randomly dropping packets on port 5060' |
| 08:26.18 | gamedna | port is 5062 |
| 08:26.21 | gamedna | hmm |
| 08:27.06 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.95.159) |
| 08:29.05 | FILLVAIO3 | Hi Guys! Is there possible to limit 1-incoming and 2-outgoing call by each SIP account (type=friend) in Asterisk 1.6.2.9-1? |
| 08:29.51 | gamedna | ChannelZ: while i do love diving into asterisk trying to figure this out, i think its probably best if i just #1 swap the phone, #2 upgrade asterisk to the latest 1.6, and #3 make sure the firewall is playing nice |
| 08:34.50 | gamedna | ChannelZ: w/ all that said, it still bugs the crap out of me that i cant really narrow it down |
| 08:35.19 | Alton35 | randomly dropping UDP packets is a symptom of an overloaded/unreliably connection.... |
| 08:35.38 | Alton35 | if you're sure it's that, then maybe something's not right, even a bad cable, hard to say. |
| 08:36.41 | Alton35 | or gotta be a way to test the same thing on another connection |
| 08:36.46 | gamedna | Alton35: firewall load is around 0.04 asterisk box load is around 0.02 |
| 08:36.58 | gamedna | Alton35: will test it when i get the phone back |
| 08:37.21 | Alton35 | I understand. These boxes can push quite a bit of traffic. But cables/etc can make some very weird errors. |
| 08:37.29 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 08:37.49 | gamedna | Alton35: already swapped cables |
| 08:38.03 | Alton35 | ok, just guessing here |
| 08:38.35 | gamedna | its a good suggestion... many overlook that... had them swap phone cable, network port, wall port.. |
| 08:38.52 | gamedna | we replaced the cables on the firewall |
| 08:39.04 | gamedna | all are good cat 5e wires |
| 08:39.21 | Alton35 | g729 or something more demanding of bandwidth? |
| 08:39.36 | gamedna | G711 |
| 08:39.47 | gamedna | but their internet upload is 5Mit |
| 08:39.58 | Alton35 | could be something to test anyway if it's not too difficult for you |
| 08:40.08 | gamedna | switch to G729? |
| 08:40.25 | Alton35 | just to see what happens |
| 08:40.27 | gamedna | i can try... i have a few extra G729 licenses laying around |
| 08:40.45 | Alton35 | although many people just stick with it exclusively, depends on what you are doin |
| 08:40.46 | Alton35 | doing |
| 08:41.10 | gamedna | im thinking of switching to G729 exclusively, but i like G711 for the audio quality... |
| 08:41.11 | Alton35 | I try to do that, even record all files in g729, I guess it's as good a standard as any |
| 08:41.59 | Alton35 | a friend of mine is telling me how he doesn't have any standard on his 2 systems and several phones... but strangely enough has troubles with any recorded audio! so some sort of standard might help him. |
| 08:42.04 | gamedna | do you license the codec's on your * box? or are you just using passthrough? |
| 08:42.24 | Alton35 | I have 10 licenses from the past. |
| 08:42.38 | Alton35 | I think you need them to record and play back files. |
| 08:43.08 | gamedna | i believe you are correct |
| 08:43.42 | gamedna | are there G729 processor cards? |
| 08:44.12 | Alton35 | golly, how many connections are you thinking of? and unless you are transcoding I don't think you need anything like that. |
| 08:44.25 | gamedna | that is for another setu |
| 08:44.34 | gamedna | and yes for transcoding |
| 08:44.41 | Alton35 | aha |
| 08:44.55 | Alton35 | hmm, don't know, kinda avoided that problem |
| 08:45.08 | Alton35 | you might be able to do it anyway, or dedicate a machine or two to it. |
| 08:45.24 | gamedna | one option is to gang all my licenses on two machines |
| 08:45.27 | gamedna | and just have them transcode |
| 08:45.33 | gamedna | and the other just passthrough |
| 08:45.41 | FILLVAIO3 | Hi Guys! Is there possible to limit 1-incoming and 2-outgoing call by each SIP account (type=friend) in Asterisk 1.6.2.9-1? |
| 08:47.27 | gamedna | just thought having a card may make things a bit easier |
| 08:48.12 | Alton35 | have a look at that window |
| 08:50.42 | *** join/#asterisk honree (~s@net2.icemans.co.uk) |
| 08:51.12 | honree | does anyone know how (or even if it's possible) to do immediate dialling on a sipura spa-3102? |
| 08:51.44 | honree | ie do away with the sipura's dial plan strings and just pass all dialled digits immediately to asterisk |
| 08:52.03 | honree | asterisk is the place where that sort of thing should be done, not in each sip fone |
| 08:52.21 | WIMPy | honree: That's an extremely seldom feature for sip. |
| 08:52.33 | honree | what does that mean |
| 08:52.38 | ChannelZ | "no" |
| 08:52.38 | WIMPy | couldn;t agress more |
| 08:52.39 | honree | the grandstream adapters allow it |
| 08:53.14 | WIMPy | Seed to wake up ... :-) |
| 08:53.36 | WIMPy | Need. damn. |
| 08:53.37 | honree | putting dial plans into every sip phone AND in asterisk is dupliction |
| 08:54.17 | honree | what do other people do? |
| 08:54.20 | ChannelZ | SIP doesn't really work like a touch tone phone |
| 08:54.40 | honree | as i said, the grandstream adapters and phones do |
| 08:55.01 | *** join/#asterisk bintut (~bintut@cm224.kappa10.maxonline.com.sg) |
| 08:55.10 | honree | all dialled digits are sent as soon as they're pressed |
| 08:55.11 | ChannelZ | By what mechanism in asterisk does this happen? |
| 08:55.29 | WIMPy | also uses it on the Snom. But that's really a rare thing for sip devices. |
| 08:56.26 | honree | is that because sip is usually point to point? |
| 08:56.44 | WIMPy | The Snom just sends an invite for the complete number after each digit, ignoring errors that indicate an incomplete number. |
| 08:57.05 | WIMPy | No it's because SIP dosn't really support that. |
| 08:57.06 | ChannelZ | that's what I was just going to guess. |
| 08:58.37 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 08:58.56 | honree | wonder why sipura dont add it as a feature |
| 08:58.57 | WIMPy | /If you want your phones to bahave as a normal phone would behave, avoid SIP./ |
| 08:59.11 | ChannelZ | Anyways as far as a phone dialplan goes I've only got a few broad patterns in there, it's not a 'duplicate' of the asterisk dialplan by any means |
| 08:59.57 | honree | how does that work in view of being able to dial via a pstn gateway? |
| 09:00.26 | fenrus | honree, i just thought of that too |
| 09:00.26 | WIMPy | And acutually getting Asterisk to find the end of overlap dialling can be quite challanging as well and in fact prohibits the use of some features. |
| 09:00.30 | fenrus | must be really shitty ;) |
| 09:00.51 | honree | the work around is u can press # when youve finished dialling |
| 09:01.05 | honree | that comes under your really shitty clause imo :) |
| 09:01.07 | ChannelZ | PSTN numbers have a pattern |
| 09:01.07 | WIMPy | (That does not include the emulated sip way) |
| 09:01.39 | fenrus | a number in sweden can vary from three to 16 digits i think |
| 09:01.45 | WIMPy | ChannelZ: Only in some parts of the world. So no. |
| 09:01.49 | honree | similarly in the uk |
| 09:02.08 | beardy | My phones have a timeout (default 5 seconds) after which all pressed digits are sent. |
| 09:02.09 | honree | new numbering schemes mixed with older ones |
| 09:02.11 | fenrus | but sure, "most" works.. |
| 09:02.14 | WIMPy | And in 'evil Germany' :-) |
| 09:02.29 | ChannelZ | So you dial a certain number, and then just sit there and wait for the phone company to 'time out' before it puts the call through? |
| 09:02.31 | fenrus | beardy, that's how my 7960's work too |
| 09:02.41 | honree | no |
| 09:02.41 | beardy | fenrus: (7940s here) |
| 09:02.52 | WIMPy | ChannelZ: No. They know when the number is complete. |
| 09:03.02 | honree | the phone company's exchanges know how to match |
| 09:03.03 | ChannelZ | Like I said. They have a pattern. |
| 09:03.07 | fenrus | ChannelZ, the phone company does not "time out", they have intelligent number analysis in their switches here in sweden ; |
| 09:03.10 | fenrus | ;) |
| 09:03.14 | honree | they have a record of all possible 'dial plans' |
| 09:03.28 | WIMPy | i.e. you type your number and expect to hear a ringing tone immediately after you pressed the last digit. |
| 09:03.34 | honree | aye |
| 09:03.37 | honree | and that works |
| 09:03.39 | fenrus | their "pattern" often exist of hundreds of thousands of numbers |
| 09:03.43 | honree | and asterisk is the place for it |
| 09:03.51 | honree | as that is the equiv of the central office |
| 09:03.58 | honree | certainly not your telephone |
| 09:03.59 | fenrus | because you can move a number betwen different telcos here ;) |
| 09:04.00 | honree | :) |
| 09:04.07 | WIMPy | There is no patterns. It's just the list of all numbers in use. |
| 09:04.15 | beardy | Well, they just read the number, switch if necessary after the "area code", use the rest for customer extension. If the number doesn't match, due to either length or just not used, the proper tones are played back. |
| 09:04.19 | fenrus | but hey, i dont need to do that, because i dont have crap-phones ;) |
| 09:05.04 | honree | grandstreams have a simple drop down box in their config for it |
| 09:05.09 | honree | turn the feature on/off |
| 09:05.12 | honree | end of story |
| 09:05.24 | honree | its not rocket science |
| 09:05.36 | fenrus | case closed, problem solved. |
| 09:05.40 | WIMPy | But I know that this functionality is no very well known in this cimmunity, what unfortunaletly shows in what you can achieve with Asterisk. |
| 09:05.52 | honree | er so anyway the answer to my question appears to be my sipura adapter is rubbish |
| 09:05.58 | honree | in this regard at least... |
| 09:06.17 | fenrus | honree, have you checked all configuration options |
| 09:06.22 | honree | yea :( |
| 09:06.26 | fenrus | that's a shame |
| 09:06.27 | beardy | In general people should drop old expectencies of analog phones.. more than once I've seen people wanting to dumb down SIP to act stupid, aka like an analog phone. |
| 09:06.42 | honree | i was hoping that some sort of special dial plan syntax existed |
| 09:06.45 | WIMPy | fenrus: No not the end of the story. Your dialplan will come with some challanges as well. |
| 09:07.24 | WIMPy | beardy: Yes, but not for the worse. |
| 09:07.25 | fenrus | WIMPy, my dialplan is quite easy since i only have one telco, and connected to it via sip.. :) |
| 09:07.34 | fenrus | s/telco/provider |
| 09:07.46 | honree | eardy im not advocting removing the dial plan capability from sip fones. |
| 09:07.59 | fenrus | but the company i work for has 8 nokia AXE's and a couple of ericsson ones ;) |
| 09:08.02 | honree | im merely saying that /as an option/ it would be nice to be able to bypass it |
| 09:08.04 | WIMPy | I have some hard time getting dialling to local exensions right. |
| 09:08.18 | fenrus | WIMPy, oh, okay - what seems to be the problem? :) |
| 09:08.41 | WIMPy | The part where you want to be able to dial them just using the local number as well as with their areacode added. |
| 09:09.09 | WIMPy | honree: I'm absolutely on your side. |
| 09:09.22 | fenrus | honree, did you check for firmware upgrades? |
| 09:09.38 | beardy | (I just made a comment, not on anyone's side. ;) ) |
| 09:09.40 | WIMPy | The only way I found that to work is to add the extensions twice. |
| 09:10.02 | WIMPy | Well, actually 4 times, as I want it with international prefix as well. |
| 09:10.03 | ChannelZ | you could do something gay like make a single digit in your phone's dialplan to start the call.. that hits a matching extension in your * dialplan which then just answers, does a WaitExten and the rest gets handled by your asterisk dialplan |
| 09:10.21 | WIMPy | But that only works as long as all numbers are from the same area code. |
| 09:11.07 | WIMPy | ChannelZ: Yes, but I guess it would fail if you dial from phonebook. |
| 09:11.51 | fenrus | WIMPy, you want to force a call going to a local phone to not leave your pbx, turn at your provider and get back to your pbx and the correct phone ? |
| 09:12.09 | WIMPy | Sure! |
| 09:12.36 | beardy | I guess you could match on areacode+yourprefix+$lengthofextension with two IF:s, letting through lower than, and higher than, your first and last extension? |
| 09:12.38 | WIMPy | In fact it wouldn't work otherwise as not all numbers are hosted at the same provider. |
| 09:13.25 | fenrus | sounds like that it'll be quite a long dialplan then ;) |
| 09:13.27 | beardy | (That's for a series that is..) |
| 09:14.07 | WIMPy | beardy: No, a) you that length aproach wouldn' work here and b) you'd need to match before you know the number is complete which breaks dialling. |
| 09:14.31 | WIMPy | -you |
| 09:18.22 | WIMPy | Actually it even prevents me from having a default route. |
| 09:18.22 | fenrus | of course you're using ranges in the dialplan, but if there's alot of them it'll be some config to mess around with.. |
| 09:18.23 | ChannelZ | How come the Sipura phones don't have a config option for reading minds so it knows what number I intended to dial and when I'm done? |
| 09:18.23 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 09:18.23 | fenrus | ChannelZ, just add a feature request :) |
| 09:18.23 | ChannelZ | people lament about how steering digits are evil |
| 09:18.23 | honree | it doesnt need to read minds or anything like that |
| 09:18.24 | honree | all it needs to do is blindly pass dialled digits to asterisk |
| 09:18.24 | honree | its very very simple |
| 09:20.12 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 09:20.12 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta3 (2010/08/10), 1.6.2.11 (2010/08/10), 1.4.35 (2010/08/10), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 09:20.13 | WIMPy | Don't use extensions that clash with real numbers. |
| 09:20.35 | ChannelZ | Yes but "dial as you go" via SIP is going to hit extension 303 as soon as you type the second 3, even though you intended to continue on with 5551212 |
| 09:20.55 | honree | same if u do it in the sip fone |
| 09:21.17 | WIMPy | Correct? |
| 09:21.20 | beardy | ChannelZ: It's routed according to best match. If you have an "5 digits+whatever" extension, that is used. |
| 09:21.25 | beardy | ChannelZ: Yes, it would. |
| 09:22.01 | ChannelZ | except you can put a pause in the phone's dialplan so it would only accept 303 if you stopped dialing more digits for a second etc. I don't think you can do the same in the * dialplan |
| 09:22.14 | honree | if u have some sort of 'escape' like pressing pound before the special range then u dont need to do that in the sip fone dial plan u cn do it in asterisk |
| 09:22.36 | beardy | ChannelZ: But it would be a bad design choice to have local extensions matching known areacodes or extensions. |
| 09:22.47 | honree | ok well that sounds a bit odd way of having your dialling |
| 09:24.19 | honree | sigh oh well i spose ill have to just do it the long way :) |
| 09:25.04 | ChannelZ | Odd is a localized point of view |
| 09:25.12 | honree | :) |
| 09:25.30 | honree | well odd as in i dont know of any other system tht does that |
| 09:25.38 | ChannelZ | does what |
| 09:27.08 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 09:27.49 | honree | what u said |
| 09:28.23 | honree | has a system where there are two (or more) overlapping number ranges, differentiated by an arcane system of the caller pausing during their dialling |
| 09:32.48 | ChannelZ | it was just an example; I just use a steering digit which is perhaps old-fashioned, but keeps me from having to write some crazy-ass dialplan, regardless whether it's in the phone or in asterisk |
| 09:33.02 | ChannelZ | It also helps that in the US our phone numbers are a consistent length |
| 09:33.31 | honree | heh ok |
| 09:38.40 | WIMPy | Yes some sort of prefix IS a good idea. |
| 09:39.16 | WIMPy | I however always preferred the idea of using * as an internal prefix. |
| 09:40.01 | ChannelZ | Yeah I should have put all my local extensions as *xxx or #xxx |
| 09:41.33 | WIMPy | But that clashes with standard service codes. And as we can't use normal facility based signalling with Asterisk, they seem like a good idea. |
| 09:41.41 | WIMPy | (read workaround) |
| 09:44.33 | WIMPy | Maybe if it becomes a good idea to swaitch to dahdi for bri in the (hopefully) near future, a patch for facilities might be possible. |
| 09:44.55 | honree | it might depend on the location your in in the uk, but i /think/ i can get away with overloading local 2xx and 3xx numbers |
| 09:45.09 | WIMPy | But it might be a good idea to add screening capabilities first. |
| 09:45.16 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 09:45.41 | WIMPy | The word "think" makes me shiver. |
| 09:46.19 | WIMPy | Reminds me of adapters that used 999 as configuration prefix as someone *thought* that was safe. |
| 09:46.55 | honree | heh |
| 09:47.00 | honree | lol |
| 09:47.41 | honree | well if the emergency number changes to 2xx something, ill be knackered |
| 09:48.02 | honree | 999 i always thought was a poor choice of number |
| 09:48.15 | honree | especially as it came from rotary dial days |
| 09:48.36 | honree | someone must have thought mmm let's see what 3 digit number takes longest to dial in an emergency... |
| 09:48.54 | honree | 911 is better |
| 09:49.08 | honree | maybe they thought 999 was easier to remember |
| 09:50.33 | WIMPy | Oh right, 999 is even more interesting in the uk. Here they are just ordinary local numbers. |
| 09:50.51 | honree | 999 is a local number? |
| 09:50.55 | honree | local police station... |
| 09:51.48 | WIMPy | No, ordinary subscriber numbers. Probably longer than 3 digits. |
| 09:52.14 | honree | i think 911 and 112 work here also |
| 09:52.29 | honree | altho not widely publicised - so as not to confuse people i spose |
| 09:52.43 | WIMPy | But I imagine those adapters would have been interestin in th uk. If you have an emergency it's surely interesting to find out that you actually can't call for help. |
| 09:52.50 | honree | yet at the same tme allow foreign visitors to use the number theyre familiar with |
| 09:53.17 | honree | yes and have to log into asterisk to edit the dial plan while your house is burning down |
| 09:53.35 | honree | then find that you forgot to edit the dial plan on your sipura... |
| 09:53.43 | WIMPy | With a black box there is no logging in and editing anything. |
| 09:53.55 | WIMPy | Errr. Yes. |
| 09:54.07 | WIMPy | hates dialplans in terminals! |
| 09:54.09 | EmleyMoor | honree: 999 was invented because it was easier to modify the old payphones to dial it free than any other reasonable number |
| 09:54.19 | WIMPy | That's definitely not where they belong. |
| 09:55.05 | honree | payphone? |
| 09:55.37 | honree | i shouldnt think the mod was done in the payphine itself but i take your point |
| 09:55.53 | WIMPy | Ooooh. That reminds me of those really old payphones that had an extra emergency box with a lever to turn left or right for either police or the fire brigade. |
| 09:56.14 | honree | haha yea |
| 09:56.31 | EmleyMoor | honree: It was - it was a short circuit on the dial - originally released by inserting minimum fee |
| 09:57.02 | honree | interesting |
| 09:57.03 | EmleyMoor | It was on 1-9 until 999 came in, then they modified the dial so that 9 would dial as well |
| 09:57.19 | honree | ahh |
| 09:57.22 | honree | cunning |
| 09:57.39 | honree | so u could actually dial 9 or 99 or 9999 for free too... |
| 09:57.41 | WIMPy | Ha, found one on wiki. A newer type of payphone but with the emergency box still in place. |
| 09:57.41 | honree | ;) |
| 09:57.46 | EmleyMoor | We can use 112 now (EU standard) but not 911 |
| 09:57.48 | WIMPy | http://upload.wikimedia.org/wikipedia/de/0/01/MTZ-Tasten_Notrufhebel.jpg |
| 09:58.33 | EmleyMoor | (we used 0 for operator back then of course) |
| 09:59.47 | EmleyMoor | BTW are you the honree I know from elsewhere? |
| 09:59.52 | *** join/#asterisk Jasnejac (~kvirc@81.91.106.59) |
| 10:00.43 | honree | if you are the emleymoor i know from there... |
| 10:00.43 | honree | ;) |
| 10:00.47 | honree | small world innit |
| 10:00.53 | EmleyMoor | Yes, rather |
| 10:01.17 | honree | do u work in telecoms? |
| 10:01.49 | EmleyMoor | Yes |
| 10:03.09 | *** join/#asterisk comsol (~Manipur@117.242.156.60) |
| 10:03.14 | comsol | hi |
| 10:03.20 | *** join/#asterisk m0t3jl (~petr.mote@193.85.113.247) |
| 10:03.25 | honree | what company/area? |
| 10:03.25 | comsol | i need help on asterisk |
| 10:03.51 | EmleyMoor | Cable&Wireless Worldwide, London, mostly in acceptance testing |
| 10:03.55 | comsol | i want to setup a no. for so that every user on extension can listen to simultaneously |
| 10:04.00 | honree | i used to work at GEC/GPT on system x |
| 10:04.18 | honree | i was in the processor group (PUS) |
| 10:04.24 | honree | (popus) |
| 10:04.30 | comsol | and the audio source is to be from streaming server |
| 10:04.53 | honree | although i was only really involved in teh operating system on the processor i didnt have much to do with CPS |
| 10:05.20 | EmleyMoor | My asterisk use is purely domestic. btw |
| 10:05.48 | honree | so is mine really |
| 10:05.52 | comsol | anybody who is asterisk expert |
| 10:05.55 | comsol | plz help me |
| 10:06.35 | honree | i have a vpn voip connection to a friends and my office number is routed thru it but that's about it |
| 10:07.08 | honree | comsol sorry - not ignoring u - i just dont know the answer |
| 10:07.28 | EmleyMoor | comsol: Same here sorry |
| 10:07.37 | comsol | any other guys |
| 10:07.57 | honree | id be mildly interested in the answer too |
| 10:08.11 | honree | im just fiddling with ring cadences now... |
| 10:08.28 | EmleyMoor | I too am interested in the answer |
| 10:08.37 | EmleyMoor | honree: Fiddling with them in what way? |
| 10:08.50 | honree | well trying to get them to work basically |
| 10:09.04 | WIMPy | comsol: MusicOnHold |
| 10:09.04 | honree | differentiate between an external and an internally originated call |
| 10:09.17 | honree | i can do music on hold |
| 10:09.17 | EmleyMoor | honree: Zap/DAHDI? |
| 10:09.31 | honree | but i think he means live streaming |
| 10:09.39 | honree | em: dunno |
| 10:09.57 | honree | ive googled and found : |
| 10:09.58 | comsol | how to listen to music on hold or any no. to listen music on hold |
| 10:09.59 | honree | exten => 5555,1,SetVar(_ALERT_INFO=Bellcore-r7) ; |
| 10:10.26 | EmleyMoor | honree: I know how to do it for Zap/DAHDI |
| 10:10.43 | honree | this is on sip |
| 10:11.03 | EmleyMoor | Ah - no idea on SIP - got nothing that supports them |
| 10:11.10 | honree | heh ok |
| 10:11.27 | honree | not sure if the fone will even pass the cadence thru... |
| 10:11.39 | comsol | how to listen to musiconhold |
| 10:11.46 | comsol | any specific no? |
| 10:12.06 | EmleyMoor | comsol: Set one up that calls the app MusicOnHold |
| 10:12.09 | honree | exten => 1201,1,MP3Player(/var/lib/asterisk/sounds/MOCR.mp3) |
| 10:12.36 | EmleyMoor | I have a MOH-test in among mine |
| 10:12.51 | honree | see if u can guess what that is ;) |
| 10:12.59 | honree | its a test yes... but what of |
| 10:13.28 | comsol | how to setup? |
| 10:14.22 | WIMPy | comsol: |
| 10:14.47 | WIMPy | ~book |
| 10:14.47 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 10:16.49 | honree | mmm adding that bellcore cadence line seems to have stopped the phone ringing altogether |
| 10:17.47 | EmleyMoor | honree: I suppose some change is better than no change... |
| 10:19.11 | honree | heh |
| 10:19.19 | EmleyMoor | My system currently supports full London-style dialing btw |
| 10:19.47 | honree | mine works ok with cov local and also 18866... |
| 10:19.55 | honree | automatically adding the appropriate prefix |
| 10:20.09 | EmleyMoor | Might move my internal extensions about a bit depending on what happens with local numbers |
| 10:20.09 | honree | so i can dial a local or national number and it does the right thing |
| 10:20.54 | EmleyMoor | At the moment, _[378]XXXXXXX works here |
| 10:23.04 | honree | right |
| 10:24.38 | EmleyMoor | My internals have the last four digits of their appropriate DID at present. Would have gone for last 3 if that hadn't have meant starting with 8 |
| 10:27.12 | EmleyMoor | Non-DID tend to begin with the same digit |
| 10:27.36 | EmleyMoor | Do all Coventry numbers still begin 76 or have they launched any new ranges yet? |
| 10:27.37 | honree | from what i can tell this dect phone doesnt support different ring cadences |
| 10:28.03 | honree | funny u should say that i think there's a new ranging coming soon |
| 10:28.05 | honree | 78? |
| 10:28.07 | honree | cant recall |
| 10:30.06 | honree | actually this phone is registered on a different base |
| 10:30.18 | honree | so its possible not all features of the phone can be used |
| 10:30.28 | honree | just lowest common dect feature set |
| 10:30.49 | WIMPy | "gap" |
| 10:30.56 | honree | possibly |
| 10:31.12 | honree | dunno if gap mandates caller display? |
| 10:31.37 | WIMPy | Not sure. But never had any issues with that. |
| 10:32.09 | EmleyMoor | honree: I doubt they'll use a range not beginnig with 7 in Coventry for a while unless and until they expand it out to, say, Warwick |
| 10:33.04 | honree | 01926 |
| 10:33.26 | honree | well ive just about got this working well enuf now |
| 10:33.31 | EmleyMoor | 024 70 is seemingly allocated |
| 10:33.50 | honree | it does a windows messenger broadcast of the caller info to anyone on the lan who cares to listen |
| 10:33.57 | honree | little popup of who's calling |
| 10:34.24 | EmleyMoor | honree: I send caller IDs over Jabber |
| 10:34.29 | honree | i also wrote a simple filter program that consults a flat file for barred numbers or categories and transfers them straight to voicemail |
| 10:34.56 | EmleyMoor | I use the DB at present for stuff like that |
| 10:35.14 | honree | cool |
| 10:35.32 | honree | i chose messenger as it's on all pcs as standard |
| 10:35.38 | honree | (albeit not turned on) |
| 10:35.45 | EmleyMoor | It gets to be a pig though - we get a lot of calls from "that can't be right" caller IDs |
| 10:35.56 | honree | heh |
| 10:36.02 | honree | on pstn? |
| 10:36.14 | honree | i used to get all sorts on the gradwell office line |
| 10:36.18 | EmleyMoor | 0000000000, 009145... on IAX |
| 10:36.20 | honree | i moved it to a hardwired bt line |
| 10:37.24 | EmleyMoor | I suppose Meriden is actually the most likely "next intake" for 024 |
| 10:37.45 | WIMPy | Hmm. Is there an issue with pri set debug file in 1.8? There seems to appear no data. |
| 10:42.15 | WIMPy | And what's a channel type 1? |
| 11:26.11 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
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| 12:31.48 | *** join/#asterisk ahuemer (ahuemer@83-65-26-208.static.xdsl-line.inode.at) |
| 12:32.38 | ahuemer | hi, i have a problem with the asterisk build process |
| 12:32.59 | ahuemer | i.e. with the variables for the used compiler |
| 12:33.33 | honree | has anyone tried running asterisk on an alix board? should work i think... |
| 12:33.52 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 12:34.06 | ahuemer | i want ./configure and make to use something as compiler that is user given |
| 12:34.13 | ahuemer | but i don't know what to do |
| 12:34.20 | ahuemer | could somebody please help me ? |
| 12:35.01 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 12:35.51 | ahuemer | there is a definition for CC in makeopts.in, but setting CC for configure does not have any effect when running make |
| 12:35.56 | ahuemer | any ideas ? |
| 12:36.02 | EmleyMoor | ahuemer: How did you set it? |
| 12:36.55 | ahuemer | EmleyMoor: thanks for your answer. one sec please |
| 12:37.00 | honree | any reason youre not using the system CC? |
| 12:37.12 | ahuemer | yes |
| 12:38.00 | ahuemer | i want to build asterisk in a distcc environment and i have problems with that. although i believe asterisk _would_ compile find with just the right variables |
| 12:38.12 | ahuemer | CC=x86_64-pc-linux-gnu-gcc ./configure |
| 12:38.26 | ahuemer | when i run make afterwards, i get as first line |
| 12:38.37 | ahuemer | CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts |
| 12:38.41 | ahuemer | which is not what i want |
| 12:38.55 | EmleyMoor | ahuemer: Try the other way round - you want it as a parameter, not an environment variable |
| 12:39.23 | ahuemer | running |
| 12:39.57 | ahuemer | i did |
| 12:39.59 | ahuemer | ./configure CC=x86_64-pc-linux-gnu-gcc |
| 12:40.00 | ahuemer | and got |
| 12:40.06 | ahuemer | CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts |
| 12:43.17 | ahuemer | i tried even this |
| 12:43.19 | ahuemer | ./configure HOST_CC=x86_64-pc-linux-gnu-gcc BUILD_CC=x86_64-pc-linux-gnu-gcc CC=x86_64-pc-linux-gnu-gcc |
| 12:43.32 | *** join/#asterisk bintut (~bintut@cm224.kappa10.maxonline.com.sg) |
| 12:43.35 | ahuemer | since i found these 3 variables in makeopts |
| 12:48.35 | ahuemer | EmleyMoor: anything i could try ? |
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| 13:00.56 | ahuemer | nobody ? |
| 13:04.26 | honree | arent they supposed to be paths? |
| 13:04.44 | ahuemer | ? |
| 13:15.28 | *** join/#asterisk ahuemer (ahuemer@83-65-26-208.static.xdsl-line.inode.at) |
| 13:15.41 | ahuemer | sorry, my screen session died |
| 13:15.48 | ahuemer | any ideas ? |
| 13:20.16 | *** join/#asterisk Rajaie (~IceChat09@a58-234.adsl.paltel.net) |
| 13:20.51 | Rajaie | hi |
| 13:20.59 | Rajaie | anybody here can help ? |
| 13:22.10 | UQlev | !seen anybody |
| 13:23.14 | Rajaie | hi , I have bought A TE405P card , it was 5th generation card |
| 13:24.01 | Rajaie | i am losing the d-channel whil the pri is up |
| 13:24.20 | Rajaie | I have tried the configuration on 4th gneration card with the same configuration and it went fine |
| 13:24.45 | Rajaie | Pri have been shown to be tested on other working servers ; so we are sure there is no problem with span 1 & span 3 ( span 2 and span 4 are not working) |
| 13:25.03 | Rajaie | <PROTECTED> |
| 13:25.15 | Rajaie | pri show spans is showing the following |
| 13:25.26 | Rajaie | PRI span 1/0: Provisioned, Down, Active |
| 13:25.34 | UQlev | Rajaie: regret I have never used tel cards yet |
| 13:25.51 | Rajaie | where I can have help regarding this issue , I am running out of time |
| 13:26.57 | Rajaie | sorry to bother you UQlev .. thank you bye |
| 13:30.57 | UQlev | Rajaie: http://www.asteriskguru.com/tutorials/wildcard_te405p_wildcard_te410p.html < have you seen it? |
| 13:31.23 | *** join/#asterisk bintut (~bintut@cm224.kappa10.maxonline.com.sg) |
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| 13:38.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:38.35 | Rajaie | UQlev: very usefull always used it in my configuration |
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| 13:52.07 | ariel_ | ~book |
| 13:52.08 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 15:54.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 16:15.38 | *** join/#asterisk Carp1 (~none@cpe-24-92-37-23.nycap.res.rr.com) |
| 16:15.44 | Carp1 | Hey. |
| 16:15.50 | honree | <PROTECTED> |
| 16:16.08 | Carp1 | Can I ass SIP users via the manager? |
| 16:16.09 | Carp1 | api |
| 16:16.36 | [TK]D-Fender | pardon? |
| 16:17.11 | Carp1 | using the API |
| 16:17.16 | fenrus | (har)ass ? |
| 16:17.32 | Carp1 | Or do I have to manually edit the file? |
| 16:17.33 | fenrus | ass(isst)? |
| 16:17.36 | [TK]D-Fender | "ass" is a nound, not a verb |
| 16:17.41 | [TK]D-Fender | noun* |
| 16:17.47 | fenrus | ah, you mean add ? |
| 16:18.10 | Carp1 | lol |
| 16:18.19 | Carp1 | i was like what the hell are they talking about haha |
| 16:18.24 | Carp1 | yes, add* |
| 16:18.43 | [TK]D-Fender | Carp1: Nothing vaguely resembling a "natural" way. |
| 16:19.12 | Carp1 | slaps Carp1 around a bit with a large trout |
| 16:19.17 | Carp1 | whoops |
| 16:19.18 | Carp1 | lol |
| 16:20.34 | honree | haha |
| 16:22.38 | Carp1 | or maybe I should ask this....Whats the best way to add a SIP user via a PHP script? :) |
| 16:23.11 | [TK]D-Fender | Carp1: Where does a "sip user"'s definistion EXIST? |
| 16:23.42 | p3nguin | sip.conf, type=user? |
| 16:24.34 | Carp1 | Sorry [TK] I don't understabd.. |
| 16:24.37 | Carp1 | understand* |
| 16:25.05 | [TK]D-Fender | [12:23]<[TK]D-Fender>Carp1: Where does a "sip user"'s definistion EXIST? <- what stores the data that DEFINES the account |
| 16:25.30 | p3nguin | oh |
| 16:25.59 | Carp1 | oh... |
| 16:26.21 | Carp1 | Sorry, its on FreePBX. I wasn't thinking...Does that use MySQL? |
| 16:26.44 | [TK]D-Fender | Carp1: 2nd channel to your left. Go. Now. |
| 16:27.35 | Carp1 | :) |
| 16:27.52 | Carp1 | I was here because it wasn't really a FreePBX question |
| 16:28.26 | [TK]D-Fender | carYou don't even know WHERE the thing you want to change is |
| 16:28.35 | [TK]D-Fender | Carp1: You can't even tell your head from your ass on this subject |
| 16:29.00 | [TK]D-Fender | Carp1: And the answer is VERY different because of FreePBX |
| 16:29.53 | Carp1 | easy killer |
| 16:29.54 | Carp1 | sorry |
| 16:30.39 | Carp1 | and you're right....otherwise I wouldn't be here asking, right? |
| 16:31.06 | [TK]D-Fender | [12:27]<Carp1>I was here because it wasn't really a FreePBX question <- it IS. |
| 16:31.25 | [TK]D-Fender | Carp1: Because anything you tend to do outside of it gets blown away by it later |
| 16:38.34 | Alton35 | Well, I'm back with the same question, how to originate calls and get back a dialstatus. |
| 16:38.51 | Alton35 | I tried AMI and it makes calls but really no different than creating a call file as far as I can see. |
| 16:39.42 | [TK]D-Fender | Alton35: Get back where/how? |
| 16:40.33 | Alton35 | I guess you tell me. I haven't found any way to get it back, |
| 16:41.12 | Alton35 | <PROTECTED> |
| 16:42.09 | [TK]D-Fender | Alton35: Then dial a LOCAL channel and have that do the outbound attempt AND cleanup for you. |
| 16:42.59 | Alton35 | I thought of that. In that case I'd be inside of my AGI and be quite happy, |
| 16:43.07 | *** join/#asterisk d00gster (~dt@77.31.90.141) |
| 16:43.10 | Alton35 | just not sure how to dial a local channel. :-) Can you give me an idea? |
| 16:43.15 | [TK]D-Fender | Alton35: If you need it, sure, why not |
| 16:43.37 | [TK]D-Fender | Alton35: Local/extension@context/n |
| 16:44.27 | Alton35 | I will read up on it. I assume it "always works" and basically connects me to a dummy channel.? |
| 16:44.39 | *** join/#asterisk xheliox (jeff@heliosj.iddings.us) |
| 16:45.09 | [TK]D-Fender | Alton35: Chan_local <- Learn it. It is what makes * really usable |
| 16:45.44 | Alton35 | good, I knew I just had to keep asking. :-) let me read a bit, and I appreciate it. |
| 17:07.33 | p3nguin | Is /n an undocumented option for chan_local? I've never been able to find out anything about it. |
| 17:07.36 | *** join/#asterisk BANSAL (~bansal@117.199.127.186) |
| 17:08.01 | leifmadsen | p3nguin: then you didn't read doc/tex/localchannel.tex |
| 17:08.44 | p3nguin | I admit I don't know my way around the included documentation. |
| 17:09.11 | [TK]D-Fender | p3nguin: No, there are comments about it... normally when a local channel calls out and gets bridged, the local channel is dissolved, and the outer channels get bridged directly. This means post-call processes, etc usually get cut off. "/n" forces chan_local to always sit as the intermediary |
| 17:10.26 | leifmadsen | i.e. /n makes chan_local act as a "real channel" which lives and is not processed out of the call |
| 17:12.06 | [TK]D-Fender | p3nguin: It is a great way around all sorts of things users find problematic. |
| 17:13.34 | p3nguin | I use local channels for several things, but I've never used /n on any of them. I've never known why to or when to. |
| 17:13.35 | *** join/#asterisk ibercom (58006256@gateway/web/freenode/ip.88.0.98.86) |
| 17:16.03 | leifmadsen | p3nguin: all explained in that file I told you about |
| 17:16.11 | leifmadsen | it's a very good read for people who use chan_local |
| 17:16.12 | p3nguin | Would it be fair to say that the /n should be used any time you dial a local channel and want to have additional priorities for the exten after the Dial command? |
| 17:16.14 | leifmadsen | (or those who want to) |
| 17:16.21 | leifmadsen | p3nguin: read the file -- it'll answer many questions |
| 17:16.54 | leifmadsen | grab the latest 1.6.2.11 and read the text file if you're bothered by the latex formatting -- both a .txt and .pdf version of the documention is generated for you |
| 17:21.17 | p3nguin | I've got a basic txt in 1.4.34. |
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| 17:28.29 | *** join/#asterisk Tigerplug (~chatzilla@089-101-055043.ntlworld.ie) |
| 17:29.20 | Tigerplug | Hi all, I'm new to asterisk and just about to build a testbox. I'm thinking about CentOS 5.4 and compile asterisk from source? - is there any better choice? |
| 17:29.51 | ChannelZ | uh oh, distro wars.. |
| 17:30.18 | xheliox | Tigerplug: That's the way I'd go. But you'll probably get 5 different answers. |
| 17:30.44 | xheliox | Use whatever you're most comfortable with, really.. there's no much difference in what's under the hood from distro to distro, just different styles realy. |
| 17:30.47 | xheliox | l |
| 17:31.26 | Tigerplug | xheliox: thanks :-) |
| 17:38.16 | [TK]D-Fender | Tigerplug: Good way to start |
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| 17:58.02 | p3nguin | Is there any way to craft an exten so that when phone1 makes a call, the call goes where it is supposed to go plus the exten somehow places a call to phone2 and launches ExtenSpy or ChanSpy? |
| 17:58.28 | p3nguin | When calling two local channels with one Dial, whichever device answers first wins. So that doesn't work. |
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| 18:03.11 | p3nguin | The purpose would be to monitor calls from a specific phone without recording the calls. |
| 18:08.23 | *** join/#asterisk darkpixel (~darkpixel@173-12-179-89-oregon.hfc.comcastbusiness.net) |
| 18:09.34 | p3nguin | Maybe use Dial's G option and dump the calls into a MeetMe? Would that work? |
| 18:11.24 | p3nguin | I'd rather just use ChanSpy or ExtenSpy if possible. |
| 18:11.54 | darkpixel | I'm trying to find a clearer definition of the AMD return variable AMDCAUSE. Using the Ubuntu-installed defaults, AMD consistently returns 'TOOLONG-5500'. I've tried answering the phone saying 'Hello?', 'This is Aaron', "Big Bob's Pizza, what can I get you", and I've tried mixing up the amount of time it takes to get the phone up to my mouth to start talking. AMD always returns TOOLONG-5500. What is too long? ... |
| 18:12.00 | darkpixel | ... The greeting? The silence? The docs and google aren't any help. |
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| 18:22.04 | p3nguin | I just don't know how to get started. Can someone give me some ideas? |
| 18:23.58 | [TK]D-Fender | p3nguin: M() + Originate |
| 18:24.10 | [TK]D-Fender | p3nguin: Spawn the spying call-out. |
| 18:25.10 | [TK]D-Fender | p3nguin: You could spawn a call to meetme + chanspy so the person spying can "log in" to it without having be be waiting and that also lets multiple people jump in |
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| 18:50.32 | p3nguin | I don't have any Originate dialplan application on 1.4.34, so I guess I am going to be using the MeetMe after all. |
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| 18:56.22 | [TK]D-Fender | p3nguin: ... AMI / CLI Originate .. or call file.... |
| 18:56.36 | [TK]D-Fender | p3nguin: Immediate meetme = trouble |
| 18:56.48 | p3nguin | I want this to be purely in dialplan. |
| 18:57.02 | [TK]D-Fender | p3nguin: CLI it is then |
| 18:57.17 | p3nguin | No user interaction. All dialplan. |
| 18:57.22 | [TK]D-Fender | ..... |
| 18:57.35 | [TK]D-Fender | p3nguin: Holy crap think a little when i tell you that is the answer... |
| 18:58.08 | *** join/#asterisk guilhermebr (~Guilherme@189.63.72.101) |
| 18:58.29 | [TK]D-Fender | p3nguin: M() + System(/usr/bin/asterisk -rx 'originate QWERTYUIOP') |
| 18:58.31 | [TK]D-Fender | :p |
| 19:01.54 | p3nguin | Very interesting. |
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| 19:33.51 | honree | adios folks |
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| 19:52.05 | JerJer | ugh i really hate app_queue |
| 19:52.46 | JerJer | why would it keep sending calls to a given member when that member already has an active call ? |
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| 20:02.07 | p3nguin | Using the System(asterisk -rx "originate ...") method in dialplan, is there no way to set the CALLERID information that will be sent to the phone through the originate? |
| 20:02.11 | *** join/#asterisk guilhermebr (~Guilherme@189.63.72.101) |
| 20:03.25 | p3nguin | Using Set() right before the System() does not do it. |
| 20:23.39 | [TK]D-Fender | p3nguin: No, in that case you might instead build a callfile with System calls (pure dialplan) and in there yuo can set those |
| 20:24.03 | [TK]D-Fender | JerJer: perhaps you could try looking at the actual calls in progress and what it's making... |
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| 20:42.14 | JerJer | heh ? i have one caller active in the queue and when a second one comes in it calls the already active member |
| 20:42.56 | [TK]D-Fender | JerJer: So I see...... |
| 20:43.03 | [TK]D-Fender | JerJer: Oh wait... I DON'T :p |
| 20:43.25 | JerJer | very simple ... member Bob has an active call, app_queue gets another call and sends it to member Bob |
| 20:44.13 | JerJer | instead of member Sally |
| 20:44.22 | JerJer | who does not have any calls |
| 20:46.11 | seanbright | ringinuse |
| 20:48.33 | [TK]D-Fender | JerJer: Remove story, insert configs and call debug |
| 20:48.54 | seanbright | [queue_name] |
| 20:48.55 | seanbright | ... |
| 20:48.58 | seanbright | ringinuse=no |
| 20:50.33 | JerJer | i can see why people have problems with getting help |
| 20:50.48 | [TK]D-Fender | I certainly can. |
| 20:51.57 | seanbright | ringinuse=no |
| 20:53.14 | JerJer | seanbright: obviously i have that set |
| 20:53.23 | JerJer | this is not my first rodeo |
| 20:53.35 | seanbright | oh good |
| 20:53.39 | seanbright | what version of asterisk? |
| 20:53.50 | [TK]D-Fender | Yes, but we've found our clown already ;) |
| 20:53.57 | JerJer | 1.6 svn from two days ago |
| 20:54.11 | seanbright | 1.6.what? |
| 20:54.25 | seanbright | there are 3 1.6 branches |
| 20:54.46 | JerJer | so there must be 3 different versions of app_queue then huh ? |
| 20:55.01 | seanbright | cool |
| 20:55.05 | seanbright | sorry i couldn't help |
| 20:55.15 | [TK]D-Fender | seanbright: Its ok.... |
| 20:56.27 | [TK]D-Fender | seanbright: You can't save all of them... |
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| 21:17.44 | seanbright | JerJer: when the first rep is on the call, what does 'queue show' and 'core show hints' output? |
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