| 00:02.04 | saxa | p3nguin: i need to send you a box of bears already |
| 00:02.14 | saxa | [TK]D-Fender: you are invited too :) |
| 00:02.22 | saxa | it works now |
| 00:02.27 | p3nguin | What kind of bears are we talking about? |
| 00:02.29 | saxa | many thanks |
| 00:02.45 | saxa | big ones := |
| 00:02.53 | saxa | s/bear/beer |
| 00:02.59 | saxa | s/bear/beers |
| 00:03.09 | saxa | you choose |
| 00:03.14 | p3nguin | s/bear/beer/ |
| 00:03.25 | saxa | can be a good grilled bear also :) |
| 00:03.43 | [TK]D-Fender | I want mine live. |
| 00:03.48 | [TK]D-Fender | And riding a unicycle |
| 00:03.48 | saxa | yeay |
| 00:03.51 | p3nguin | Do they even have bears in Italy? |
| 00:03.59 | saxa | that would be difficult |
| 00:04.10 | saxa | yes, few exemplars |
| 00:04.19 | saxa | but I can get one in Slovenija |
| 00:04.20 | ChrisInSydney | http://upload.wikimedia.org/wikipedia/commons/thumb/e/ef/Friendly_Male_Koala.JPG/220px-Friendly_Male_Koala.JPG |
| 00:04.30 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 00:04.57 | p3nguin | But a Koala is not a bear. |
| 00:05.07 | [TK]D-Fender | indeed |
| 00:05.10 | ChrisInSydney | true |
| 00:05.19 | ChrisInSydney | but its the closest thng we have |
| 00:06.33 | ChrisInSydney | mind you there are plenty of otehr things to kill you here |
| 00:07.29 | saxa | heh, we have about 200-300 pieces here in Slovenija, about 100km from where i am |
| 00:07.39 | saxa | of course not Koalas |
| 00:07.44 | saxa | but bears |
| 00:08.10 | saxa | koala also should be in the zoo. Probably 1 piece only :D |
| 00:09.20 | ChrisInSydney | Japanses bought them all in the 80s |
| 00:09.32 | ChrisInSydney | Japanese |
| 00:10.10 | saxa | but ok, going back to my iax.conf, what those username= statements say, the name of the opposite side box who will register ? |
| 00:10.28 | p3nguin | No. |
| 00:10.57 | p3nguin | When you are on box A, you have a peer entry for box B... |
| 00:11.06 | saxa | ok |
| 00:11.16 | p3nguin | So you Dial(IAX2/boxB/extension)... |
| 00:11.22 | saxa | so the username is not in the [] ? |
| 00:12.02 | p3nguin | But the peer entry in box B has a user name in its peer entry for box A. |
| 00:12.05 | p3nguin | So that is a mismatch. |
| 00:12.23 | ChrisInSydney | http://www.koalabeer.com/ |
| 00:12.24 | p3nguin | You specify what user name to send for auth with the username parameter. |
| 00:12.44 | saxa | ok I think I got it |
| 00:13.06 | saxa | so basically a was telling to B that he is B |
| 00:13.14 | saxa | inastead of telling he is A |
| 00:13.23 | p3nguin | Yes, and B has an entry for A. |
| 00:13.29 | p3nguin | So it did not match. |
| 00:13.50 | saxa | ok got it, but why then without user names in the book example work ? |
| 00:14.09 | saxa | just because they use the samw pass ? |
| 00:14.13 | p3nguin | Give me a page number or a link to the online page so I can examine it, and I will try to determine. |
| 00:14.14 | saxa | welcome |
| 00:14.17 | saxa | 112 |
| 00:14.20 | saxa | <PROTECTED> |
| 00:14.37 | saxa | the future of telephony |
| 00:15.11 | saxa | and they register one to another by register statement |
| 00:15.18 | saxa | maybe this was in 1.4 |
| 00:15.27 | saxa | but later changed |
| 00:15.43 | p3nguin | This book wasn't written with 1.4 in mind. |
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| 00:17.23 | p3nguin | Okay, on 112, we are Toronto. |
| 00:17.44 | saxa | yes |
| 00:17.59 | saxa | and 113 is osaka |
| 00:18.02 | p3nguin | Osaka has a peer entry for us. |
| 00:18.40 | saxa | where = |
| 00:18.41 | saxa | ? |
| 00:18.42 | p3nguin | We register to the osaka system with the user/pass that it has configured for us, toronto:welcome |
| 00:19.13 | saxa | yes |
| 00:19.28 | p3nguin | The registration should be successful, because that appears to be correct. |
| 00:19.29 | saxa | but both types are friend |
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| 00:19.59 | saxa | i saw one example somewhere where one was friend and the other was peer |
| 00:20.21 | p3nguin | The same is true with the Osaka system registering back to Toronto -- it uses osaka:welcome because that is what toronto has configured for it. |
| 00:20.39 | p3nguin | So registrations should be no problem in this example. |
| 00:21.13 | saxa | yes, .107 is osakas ip |
| 00:21.45 | saxa | but i got before on both sides registered and seed confirmations |
| 00:21.57 | saxa | but no workie :) |
| 00:22.17 | saxa | <PROTECTED> |
| 00:23.00 | saxa | <PROTECTED> |
| 00:23.00 | saxa | [Feb 16 00:35:39] NOTICE[23756]: iax2-provision.c:551 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. |
| 00:23.03 | saxa | <PROTECTED> |
| 00:23.24 | saxa | and why do I get this provisioning notice on one box ? |
| 00:23.38 | p3nguin | I don't see how that example would work. I would expect exactly what happened to you to happen with the example scenario. |
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| 00:24.27 | saxa | ok, thats probably the reason why for me was working when only one box was registered to the other and not both |
| 00:25.02 | p3nguin | I'm not sure why this problem was not addressed in the example. |
| 00:25.05 | p3nguin | Maybe it wasn't tested. |
| 00:25.14 | saxa | could it be |
| 00:25.42 | p3nguin | Maybe it was only an idea, and that issue was never considered. |
| 00:25.52 | p3nguin | I don't know. I didn't write the book. |
| 00:26.06 | saxa | anyway, now its ok, I for sure won't mess with the confs. Don't touch until its working :D |
| 00:26.29 | p3nguin | The included dial plan example looks pretty good to me, though. |
| 00:27.10 | saxa | ok, now I just need to understand why I don't hear anything when i call from my SIP phone to my cell through dahdi |
| 00:27.15 | saxa | its a codec problem |
| 00:27.23 | saxa | as far as i can understand |
| 00:27.34 | p3nguin | Actually, now that I looked again, even the example dial plan is wrong. |
| 00:27.40 | saxa | p3nguin: thats how my dialplan is |
| 00:28.00 | saxa | I have those incoming contexts where i include the phones |
| 00:28.09 | p3nguin | I'll explain why it is also wrong. |
| 00:28.15 | saxa | huh |
| 00:29.08 | p3nguin | In the Toronto system, there is a peer entry for osaka, which has a context of incoming_osaka. In extensions.conf on Toronto, there is no incoming_osaka context. |
| 00:29.23 | p3nguin | But there is, erroneously, a context called toronto_incoming. |
| 00:29.42 | p3nguin | The same is true with Osaka. |
| 00:30.10 | p3nguin | The Osaka system has a peer for toronto with a context of incoming_toronto. |
| 00:30.45 | p3nguin | The Osaka extensions.conf does not have an incoming_toronto context, but does have, incorrectly, an osaka_incoming context. |
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| 00:31.14 | saxa | they are inversed |
| 00:31.32 | p3nguin | Fix up those bits, and the rest is pretty good. |
| 00:31.40 | saxa | you are right |
| 00:31.41 | p3nguin | I like the way the include is used. |
| 00:31.47 | p3nguin | Of course I am. :) |
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| 00:32.13 | saxa | ok on my machines I put it the right way |
| 00:32.18 | p3nguin | Good! |
| 00:32.26 | saxa | at least one thing :D |
| 00:33.28 | p3nguin | If you are looking at the book in print, it might be wise to take a pencil and make some notes on this. |
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| 00:40.30 | saxa | p3nguin: yep, I already corrected that |
| 00:47.27 | saxa | p3nguin: do you have any idea of where do i set this provisioning stuff ? |
| 00:47.42 | p3nguin | What are you trying to provision? |
| 00:47.50 | saxa | I get that error |
| 00:48.08 | saxa | No IAX provisioning configuration found, IAX provisioning disabled. |
| 00:48.21 | saxa | but only on rcitaly side |
| 00:48.37 | saxa | braserv does not issue this NOTICE when I do iax reload |
| 00:48.51 | [TK]D-Fender | saxa: iaxprov.conf |
| 00:48.55 | [TK]D-Fender | ignore it |
| 00:49.27 | saxa | but I do not have that fine on no one of the machines, at least I think so |
| 00:49.32 | saxa | let me check |
| 00:51.11 | saxa | correct no one of the boxes has this |
| 00:51.23 | saxa | but one issues this error and the other one no |
| 00:55.24 | [TK]D-Fender | ignore it |
| 00:56.07 | saxa | i do |
| 00:56.17 | saxa | but its ugly to see it :) |
| 00:56.25 | saxa | i know its a NOTICE only |
| 00:56.50 | [TK]D-Fender | If you don't have the file... put it a bare sample file then |
| 00:58.05 | p3nguin | How often will you really be reloading iax2? |
| 00:58.25 | saxa | from today on I think not much at all |
| 00:58.28 | saxa | :) |
| 01:00.42 | saxa | so now to be honest is it possible that that DAHDI issue I have is some localnet misconfiguration instead of the codec ? |
| 01:01.02 | saxa | why I'm doubting in the codec, because on the rtp set debug on |
| 01:01.14 | saxa | I do not see the Got it reply |
| 01:01.24 | saxa | I see only rtp sent |
| 01:01.41 | pdtpatr1ck | Question - there isn't a way to turn the volume down on MOH right? AFAIK that's not a built in. |
| 01:01.48 | p3nguin | I'd guess it's likely a misconfiguration of some sort. |
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| 01:02.51 | saxa | it is strange because I can call sip to sip , sip to iax |
| 01:02.56 | saxa | without problem |
| 01:03.06 | p3nguin | You can increase the volume for the whole channel, but I don't know about moh specifically. |
| 01:03.10 | saxa | but sip to dahdi I cant hear anything |
| 01:03.18 | p3nguin | s/increase/adjust up or down/ |
| 01:03.44 | p3nguin | What kind of files are you playing in moh that need adjusted? |
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| 01:35.21 | ChrisInSydney | pdtpatr1ck: If you are ripping your own audio from CDs, I have found that you will have to drop the level by 1/2 to 2/3 to avoid clipping on Asterisk |
| 01:35.54 | pdtpatr1ck | using something like sox i take it |
| 01:36.29 | ChrisInSydney | pdtpatr1ck: sox -v .3 loud.wav quiet.wav |
| 01:36.43 | ChrisInSydney | thats what I use |
| 01:36.49 | ChrisInSydney | correct |
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| 02:06.30 | blizzow | Has anyone here experienced a SIP trunk hijacking? |
| 02:07.37 | ChrisInSydney | blizzow: ?? expalin |
| 02:07.51 | WIMPy | probably |
| 02:11.57 | carrar | thats just your boss listening in and taking over your calls |
| 02:24.54 | blizzow | ChrisInSydney: I have a SIP trunk provider and they claim that someone from an IP in egypt registered a PBX using our username and password. |
| 02:25.23 | blizzow | I'm trying to figure out if I should be pissed that my SIP trunk provider hasn't locked down their trunk server to only accept PBX registrations from my IP range. |
| 02:25.35 | ChrisInSydney | That would be the case here |
| 02:26.01 | ChrisInSydney | how much did they rack up ? |
| 02:26.32 | ChrisInSydney | Did they brute force ?? |
| 02:26.34 | blizzow | Just a couple hundred minutes of calls to Africa and the Middle East. |
| 02:26.55 | blizzow | I don't know if they brute forced. I can't even tell how I would figure that out. |
| 02:27.14 | ChrisInSydney | You cant. Its the ITSPs problem |
| 02:27.38 | ChrisInSydney | Do you have a public IP with 5060 open ? |
| 02:27.49 | ChrisInSydney | or are you behind NAT ? |
| 02:28.01 | blizzow | It's behind a firewall. |
| 02:28.14 | ChrisInSydney | no 5060 available from the public IP ? |
| 02:28.23 | blizzow | You can check if you'd like: 74.7.49.236 |
| 02:28.57 | blizzow | I have nmapped the shit out of my public IP and don't see any way to get in there. |
| 02:29.12 | ChrisInSydney | dont have a UDP scanner on this machine. |
| 02:29.28 | ChrisInSydney | K so its their problem. |
| 02:29.58 | blizzow | I figured as much. ChrisInSydney: Thanks. I will be yelling at some people tomorrow I guess. |
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| 02:30.38 | ChrisInSydney | No need to yell. Just be unwavering in your insistance that they cop the costs |
| 02:30.41 | ChrisInSydney | they will |
| 02:31.07 | ChrisInSydney | Their downside is bad publicity for the security on their network. |
| 02:31.51 | ChrisInSydney | You have given them an "Opportunity" to improve their security practices. They shoul dbe thanking you ;-) |
| 02:32.00 | blizzow | Thanks. |
| 02:32.52 | blizzow | Shouldn't they be firewalling to only allow registrations from customer IPs? |
| 02:33.42 | ChrisInSydney | pretty standard practice I would imagine |
| 02:33.59 | blizzow | Well, I'm off to catch dinner, thanks for the answers! |
| 02:34.36 | ChrisInSydney | cheers |
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| 02:39.43 | luizfbmiranda88 | help with DAHDI |
| 02:41.46 | p3nguin | ITSPs do not firewall to allow only customers, because customers come from millions of different IP addresses. |
| 02:51.05 | p3nguin | If you use registrations, they have no way to correctly do that, anyway. If you want them to only allow your IP address, use IP auth instead of registrations. |
| 02:56.57 | ChrisInSydney | Many of our ITSPs here have contry based restrictions on the reg servers |
| 02:57.05 | ChrisInSydney | country |
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| 03:09.37 | p3nguin | That wouldn't necessarily be a bad idea, but it could be a problem maintaining such access lists. |
| 03:16.19 | ChrisInSydney | Then you have an alternative server which is unrestricted but you have to nominate that you will be needing it |
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| 03:41.37 | mcrownover | evening all |
| 03:42.28 | tuxd00d | Evening |
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| 04:44.32 | puzzled | WIMPy: do you by any chance have a zaphfc patch for 2.6.0? the dahdi-zaphfcs repo is using trunk and has oslec mixed in |
| 04:44.44 | puzzled | evening all |
| 04:45.12 | WIMPy | zaphfc is outdated. Use dahdi-hfcs |
| 04:48.37 | puzzled | WIMPy: I though Raoul was justing changing the name. Is he messing with the code too? |
| 04:49.39 | WIMPy | It was updates as well. I'm not sure if it (already?) works with 2.6. |
| 04:50.46 | puzzled | WIMPy: ok thanks |
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| 06:38.11 | ziz212 | Dear All, I am writing a dial plan to receive fax. I am having a issue in there. System() is not working in the dial plan. http://pastebin.com/kDHg6te0 will give the part of the dial plan. If I execute the command directly in the bash it work fine. Pls help me on this. |
| 06:39.04 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:39.45 | ziz212 | out put is http://pastebin.com/drjAMdFh if I execute the dial plan |
| 06:46.52 | kaldemar | ziz212: add full path to send_mail.sh |
| 06:47.55 | kaldemar | what does look strange is that your CLI output does not show asterisk executing the priority with the System app at all. |
| 06:54.34 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:54.36 | v0lZy | hello |
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| 07:18.50 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:18.54 | schmidts | good morning |
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| 07:23.25 | v0lZy | hi |
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| 07:25.33 | ziz212 | @kaldemar thanks I was disconnected due to internet issue |
| 07:25.47 | ziz212 | script is not executing |
| 07:25.56 | ziz212 | through dial plan |
| 07:26.11 | kaldemar | ziz212: did you add full path for it? |
| 07:26.23 | kaldemar | ziz212: is the System app executing in dialplan? |
| 07:26.35 | ziz212 | yes |
| 07:26.49 | ziz212 | it is just a mail sending command |
| 07:26.55 | kaldemar | yes for both questions? |
| 07:27.10 | ziz212 | yes |
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| 07:27.14 | ziz212 | both yes |
| 07:27.26 | kaldemar | does your asterisk user have rights to run the script? |
| 07:27.37 | ziz212 | o m g |
| 07:27.59 | ziz212 | how could i add those permissions to that |
| 07:28.09 | ziz212 | thanks for the point |
| 07:28.25 | kaldemar | depends on what the user you're running asterisk is. |
| 07:29.18 | ziz212 | if not then what will be the option for asterisk application System( |
| 07:29.51 | ziz212 | does it have any special command to get permission like sudo |
| 07:31.06 | kaldemar | you're chasing the wrong end. find out what user asterisk is running as, and then change the permissions in your file system with chmod. |
| 07:31.36 | *** join/#asterisk Henchman21 (~rakata@208.102.127.220) |
| 07:31.55 | Henchman21 | ew man i cant wait to take back control of my phone line |
| 07:32.02 | kaldemar | you really can't do that from asterisk, unless you grant sudo rights for the asterisk user in sudoers and run the system app with sudo. but that's not good practice. |
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| 07:52.59 | ziz212 | @kaldemar if my application need root permission to execute, how could I execute that through system() call in a dial plan ? (my asterisk is run in asterisk user and group permission) |
| 07:57.12 | kaldemar | needing root permissions is kind of nasty, i'd avoid that if possible. if you can't for some reason, use sudoers to grant permissions for the asterisk user for that particular command and run the command in app System with sudo. |
| 07:58.37 | Henchman21 | my momma always said not to use rewt cept to rebewt |
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| 08:45.46 | ziz212 | @kaldemar My case I have added the asterisk to sudos and now it is working. But I feel that system call will not end before going to next like in dial plan? Is it possible? |
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| 08:47.53 | kaldemar | ziz212: the app will wait for send_mail.sh to exit. if that script launches something in the background, app System won't know anything about it. |
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| 09:00.10 | aurs | hello world, does anyone know how to configure a polycom phone so that it is not possible to call it with ip-ip dialing? |
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| 10:36.55 | as001 | hi can I use session_timers instead of session-timers in my sip_conf realtime mysql table ? Will Asterisk find it ? |
| 10:38.23 | as001 | I use 1.8.8.0 asterisk on debian squeeze |
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| 10:42.53 | kaldemar | as001: no. |
| 10:43.35 | as001 | ok thanks |
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| 10:44.03 | kaldemar | which you will see by trying it and command "sip show settings" in CLI. |
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| 11:08.04 | mcrownover | good morning - anyone awake? |
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| 13:52.02 | saxa | ok, just got my new yealink T38 |
| 13:52.07 | saxa | nice thingy |
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| 14:22.13 | ilj | what's :0:11 called in: exten => _.,1,Dial(IAX2/${EXTEN:0:11}/${EXTEN:11}) ? |
| 14:22.23 | ilj | I know what it does not sure what it's called properly |
| 14:22.58 | [TK]D-Fender | ilj, a sub-string reference |
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| 14:33.02 | ilj | thanks [TK]D-Fender |
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| 15:01.24 | Srini | Hi room |
| 15:02.05 | Srini | How do I transfer a call from pstn/pri to a sip extension... need help in writing a dialplan |
| 15:02.23 | Srini | Rather I want to transfer the call to a sip extension which is free |
| 15:03.49 | edge | Srini: since its pretty quiet i'll add 2 cents. I would assume you would use the dial method |
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| 15:04.03 | edge | Srini: from the PRI context |
| 15:04.16 | Srini | edge, I use a context from-pstn |
| 15:04.34 | edge | Srini: so in that context specificy where you want to dial to |
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| 15:05.19 | Srini | edge, is it something like: Dial(SIP/${EXTEN}) |
| 15:05.29 | edge | Srini: yupp |
| 15:05.33 | edge | Srini: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 15:06.18 | edge | Srini: assuming you've provided the ${EXTEN} side |
| 15:06.45 | Srini | edge, yes, I have the extension with that number |
| 15:07.18 | edge | Srini: if a call comes in from lets say a POTS line, it goes into the context you specifiy in the DAHDI configuration. It then starts at extension S, and then routes the call based on whatever you need |
| 15:09.03 | Srini | edge, if I am recieving call on 88888 (pri board number) and want to recieve the call on a sip extension 88888 how do I write the dial plan (I am actually confused!) |
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| 15:10.06 | edge | Srini: I'm not 100% sure, but I'm sure that number is a variable in Asterisk some where |
| 15:10.15 | edge | Srini: let me see if i can find a document that describes it. |
| 15:10.53 | Srini | Also I have another thing to ask, what if I want to recieve those calls on some free sip extension between 88888 and 88896 |
| 15:11.11 | edge | Srini: if each PRI board number had its own conext you could just dial directly and not use variables, but thats dirty and not scalable |
| 15:11.26 | Srini | edge, very true! |
| 15:12.07 | edge | Srini: the Dial function is very smart. it can see that an extension is busy and jump to the next in the dial plan. If you have a check down list of extensions just put them in the dial block |
| 15:12.59 | Srini | edge, what I understand by your words is, to put all those extensions into a group, is that correct? |
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| 15:14.29 | edge | Srini: the dial plan is very linear. So lets say we figured out how to get the PRI # and it was 88888 we can start that extension with dialing say sip/${BOB_ACC} , if that doesn't work or fails dial goes to the next command in that extension |
| 15:15.06 | Srini | oh! correct |
| 15:15.33 | [TK]D-Fender | Srini, exten => 88888,1,Dial(SIP/88888) |
| 15:15.35 | [TK]D-Fender | ^^^66 |
| 15:16.21 | edge | [TK]D-Fender: Srini , that is a lot easier to do it that way. I didn't know how Asterisk would handle the board number. seems like it carries it as the extension # |
| 15:16.52 | [TK]D-Fender | I not familiar with "board number" being a valid term for anything... |
| 15:17.34 | edge | Srini: then all you have to do is do below the first line [TK]D-Fender send, is same =>n,Dial(SIP/${NEXTGUY}) |
| 15:18.02 | [TK]D-Fender | edge, I would refrain from making variable references to things like that... |
| 15:18.14 | edge | [TK]D-Fender: of course |
| 15:18.20 | [TK]D-Fender | Srini, And keep things in mind like dial timeouts, etc |
| 15:18.35 | Srini | [TK]D-Fender, sure.. |
| 15:19.38 | edge | Srini: the timeout is after the extension. Dial(SIP/88888,10) would ring for 10 |
| 15:20.00 | edge | Srini: without that Dial wouldn't give up if somebody wasn't at their desk |
| 15:24.20 | Srini | edge, when we say ${BOB_ACC} how am I going to specify and where am I going to specify? in the extensions.conf itself? |
| 15:25.01 | edge | Srini: like [TK]D-Fender was saying, it isn't good practice to make variables like that. I was just not wanting to put random SIP numbers in there |
| 15:25.31 | Srini | Then perhaps it is going to be hardcoded for all the extensions prevailing? |
| 15:25.37 | edge | Srini: how i do it is I use MAC addresses |
| 15:26.12 | edge | So if i wanted to call a device on my network its Dial(SIP/708457845142) |
| 15:26.29 | Srini | edge, yes it is working well that way |
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| 15:26.37 | edge | Srini: and i create at the top (which prob isn't good practice) users. |
| 15:27.00 | edge | Srini: at the top of my extension.conf is TIM = SIP/745896587456 |
| 15:27.07 | edge | and when i want to call Tim in my dial plan its just |
| 15:27.29 | edge | dial(${TIM}) |
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| 15:29.10 | Srini | edge, Yup, I could understand that- So If the PRI has 31 dialable extensions, we would need to write dialplan for each of them seperately and ending transfering call landing to corresponding extension only |
| 15:29.14 | edge | Srini: again i'm sure that that isn't best practice but it works great for me. |
| 15:29.30 | azv4 | Anyone know what Panasonic calls ring groups for their Digital Hybrid generation of hardware? |
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| 15:30.21 | edge | Srini: I would read up about Macros, as they offer a bit of scripting that might come in handy for you. |
| 15:31.26 | Srini | edge, |
| 15:31.27 | Srini | how the dial plan would look like when I am dialling a POTS number from a SIP extension using dahdi trunk (through PRI) |
| 15:32.03 | edge | Srini: well the SIP device is configured with a context as well |
| 15:32.25 | edge | Srini: So all extension (even outside phone numbers) land in that context |
| 15:33.05 | Srini | ok I understand.... ! |
| 15:33.20 | Srini | edge, That makes a lot of things clear for me! |
| 15:33.29 | edge | Srini: the dial plan then has a wild card to pickup say 7 digit numbers which are 4 longer than a 3 digit extension. and then it performs a Dial to the channel you wish to send it to |
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| 15:34.53 | Srini | So it is like n,Dial(${TRUNKX}/${EXTEN:1} |
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| 15:35.55 | [TK]D-Fender | Srini, exten => _NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN},60) |
| 15:36.26 | [TK]D-Fender | Srini, Sample for north american formatted number dialing out DAHDI group 1. |
| 15:36.28 | edge | Srini: the :1 performs a cut to the extension which would result in it only being a 6 digit number. [TK]D-Fender has the right plan for you |
| 15:36.54 | [TK]D-Fender | Srini, Pay attention to the patterns you need |
| 15:37.40 | Srini | [TK]D-Fender, That definitely helped me! |
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| 15:39.06 | edge | Srini: A lot of what we've shown you is in a very helpful guide http://www.asteriskdocs.org/ . I read it cover to cover and I go back to it a lot. It is a very good tool to have and i strong suggest you read it. But we are glad to help as well. |
| 15:39.08 | [TK]D-Fender | <Srini> So it is like n,Dial(${TRUNKX}/${EXTEN:1} <--- this one you showed assumes there is a priority 1 above it somewhere, doesn't show the pattern that was dialed, you are stripping off a leading digit from whatever that number dialed was before passing it on, and you were passing it on to some unknown tech because even that part is a variable/constant reference |
| 15:39.14 | [TK]D-Fender | ~book |
| 15:39.14 | 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 |
| 15:39.15 | [TK]D-Fender | ^^^ |
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| 15:41.58 | edge | [TK]D-Fender: Should of known the book would be hot keyed and not to google it |
| 15:43.14 | edge | Can anybody see a reason this Cisco SPA5xx dial plan for the phone would not work: |
| 15:43.15 | *** join/#asterisk techwerkz (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 15:43.17 | edge | L:15,S:2,(p5 | *9xxx | *9x | [1-4]xx | 6xx | 7xx | xxx xxxx | 608 xxx xxxx | <:1>xxx xxx xxxx | x xxx xxx xxxx ) |
| 15:43.35 | edge | I can't dial 601 - 699 or any 700 - 799. it just says address incomplete |
| 15:44.00 | edge | document says white space is ignored so i formatted it that way for easy reading |
| 15:44.11 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 15:45.54 | [TK]D-Fender | edge, NEVER put in wuitspace, and where do you see that error? |
| 15:46.09 | [TK]D-Fender | whitespace* |
| 15:46.10 | edge | [TK]D-Fender: i see the error on the Cisco SPA502G phone itself |
| 15:46.20 | edge | [TK]D-Fender: It says addressincomplete |
| 15:46.34 | edge | [TK]D-Fender: and the asterisk CLI never sees anything from the device |
| 15:47.01 | [TK]D-Fender | L:15,S:2,(p5 <- don't recognize what this is supposed to be offhand |
| 15:48.05 | edge | [TK]D-Fender: the L:15 is long timer for incomplete dialing |
| 15:48.28 | edge | the S:2 is for short timmer the ammount of time it waits once something matches a pattern before it executes |
| 15:48.54 | edge | the p5 is for phone being off the hook for 5 seconds without something being pressed that it errors out |
| 15:49.20 | techwerkz | edge: On my SPA devices I don't have whitespace, and I have it set as 6xx. |
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| 15:49.54 | Srini | In the asterisk CLI how do I list all availabe TRUNKS? |
| 15:50.21 | Srini | or should I be using some dahdi command? |
| 15:51.01 | [TK]D-Fender | Srini, "trunk" is not a specific thing |
| 15:51.34 | [TK]D-Fender | Srini, and "available" is somewhat vague |
| 15:52.50 | edge | Srini: dahdi show channels |
| 15:53.02 | edge | Srini: shows me all my channels and their status |
| 15:53.07 | Srini | [TK]D-Fender, how do I know what DAHDI/gX I have.. then how do I get to know that |
| 15:53.53 | edge | techwerkz: [TK]D-Fender i removed all the white space and i get the same result of 601 being an incomplete address |
| 15:54.19 | [TK]D-Fender | Srini, its your config file. Go look what you put in it |
| 15:54.26 | edge | I'm going to remove all the time formatting to see if that is the issue |
| 15:55.26 | techwerkz | Can anyone further explain what the srvlookup=yes in iax.conf actually does? Currently I have a peer definition setup on two locations with their type as friend and setup as a trunk for inbound/outbound calling through each peer. However when I use my DNS srv record and force an outage, the failover never happens. It still has the old IP address cached for the peer. Should I be using register instead? This does work as expected with |
| 15:55.30 | edge | even without the formatting it doesn't work |
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| 16:22.32 | RZero | Hi all, I am testing sending faxes via ami using Telnet so I can script it once its working, Im trying to set the fax headers, here is what I have so far http://pastebin.com/4M8pf9xc if I remove the two set commands it sends the faxes fine, how do I set the headers for faxes in AMI as these are not working |
| 16:22.33 | Nugget | telnet is eeeeeeevil! |
| 16:27.15 | [TK]D-Fender | RZero, there is no such thing as "command" for Originate. |
| 16:27.24 | RZero | oh ok |
| 16:27.27 | RZero | :) |
| 16:27.39 | RZero | what do I use instead ? |
| 16:28.56 | [TK]D-Fender | RZero, You stop using Application, and you dump it into the dialplan instead. |
| 16:29.32 | RZero | Thats what I normally do, but this a system that talks to asterisk via AMI |
| 16:29.42 | RZero | this for* |
| 16:29.44 | [TK]D-Fender | RZero, You stop using Application, and you dump it into the dialplan instead. <------------- |
| 16:29.49 | Qwell | RZero: see 'manager show command originate' |
| 16:29.50 | Qwell | Variable: |
| 16:30.21 | [TK]D-Fender | RZero, https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Originate |
| 16:32.07 | edge | techwerkz: do you use time out settings? |
| 16:33.01 | RZero | so it looks like this Variable: FAXOPT(headerinfo)=Faxy fax team ? |
| 16:33.38 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:34.41 | Srini | [TK]D-Fender, so trunk is whatever specified in the system.conf? |
| 16:35.30 | [TK]D-Fender | Srini, huh? |
| 16:36.12 | Srini | [TK]D-Fender, sorry... could not understand thus had to ask... |
| 16:36.20 | [TK]D-Fender | Srini, I highly recommend you abandon all use of the word "trunk". It is vague at best......... |
| 16:36.30 | Srini | [TK]D-Fender, ok |
| 16:43.00 | anonymouz666 | [TK]D-Fender: I setup an IAX2 trunk today |
| 16:43.01 | anonymouz666 | :P |
| 16:43.41 | [TK]D-Fender | anonymouz666, SHUP YUO |
| 16:44.13 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 16:44.19 | Donper | Hi all, i've some problems since i update from 1.4 to 1.8.9.2, sometimes when users make tranfers, the call is directly hangup, i can see in the cli : chan_sip.c: Retransmission timeout reached on transmission 3633c44c-c0a80101-0-34@192.168.X.X for seqno 104 (Critical Request) Packet timed out after 6403ms with no response |
| 16:44.19 | Donper | [Feb 16 11:28:23] WARNING[16315] chan_sip.c: Hanging up call 3633c44c-c0a80101-0-34@192.168.X.X - no reply to our critical packet |
| 16:44.26 | p3nguin | I stuffed something in the trunk. |
| 16:44.34 | Donper | if someone have any answer... |
| 16:44.35 | Donper | ;) |
| 16:45.18 | p3nguin | donper: "sip set debug on" AND "core set verbose 3" |
| 16:45.24 | p3nguin | donper: Then do it again. |
| 16:45.30 | Donper | i did it |
| 16:45.31 | p3nguin | Make it fail. |
| 16:45.39 | p3nguin | Then pastebin the entire output. |
| 16:45.43 | Donper | ok |
| 16:46.07 | *** join/#asterisk zkn (~zkn@82.131.68.58.cable.starman.ee) |
| 16:46.31 | Donper | http://pastebin.com/eb18XT7j |
| 16:47.32 | [TK]D-Fender | p3nguin, proabably a lot of junk in your trunk |
| 16:49.02 | p3nguin | Yep: a box of Cat 5e, a fish tape, a speaker box and amp, misc. tools, a 4-port hub, and maybe some other stuff that I haven't seen for a while. |
| 16:49.46 | p3nguin | donper: Which phone is transferring, and which extension is it transferring to? |
| 16:49.52 | Srini | my extension.conf is here - I am able to recieve call from the PRI but still not able to make outgoing calls - it says : "Call from '888888' to extension '999999' rejected because extension not found." |
| 16:50.10 | Srini | http://pastie.org/3395570 |
| 16:50.12 | p3nguin | '999999' does not exist in the context where your call is. |
| 16:50.36 | p3nguin | You have two instances of extension 888888. |
| 16:51.21 | Srini | p3nguin, the incoming part has issues now - the outbound calls are not happening |
| 16:51.36 | Donper | p3nguin : the extension 5842 to 5827 |
| 16:54.25 | [TK]D-Fender | Srini, indeed it does not exist. You have 2 sets of 88888 <- |
| 16:54.45 | p3nguin | donper: Why are you Answer()ing before Dial()ing? |
| 16:54.48 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
| 16:54.55 | [TK]D-Fender | Srini, You have no 99999 in there which is what it says it is looking for. |
| 16:54.56 | p3nguin | That's not how it should be configured. |
| 16:55.01 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:55.42 | Donper | the extension 5842 call 5827 to annonce which one he have to transfer, and next he transfer... |
| 16:56.01 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 16:56.36 | Donper | it is not very clear and my english is not very well ... |
| 16:59.03 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-104-242-22.ph.ph.cox.net) |
| 16:59.07 | Srini | Can there be some example for outbound dialplan? (sorry if I am really irritating the room here~!) |
| 16:59.35 | p3nguin | donper: A phone by the name of 5842 is calling extension 5827. |
| 17:00.12 | p3nguin | donper: Extension 5827 just _coincidentally_ is dialing a phone by the name of 5827. |
| 17:00.23 | p3nguin | But that wasn't my question. |
| 17:00.38 | p3nguin | My question was: why is there an Answer() before the Dial()? |
| 17:00.39 | Donper | first, the phone 5828 is calling 5842, next 5842 is calling 5827 to transfer the first call |
| 17:01.23 | Srini | [TK]D-Fender, If I am calling some number from my sip phone using the dahdi ... _X. would help? or am I completely missing the actual poing somewhere! |
| 17:01.24 | p3nguin | It looks like 5842 reaches 5827, but then there is a re-invite. |
| 17:01.47 | p3nguin | Set directmedia=no for both phones in sip.conf. |
| 17:02.22 | Donper | the call is droping, when 5842 press the transfer button |
| 17:02.27 | p3nguin | Set directmedia=no for both phones in sip.conf. |
| 17:02.49 | [TK]D-Fender | Srini, Yes you are completely missing the point. |
| 17:02.50 | Donper | it already set like that |
| 17:02.57 | ChannelZ | Srini: That would catch everytthing... which may or may not be what you want to do. Generally not. |
| 17:03.17 | [TK]D-Fender | Srini, It is looking for 99999 in your dialplan. Lokk at what you jsut gave me. You don't have a match for 99999 in there. It should fail. |
| 17:03.18 | p3nguin | _X. is much better than _. would be. |
| 17:03.31 | Donper | all my sip acount have directmedia=no |
| 17:03.47 | [TK]D-Fender | Srini, You have 2 sections, BOTH with 88888. You probably forgot to pay attention to the number you were typing |
| 17:04.00 | Srini | But in my case 99999 is not an extension, it is a pstn number elsewhere, probably a mobile number |
| 17:04.08 | p3nguin | donper: Do you have nat=yes set in the [general] section? |
| 17:04.12 | [TK]D-Fender | Srini, it is the number that comes in <- |
| 17:04.29 | Donper | nat=no in general and on my account |
| 17:04.33 | Srini | 888888 is the number assigned to my extension |
| 17:04.38 | [TK]D-Fender | Srini, the outside is calling that numebr. what you you do with it it is something else |
| 17:04.41 | [TK]D-Fender | Srini, NO. |
| 17:04.42 | p3nguin | donper: Change it. You are behind NAT. |
| 17:04.51 | [TK]D-Fender | Srini, the call from the outside TARGETS a number |
| 17:05.09 | p3nguin | donper: And remove nat= from all peer entries. |
| 17:05.23 | p3nguin | donper: You will have only one nat= line in the entire file. |
| 17:05.26 | [TK]D-Fender | Srini, the outside says "I WANT 99999". You did not make something that will match their request |
| 17:05.50 | p3nguin | donper: Also be sure to set the correct localnet value in the general section. |
| 17:06.05 | Donper | yes but, i have only 2 Cisco phone, and if i don't set nat=no for them, they aren't able to register on the server |
| 17:06.20 | Srini | [TK]D-Fender, So, if I am dialling out I dont need the exten included? just dial(DAHDI/g1) ? |
| 17:06.40 | [TK]D-Fender | Srini, Strop. You re going in circles. Youa tre not loking at what is happening |
| 17:06.59 | Srini | [TK]D-Fender, :( really unable to follow- I am sorry |
| 17:07.01 | Donper | but, i've no localnet declare in general |
| 17:07.05 | p3nguin | If the call is looking for extension 999999, it better be there. |
| 17:07.22 | [TK]D-Fender | Srini, LOOK at your dialplan. You ahve 2 SECTIONS that both refer to 88888 in there. Look at it right now. |
| 17:07.24 | p3nguin | In your case, it is NOT. |
| 17:07.32 | [TK]D-Fender | Srini, why are there 2 sections like this? |
| 17:07.40 | Srini | ok! |
| 17:07.59 | p3nguin | I'm surprised asterisk did not bitch about having a duplicate extension 888888. |
| 17:08.51 | [TK]D-Fender | p3nguin, No, it'll happily overwrite |
| 17:09.02 | Srini | I really thought the last two lines are for the oubound calling! |
| 17:09.21 | Srini | ok sorry |
| 17:09.46 | Srini | Dial(SIP/888888,10) ? |
| 17:10.37 | p3nguin | [Feb 16 11:10:29] WARNING[22256]: pbx.c:8134 add_priority: Unable to register extension '5623', priority 1 in 'outgoing_calls', already in use |
| 17:10.38 | *** join/#asterisk jasonwert (~w3rt@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 17:10.47 | p3nguin | It does bitch about a duplicate! Just as I expected. |
| 17:10.53 | Srini | [TK]D-Fender, I missing the point in last two lines? |
| 17:11.12 | p3nguin | srini: Do you have a SIP phone by the name of 888888? |
| 17:11.19 | Srini | Yes |
| 17:11.26 | RZero | [TK]D-Fender & Qwell thanks for your help, got it working |
| 17:11.54 | Srini | pstn calls to 888888 are landing onto sip exten 888888 |
| 17:12.12 | p3nguin | There is no "sip exten 888888." |
| 17:12.30 | Srini | I am able to recieve calls on it! |
| 17:12.31 | [TK]D-Fender | <Srini> my extension.conf is here - I am able to recieve call from the PRI but still not able to make outgoing calls - it says : "Call from '888888' to extension '999999' rejected because extension not found." |
| 17:12.32 | p3nguin | There could be extension 888888. There could be sip device 888888. |
| 17:12.37 | [TK]D-Fender | Srini, it is looking for 99999 |
| 17:13.00 | Srini | 99999 happens to be an outsite number - somewhere in the public telephone network |
| 17:13.01 | [TK]D-Fender | Srini, it is NOT looking for 88888 |
| 17:13.21 | [TK]D-Fender | Srini, Your system ... is getting a call... and that call is looking for 99999 |
| 17:13.38 | Srini | ah! |
| 17:13.39 | [TK]D-Fender | Srini, Repeat this over and over and over and over and over and over in your head about 100 million more times. |
| 17:13.45 | p3nguin | A device by the name of '888888' is trying to call phone number '999999' which DOES NOT EXIST. |
| 17:14.23 | p3nguin | Extension '999999' does not exist. |
| 17:14.31 | Srini | p3nguin, but I was expecting the call from SIP/888888 to go to dahdi and pass the connection to public telephone number 999999! |
| 17:14.47 | p3nguin | What number did you dial on the 888888 phone? |
| 17:14.50 | Srini | in reality 999999 could be any valid public telephone number |
| 17:14.55 | p3nguin | Did you dial 999999 on the keypad? |
| 17:14.59 | Srini | a 10 digit number |
| 17:15.04 | Srini | yes |
| 17:15.07 | Srini | p3nguin, yes |
| 17:15.12 | p3nguin | 999999 does not exist in the dial plan. |
| 17:15.23 | [TK]D-Fender | Srini, that isn't a call from SIP/88888. |
| 17:15.29 | [TK]D-Fender | Srini, that is a call from DAHDI |
| 17:15.37 | [TK]D-Fender | Srini, came from DAHDI |
| 17:15.42 | p3nguin | <PROTECTED> |
| 17:15.44 | [TK]D-Fender | Srini, did not come from a SIP device |
| 17:15.56 | Srini | oh! |
| 17:16.02 | p3nguin | <PROTECTED> |
| 17:16.47 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:17.21 | p3nguin | "Call from '888888'" does not necessarily mean SIP/888888. The message does not indicate a channel tech. |
| 17:17.51 | p3nguin | This is one reason not to use SIP device names the same as the extensions you are calling. |
| 17:18.11 | p3nguin | Too much confusion for people who don't know what they are doing. |
| 17:18.16 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 17:19.02 | *** join/#asterisk darkfrog (~mhicks@ip68-97-6-203.ok.ok.cox.net) |
| 17:19.16 | darkfrog | Is there a PBX in a Flash IRC channel? |
| 17:19.36 | Srini | Hmmmm ! |
| 17:19.38 | p3nguin | PiaF sucks so much, I doubt it. |
| 17:19.41 | [TK]D-Fender | darkNo, but it uses vanilla FreePBX so : #freebpx |
| 17:19.58 | edge | When my phone rejects a number that i'm dialing i get this in the Asterisk CLI, "Using SIP RTP CoS mark 5" What does that mean? |
| 17:20.01 | [TK]D-Fender | darkfrog, No, but it uses vanilla FreePBX so : #freebpx |
| 17:20.20 | [TK]D-Fender | edge, Means nothing really. Enable SIP DEBUG and look at the actual attempt |
| 17:20.47 | darkfrog | [TK]D-Fender: thanks |
| 17:20.55 | edge | [TK]D-Fender: on the CLI or the phone? |
| 17:22.16 | *** join/#asterisk Russ (~russ@conference/linaro-connect/x-shdusvepvbehtvps) |
| 17:23.13 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 17:24.25 | *** join/#asterisk Katty (~Katty@64.82.199.210) |
| 17:24.30 | Katty | ohai |
| 17:24.42 | [TK]D-Fender | edge, * CLI |
| 17:24.55 | [TK]D-Fender | edge, if you want to see why * is rejecting it, look at * |
| 17:25.37 | Katty | how do i tell grep to spit out every line in a conf file that doesn't have ; at the beginning of it |
| 17:25.51 | Qwell | grep -v "^;" |
| 17:26.12 | Katty | ty dear. |
| 17:37.37 | azv4 | any TD-500 pros in the hosue? |
| 17:37.41 | azv4 | *house even |
| 17:38.34 | [TK]D-Fender | azv4, May products have that number. Care to be more specific? |
| 17:39.25 | azv4 | Panasonic TD-500, sry |
| 17:40.51 | [TK]D-Fender | azv4, Now is just sounds like you walking into McDonalds and asked for a Whopper :p |
| 17:41.09 | [TK]D-Fender | azv4, So ... what about it? |
| 17:42.00 | azv4 | ok, a ring group exists on our system, I am trying to add another extension to it, but I can't seem to find the list of extensions already associated with the group so I can add a new one, was hoping someone might remember such things |
| 17:42.44 | [TK]D-Fender | azv4, yup.. really not the place for it... |
| 17:42.57 | [TK]D-Fender | azv4, We usually try to 'sve" people using things like yours |
| 17:43.01 | [TK]D-Fender | save* |
| 17:44.06 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:44.30 | azv4 | how did I know that was going to be your reply |
| 17:44.34 | azv4 | I would love to invest in a new system |
| 17:44.44 | azv4 | and I idle here to help me plan for that when I finally get the budget |
| 17:44.50 | azv4 | in the meantime I am left for dead |
| 17:46.17 | [TK]D-Fender | azv4, Now you can relate to your PBX at least ;) |
| 17:48.25 | azv4 | yes indeed |
| 17:48.32 | azv4 | I usually do find a couple of Panasonic pros in here though |
| 17:59.32 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 18:04.52 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
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| 18:06.06 | *** part/#asterisk dxd828 (~dxd828@88-109-121-130.dynamic.dsl.as9105.com) |
| 18:09.27 | *** part/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 18:11.19 | *** join/#asterisk stev3n (~2@dynamicip-94-181-247-126.pppoe.kirov.ertelecom.ru) |
| 18:14.34 | *** join/#asterisk johnwigley (bc4027e7@gateway/web/freenode/ip.188.64.39.231) |
| 18:18.45 | johnwigley | Hi, I'm having a weird problem with Asterisk not appearing to correctly load settings from it's configuration files when it starts up. The specific issue I have is that the ACKCALL option from AGENTS.CONF is not being honoured when it is specifically disabled. When asterisk starts up, it is acting as though this option is enabled. If I change no settings at all, but simply issue a reload then most of the time it seems to apply t |
| 18:20.21 | johnwigley | I thought at first this was a race condition because the agents.conf file is generated by a #exec from a database table, but I tried manually entering the same config into the file, and tried again with exactly the same results. Further testing shows that Asterisk has to be running around 20 seconds before a reload will get it to load the agents.conf file correctly. Any reloads before then, and Asterisk still seems to ignore set |
| 18:21.24 | mjordan | johnwigley: are you issuing a reload right when Asterisk starts up? |
| 18:21.35 | johnwigley | I presumed that this was a race condition upon start with some dependent module not being available, but apparently there is a module dependency load order specified by each module so that probably isn't the cause. I suspect though haven't proved it that other options are also not being correctly loaded |
| 18:21.51 | johnwigley | If I issue a reload just after startup it still doesn't work |
| 18:22.02 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 18:22.06 | johnwigley | It needs to wait about 20 seconds and then a reload works *most* of the time. |
| 18:22.38 | mjordan | a reload will only reload a configuration file if something in that configuration file was changed |
| 18:22.49 | mjordan | so that's the first thing to check. |
| 18:23.17 | mjordan | before issuing a reload to Asterisk, you should make sure that the "fully booted" event has occurred. |
| 18:23.29 | johnwigley | Sorry this is Asterisk 1.8.8.2+pf.xivo.1.2.1 if that helps |
| 18:23.38 | Qwell | wtf version is that? |
| 18:24.13 | mjordan | I know only half of that version, what is the pf.blah? |
| 18:24.23 | johnwigley | I'm not wanting to issue a reload, I've just found that if I happen to wait 20+ seconds after it starts up then it seems to load the agents.conf file correctly - I'm not changing anything in any of the conf files, just trying to get it to load them. |
| 18:24.39 | johnwigley | It's the asterisk which comes in the Xivo PBX. |
| 18:25.00 | mjordan | so, I can't answer for anything the Xivo PBX might have done. |
| 18:25.11 | johnwigley | https://wiki.xivo.fr/index.php/Accueil for reference |
| 18:25.37 | johnwigley | As far as I'm aware it's a fairly standard and current asterisk build that comes with their embedded linux distro |
| 18:25.54 | mjordan | I'm not going to look at that reference :-) However, a module in Asterisk will load its configuration file when the module is loaded by Asterisk. During a reload, the module will only re-read the configuration file if it has been modified since the last time the configuration file was read. |
| 18:26.27 | johnwigley | ok, so if the conf file has not been changed from asterisk start to when issuing the reload, you're saying it will NOT reload it? |
| 18:26.35 | mjordan | yup. |
| 18:26.41 | mjordan | Nothing has changed, so there's nothing to reload. |
| 18:27.24 | johnwigley | ok, so then something else is going on, issuing the reload command fixes the issue 95% of the time, I've tested it time after time. |
| 18:28.02 | johnwigley | and the critical time is around 20 seconds, if I issue a reload 10s after Asterisk starts then it still won't honour the ACKCALL=no setting. |
| 18:28.58 | mjordan | I just said that if you aren't waiting for a "fully booted" back before you issue your reload, there's no guarantee that its going to do what you want. Asterisk isn't fully loaded yet. |
| 18:29.19 | johnwigley | Though I'm definitely not changing anything in any of the conf files, maybe it's because some of the other conf files contain #exec include statements, so presumably in that case Asterisk will always reload them as it wouldn't be able to tell if the #exec included bits had changed? |
| 18:30.43 | johnwigley | Surely there's some form of dependency/module load checking before it tries to apply it's configuration, so it shouldn't be the case that it needs to be fully loaded before it attempts to read and apply agents.conf? |
| 18:32.26 | mjordan | no, that's not the case. As I said, when chan_agent is first loaded it will read its configuration file. Since the module hasn't been loaded yet, it will read and parse the configuration file, always. When its reloaded, it checks to see if something has changed. |
| 18:32.49 | mjordan | The module load order is a completely separate matter. |
| 18:33.36 | mjordan | A module can't load its configuration if it isn't loaded into memory, and the module load order determines when that occurs. A reload does not remove a module from memory, it simply instructs it to reload its appropriate settings if it detects that it needs to. |
| 18:34.06 | johnwigley | So since nothing has changed in the conf files when I issue a reload, you're saying it won't do anything at all BUT that's not what my experimentation shows. Maybe it doesn't re-read the conf file, but it certainly does do *something* which means that it does apply the settings. I'm just trying to understand what it could be doing that's causing it to then start working. |
| 18:34.15 | mjordan | k. |
| 18:36.33 | mjordan | I'm not going to repeat myself ad nauseum. You can believe what you want, I'm telling you what actually happens in stock Asterisk. If you feel you have a legitimate bug, feel free to open an issue in the issue tracker. Include your configuration files, as well as DEBUG logs illustrating the behavior your seeing. Since you're issuing reloads in a somewhat odd manner (immediately upon Asterisk starting is odd), you might want to also att |
| 18:37.59 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-183.mobile.uci.edu) |
| 18:39.54 | johnwigley | I believe exactlt what you're saying, I know hardly anything about Asterisk - which is why I'm asking for help. What I want to try and do is be helpful and actually be able to describe clearly what is going on, and make sure it is a bug before I report it. Because I don't understand what's going that's why I want to try and precisely identify what's happening and why a reload fixes it. |
| 18:40.27 | johnwigley | Could you suggest how I might go about identifying what exactly is happening when I issue a reload command? |
| 18:42.19 | jrose_atDigium | Well, some functions display verbose output for reloading... |
| 18:42.36 | jrose_atDigium | But the only way to know exactly what's happening is to follow the code from the reload function. |
| 18:43.28 | TSM | has anyone had issues with the latest polycom bootrom accessing ftp server for config? |
| 18:44.32 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 18:44.33 | johnwigley | Thanks, I think to be honest I'm not going to be able to usefully follow the source code for the reload function, I only just about understand dialplans at the moment :) |
| 18:49.59 | KavanS | can someone point me to an example of dialplan for a door intercom? |
| 18:50.00 | johnwigley | Is the decision to reload conf files made on a per module basis, ie if say extensions.conf has changed but agents.conf has not, will the agents module reload the agents.conf file? I'm wondering if that is the reason why the reload does in fact cause it to reload and apply the conf even though agents.conf hasn't changed. |
| 18:54.34 | mjordan | johnwigley: chan_agent, in particular, will reload if either agent.conf or users.conf has been updated, as it uses both configuration files. Changes in an unrelated configuration file will not affect its reload. |
| 18:58.24 | *** join/#asterisk fhmiv (~fhmiv@24.214.235.162) |
| 18:58.36 | *** part/#asterisk fhmiv (~fhmiv@24.214.235.162) |
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| 19:05.13 | johnwigley | Ok, thanks. I've just had a look through chan_agent.c and exactly as you say it checks for file change when read_agent_config is called with the reload flag set. I wondered if it still did an internal update or something similar, but it appears to do absolutely nothing when reload is called and the file hasn't changed. Since I don't have a users.conf and I have a static agents.conf which hasn't changed at all since Asterisk load |
| 19:06.21 | johnwigley | seems to get it to honour ACKCALL. I've also just noticed that ACKCALL is set to false by the read_agent_config function by default anyway - so even if it couldn't read and apply the config file correctly when it started ACKCALL should be disabled by default NOT enabled is it's behaving. |
| 19:09.14 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 19:09.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:13.11 | johnwigley | Is there a way that I can see from the CLI what the default ACKCALL settings are for each Agent before I issue a reload - to confirm that it isn't reading the file correctly when first loaded? As this could actually be a bug in the handling of ACKCALL elsewhere rather than my first guess that it was just not reading the config file correctly. |
| 19:15.36 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 19:15.37 | dijib | 1 |
| 19:15.40 | dijib | hello all |
| 19:15.57 | dijib | what can you tell me about a Sangoma A-200R card? |
| 19:16.30 | dijib | i just found a intel core duo system in the dumpster with one of these cards in PCIe, plugged it in and everythings working |
| 19:21.32 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 19:23.12 | [TK]D-Fender | digiGood card |
| 19:23.18 | [TK]D-Fender | dijib,^ |
| 19:25.18 | saxa | hi anybody willing to look whats wrong here to have no audio at all between rcoffice and casasip ? |
| 19:25.21 | saxa | http://pastebin.com/2SSKD40x |
| 19:26.28 | dijib | good card yeh? |
| 19:27.03 | dijib | its FXO FXS interchangable? |
| 19:27.11 | *** join/#asterisk fofware (~fabian@host63.190-31-63.telecom.net.ar) |
| 19:27.22 | thehar | wiggles |
| 19:28.07 | malcolmd | dijib: nope, fxs and fxo are electrically different interfaces |
| 19:29.02 | talntid | as well as tasers. |
| 19:29.03 | [TK]D-Fender | dijib, Yes it is a modular card |
| 19:29.41 | [TK]D-Fender | dijib, You can't change the modules purpose, but you can swap thm out for the kind you need |
| 19:29.46 | *** part/#asterisk stev3n (~2@dynamicip-94-181-247-126.pppoe.kirov.ertelecom.ru) |
| 19:38.09 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 19:38.27 | Dovid | is there any way of setting SIP options (other than the codec) in the dialplan? |
| 19:38.50 | Dovid | like nat=yes and directrtp=no |
| 19:41.10 | saxa | is impressed on how well is done the web interface of the Yealink SIP-T32G |
| 19:42.23 | saxa | Dovid: yes |
| 19:42.37 | saxa | Dovid: not in the dialplan |
| 19:42.44 | saxa | Dovid: in sip.conf |
| 19:43.52 | dijib | so [TK]D-Fender it should say on it if its an FXO or FXS? also i love dumpster diving. found a CD1.6ghz 800fsb, 512@667mhz, dual 80gb sata's 100% working when i brought it home and plugged it in from the dump |
| 19:43.59 | dijib | with this card inside |
| 19:44.06 | [TK]D-Fender | dijib, the modules are colour coded |
| 19:44.13 | dijib | brilliant |
| 19:44.14 | [TK]D-Fender | dijib, red=FXO, green=fxs |
| 19:44.15 | dijib | i think red? |
| 19:45.21 | dijib | how does this handle t.72 |
| 19:45.33 | dijib | its that the fax proto? |
| 19:45.49 | dijib | i feel invincible right now |
| 19:46.08 | [TK]D-Fender | dijib, Card has nothing to do with T.38 |
| 19:46.12 | [TK]D-Fender | that is SIP <- |
| 19:46.57 | dijib | see i just dont know |
| 19:47.37 | dijib | ive pulled the card fxo-2 is the module, its an AFT-REMORA Model: A-200-R Rev 2.6 2007 |
| 19:47.58 | dijib | dumpster dive brothers |
| 19:48.38 | dijib | ok so i thnk only the two jacks work. as there is no second module. i need an FXO module now |
| 19:48.51 | dijib | im starting to see how its working |
| 19:49.11 | [TK]D-Fender | dijib, how many modules on it? Any expansion slots on the backplane? |
| 19:49.45 | dijib | there are two slots, one occupado with an red fxo-2 |
| 19:49.57 | [TK]D-Fender | Ok, then a base card with 2 ports |
| 19:50.07 | [TK]D-Fender | module = 2 FXO ports |
| 19:50.12 | dijib | its got a seperate ec module on the mainboard |
| 19:50.31 | [TK]D-Fender | thats an A200de then |
| 19:50.35 | [TK]D-Fender | amazing find |
| 19:50.37 | dijib | so i cant do anything with the bloody thing without ma'bell |
| 19:50.51 | [TK]D-Fender | Well.. its an analog card.. what do you want? |
| 19:50.54 | dijib | i also grabbed some big clunky plastic mac. |
| 19:51.04 | dijib | i want fxs |
| 19:51.08 | dijib | and only fxs |
| 19:51.21 | dijib | and another ups |
| 19:51.34 | dijib | added to the list |
| 19:51.56 | [TK]D-Fender | dijib, PCI(x) FXS = ASS. Get an ATA |
| 19:52.03 | [TK]D-Fender | Far cheaper, more flexible |
| 19:52.29 | [TK]D-Fender | Sell the card, it's worth over $200 |
| 19:52.35 | Kobaz | anyone have an example of polycom ldap directory usage |
| 19:53.37 | dijib | lol cards worth over $200, anybody want it for a C-note? |
| 19:53.56 | dijib | less shipping. |
| 19:54.02 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 19:54.18 | enoch | hi all |
| 19:54.25 | dijib | why is ata better than fxs? |
| 19:54.45 | enoch | should asterisk work using a pci card riser? |
| 19:54.51 | enoch | will it make noises? |
| 19:55.05 | dijib | >>will it make noise |
| 19:55.06 | dijib | ? |
| 19:55.50 | enoch | yep |
| 19:56.00 | enoch | noises in phones |
| 19:56.19 | enoch | wait |
| 19:57.51 | enoch | http://www.intel.com/cd/products/services/emea/ita/motherboards/desktop/d425kt/overview/454079.htm > http://www.logicsupply.com/products/pci122_dflex > tdm410p > OpenVox B100P |
| 19:58.09 | enoch | i need two pci but i want to make my pbx smaller |
| 19:59.27 | [TK]D-Fender | dijib, FXS does a fine job, releives having to support a card in your server, easier to deploy, lower load on your server, and considerably cheaper |
| 20:01.21 | enoch | [TK]D-Fender: will the pci riser make problems? |
| 20:01.45 | [TK]D-Fender | enoch, Not unless its messed up |
| 20:02.37 | enoch | what do u mean for "messed up"? |
| 20:03.07 | [TK]D-Fender | poorly shieled, throws off intereference, etc |
| 20:03.34 | dijib | i hate people, this MAC just booted. althought i guess it might be old it had only 256mb in it |
| 20:03.43 | dijib | i got kicked out of /g/tech ok. |
| 20:03.50 | dijib | perma ban hammer |
| 20:11.20 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 20:17.13 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-90.telkomadsl.co.za) |
| 20:22.12 | mpe | Hi I have a problem, if the phone change IP and send a new INVITE with a new VIA & SESSION, with the same Branch & to/from tag |
| 20:22.12 | mpe | asterisk reply to the old IP |
| 20:24.14 | mpe | and not to the IP in the updatet INVITE |
| 20:25.03 | *** join/#asterisk oglynn (~me@mail4.homisco.com) |
| 20:27.25 | oglynn | I am having an issue on Asterisk 10.1.1 CentOS 6 file.c:698 ast_openstream_full: File demo-congrats does not exist in any format file exists in /var/lib/asterisk/sounds/en/ |
| 20:29.12 | oglynn | I see similar error in /var/log/asterisk/messages but no mention of where its looking |
| 20:33.42 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:34.52 | *** join/#asterisk zkn (~zkn@82.131.68.58.cable.starman.ee) |
| 20:35.15 | oglynn | I have verified files in same places etc on working 1.6.0.9 |
| 20:37.28 | oglynn | anyone got any suggestion? |
| 20:38.09 | mpe | can you make a md5 to check that the files is identical |
| 20:43.54 | oglynn | ->mpe the md5 is different from 1.6 box to 10.1 box |
| 20:44.31 | *** join/#asterisk dug429 (~dug429@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 20:47.08 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 20:48.11 | dijib | any thoughts on a bewolf cluster with asterisk running on it? |
| 20:50.12 | *** join/#asterisk ipconfeng (~ipconfeng@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 20:50.49 | ipconfeng | is there a way to tell Asterisk to re-read sip.conf without disrupting any active channels? |
| 20:52.25 | WIMPy | 'sip reload' |
| 21:00.54 | *** join/#asterisk fofware (~fabian@host63.190-31-63.telecom.net.ar) |
| 21:02.02 | [TK]D-Fender | oglynn, asterisk.con <--- |
| 21:02.07 | [TK]D-Fender | oglynn, asterisk.conf <--- |
| 21:03.56 | oglynn | D-Fender http://pastebin.com/xJ8vxwka |
| 21:05.12 | [TK]D-Fender | oglynn, [directories](!) <-- the (!) disables this section |
| 21:05.26 | [TK]D-Fender | oglynn, remove, restart * and then it'll use the ones ni varlib |
| 21:05.38 | [TK]D-Fender | astvarlibdir => /var/lib/asterisk |
| 21:06.38 | [TK]D-Fender | under the sounds folder in there |
| 21:08.28 | *** join/#asterisk dxd828 (~dxd828@88-109-121-130.dynamic.dsl.as9105.com) |
| 21:08.33 | *** part/#asterisk dxd828 (~dxd828@88-109-121-130.dynamic.dsl.as9105.com) |
| 21:09.40 | oglynn | the file exists in a number of formats in /var/lib/asterisk/sounds/en/ but its still not working |
| 21:10.19 | oglynn | also tried uncommenting asterisk.conf languageprefix = yes |
| 21:11.05 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 21:15.31 | ndespres | any asterisk consultants for hire in NYC? I need someone to work with occasionally on issues for my customers |
| 21:15.50 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:16.06 | ndespres | and I'm admitting that I don't have the resources or skill to manage this stuff myself |
| 21:16.12 | oglynn | D-Fender is there anything else I can check i have 4 instances of demo-congrats in /var/lib/asterisk/sounds/en/ .gsm .alaw .ulaw & .wav |
| 21:16.36 | azv4 | Seriously I hope Panasonic falls into the ocean |
| 21:16.53 | azv4 | Any Panasonic Pros around atm? |
| 21:18.15 | [TK]D-Fender | oglynn, "core show settings" |
| 21:19.19 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 21:20.46 | oglynn | <PROTECTED> |
| 21:22.26 | Qwell | oglynn: module show like format_ |
| 21:22.45 | [TK]D-Fender | Qwell, Look what he'd have to be missing... |
| 21:22.59 | Qwell | [TK]D-Fender: none are builtin. |
| 21:23.01 | [TK]D-Fender | either the prefix is the issue (total * restart required for asterisk.conf changes) |
| 21:23.17 | [TK]D-Fender | Or the files are install with mismatched permissions, etc |
| 21:24.52 | oglynn | @Qwell D-Fender http://pastebin.com/0Y30x6VK |
| 21:27.03 | [TK]D-Fender | oglynn, modules.conf please.... |
| 21:27.03 | *** join/#asterisk ke-esc (~ke-esc@155.229.209.170) |
| 21:27.37 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:27.37 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 21:27.38 | oglynn | i have stopped fixed directories(!) , started still fail. stopped tried languageprefix =no started fail. stopped tried languageprefix = yes started fail. |
| 21:28.43 | oglynn | the permissions on the files are in the pastebin |
| 21:28.47 | ke-esc | Hey all- one quick question here... I'm trying to use func_odbc. I have a simple ready query set up (SELECT `mailbox` from `voicemail` WHERE `mailbox` = '199') just for testing. If I do odbc read ODBC 1 exec, it says Failed to execute query... I can however run this from isql, and my odbc connection is working (same dsn used for realtime which works) |
| 21:28.49 | ke-esc | any thoughts? |
| 21:29.44 | [TK]D-Fender | checkout time, BBIAB |
| 21:30.48 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 21:31.45 | oglynn | modules.conf added to http://pastebin.com/d5nCExPY |
| 21:36.03 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:40.20 | oglynn | <PROTECTED> |
| 21:56.10 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 22:03.53 | citywok | does anybody use t38/fax? how well does it work for you? |
| 22:08.55 | johnwigley | Does anyone know if a patch or other way exists to change the Queueing Strategy mid queue, using something like the penalty rules - what I want to achieve is to have a queue using call longest idle strategy, but if the call has been queueing for more than say 1 min to change into the RingAll strategy. I was wondering if maybe an extension to the queue penalty rules could accomplish it? |
| 22:10.00 | johnwigley | I realise that you could do it by concatenating two different queues, but that completely screws up the aggregate statistics on things like average waiting time, and for the hold time and position announcements etc. |
| 22:10.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:16.23 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 22:16.38 | oglynn | D-Fender sorry to be a pain. i have added modules.conf to pastebin copied a verified good prompts from a 1.6 system and am still stuck http://pastebin.com/d5nCExPY |
| 22:18.20 | *** join/#asterisk myyrdin (~textual@pbx.ipitomy.com) |
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| 22:26.14 | saxa | [TK]D-Fender: i got today my yealink phones, they seem way better than the Grandstream ones. |
| 22:28.08 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-104-242-22.ph.ph.cox.net) |
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| 22:40.53 | saxa | why is /topic still showing dahdi 2.5.x when there is 2.6.0 already out :) |
| 22:51.38 | zkn | exit |
| 22:51.43 | zkn | lol |
| 22:53.39 | *** topic/#asterisk by mjordan -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.1.2 (2012/02/09), 1.8.9.2 (2012/02/09), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 22:53.54 | *** part/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 22:53.57 | mjordan | saxa: there you go |
| 22:54.54 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:54.54 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:57.15 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:57.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 23:08.36 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ocunmlebngdsenuk) |
| 23:18.39 | saxa | [Feb 17 00:14:38] ERROR[31513]: chan_sip.c:14605 register_verify: Peer 'sasa' is trying to register, but not configured as host=dynamic |
| 23:19.07 | saxa | now I upgraded to * 1.8.9.2 and I see this error |
| 23:19.10 | saxa | huh |
| 23:21.55 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 23:33.48 | ChannelZ | saxa: you host a peer that is trying to register.... and it's not a dymic peer. |
| 23:33.52 | ChannelZ | Says what it means, means what it says. |
| 23:36.29 | ChannelZ | oops.. 'you have', not 'you host'. I'm delirious today. |
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