| 00:01.47 | p3nguin | Just for the sake of testing, change your dial plan line to this: |
| 00:02.25 | p3nguin | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|0xxxxxxxxxx.) |
| 00:02.47 | p3nguin | You're just adding |0xxxxxxxxxx. before the ) |
| 00:03.31 | p3nguin | Make that change, then submit. |
| 00:03.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 00:05.58 | autofsckk | p3nguin: done |
| 00:06.42 | p3nguin | Now call a number starting with 0 that is at least 12 digits long. |
| 00:06.58 | autofsckk | p3nguin: i have called someone and after hangup the x-lite from where the call was made still was on, didnt hung |
| 00:06.59 | *** join/#asterisk sflemming (~stefan@85.183.53.64) |
| 00:11.36 | autofsckk | p3nguin: the call was made |
| 00:11.55 | autofsckk | i sounds awful, could it be internal network congestion? |
| 00:11.59 | p3nguin | So it worked correctly? |
| 00:12.10 | p3nguin | It could be a codec problem or network usage. |
| 00:12.28 | p3nguin | Are you using alaw everywhere? |
| 00:12.41 | autofsckk | let me see |
| 00:12.47 | p3nguin | I assume you are in a part of the world which uses alaw, not ulaw. |
| 00:13.14 | p3nguin | You didn't define a codec to use on ANY peer. |
| 00:13.23 | autofsckk | in audio config from line 1 at spa says g729a |
| 00:13.38 | p3nguin | Every peer should have a disallow=all and a allow=alaw to allow only alaw. |
| 00:17.25 | *** part/#asterisk pietro (~pietro@88-149-224-154.dynamic.ngi.it) |
| 00:19.25 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 00:20.34 | autofsckk | p3nguin: done |
| 00:20.45 | p3nguin | sip reload |
| 00:21.33 | autofsckk | p3nguin: the analog telephone conected to the fxs has a "rare" sound before it gives the normal sound so you can dial, it takes about 2 or 3 secs so you can dial |
| 00:23.40 | autofsckk | p3nguin: it sounds excelent now with the alaw codec, thanks |
| 00:24.43 | p3nguin | If you have enough bandwidth, always use alaw; it will be the best quality codec available to match your PSTN. |
| 00:25.05 | autofsckk | g729 for internet conection right? |
| 00:25.27 | p3nguin | If you first disallow all, then allow alaw only, as well as set your device to prefer alaw, that will be the best sound. |
| 00:25.54 | p3nguin | G.729 will not sound as good, but uses less bandwidth. If you have enough bandwidth, use alaw. |
| 00:26.35 | autofsckk | ok i will |
| 00:27.25 | autofsckk | p3nguin: dont you have a little delay between you dialing from your phone and then the sound ringing ? is there a way to fix that? |
| 00:28.38 | p3nguin | If you have a fixed length number to dial, you can use the "instand dial" option in the dial plan. |
| 00:28.58 | p3nguin | If you have variable length numbers, you have to wait a little bit in case you need to dial more digits. |
| 00:29.58 | autofsckk | well there are land lines and cell phones, not the same amount of digits, 1 more for the cell phones |
| 00:30.40 | autofsckk | how is the instant dial option? |
| 00:30.57 | p3nguin | I think the S0 on the end means DIAL NOW |
| 00:31.13 | p3nguin | I think if you use S1, it will wait one more second for another digit. |
| 00:31.17 | p3nguin | S2 for waiting two seconds. |
| 00:31.23 | autofsckk | ahhh you mean the dial plan on the spa right? |
| 00:31.25 | p3nguin | Try tuning that part I had you add. |
| 00:31.27 | p3nguin | Yes. |
| 00:31.38 | p3nguin | Add S2 before the ) |
| 00:31.44 | p3nguin | or S1 |
| 00:31.50 | autofsckk | it says S0 |
| 00:31.51 | p3nguin | See if that changes things. |
| 00:32.03 | p3nguin | Did you add S0? I certainly did not type it. |
| 00:32.58 | autofsckk | nop |
| 00:33.00 | autofsckk | i didnt |
| 00:33.10 | p3nguin | THen it does not say S0. |
| 00:33.12 | autofsckk | theres a S0 at that string you gave me |
| 00:33.15 | p3nguin | I had you add this: 0xxxxxxxxxx. |
| 00:33.28 | p3nguin | You were to add only that pattern. |
| 00:34.05 | p3nguin | So if you add S1 immediately before the ) then you will end up with 0xxxxxxxxxx.S1 as the pattern. |
| 00:34.44 | p3nguin | The patterns are separated with | |
| 00:34.53 | autofsckk | ok let me do it |
| 00:35.05 | p3nguin | By default, there were seven patterns. |
| 00:35.11 | autofsckk | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|0xxxxxxxxxx.SO) |
| 00:35.17 | p3nguin | I had you add an eighth pattern. |
| 00:35.18 | autofsckk | its like that? |
| 00:35.34 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 00:35.42 | p3nguin | Just add S1 to the part I had you add earlier. |
| 00:36.02 | autofsckk | p3nguin: but the delay is before the dialing comes in, just before i dial it has a big time before i have a dial tone |
| 00:37.01 | p3nguin | I thought you described two delays. I understood the part about the delay before, and I have no idea what causes it. I thought you then complained about a delay before it dialed the number you entered, which I just told you how to reduce. |
| 00:37.01 | *** join/#asterisk Russ (~russ@206.29.182.223) |
| 00:37.05 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 00:39.31 | autofsckk | p3nguin: oh i see, i have a rare dial tone before i get the good dial tone, it takes about 3 secs, and then i dial, it takes the phone and after that i get a ring, and after that ring i have a diferent ring hehehehe that last ring is the good one |
| 00:40.18 | sflemming | Hello all, I have a severe problem with my asterisk that crashes every second and restarts itself. I do not really know how to find the problem, can someone help me. I'm on gentoo with the stable 1.8.7.1 and the problem seems to arise in the calendar and res_mysql module |
| 00:40.55 | WIMPy | autofsckk: Could it be some information tone like MWI? |
| 00:41.26 | autofsckk | WIMPy: whats a mwi tone? |
| 00:41.34 | autofsckk | let me google it |
| 00:41.36 | WIMPy | sflemming: You could disable autoload and load the modules by hand until it crashes. |
| 00:42.00 | WIMPy | Or you start asterisk with -c and see how far it gets or you use gdb. see |
| 00:42.07 | WIMPy | ~collectdebug |
| 00:42.07 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 00:42.16 | WIMPy | autofsckk: Message waiting. |
| 00:42.22 | autofsckk | message waiting dont think so |
| 00:42.31 | autofsckk | im not using voicemails |
| 00:43.10 | sflemming | i figured out that it does not crash when I remove res_mysql and calendar.conf. I backtrace gives me that there seems to be a double free. I recompiled the whole system and the dependencies of asterisk and I gave the config to someone else who tester my configuration. Only difference, he uses x86, me uses amd64 |
| 00:44.11 | sflemming | Asterisk is compiled with debug information and all optimization is disabled but i always get asterisk_wrapper: Asterisk terminated with return code: 1 in my messages, which occurs each second. |
| 00:44.12 | WIMPy | Ok, only two susptect. Almost there. |
| 00:46.13 | sflemming | I'm absolutely unsure whom i can ask with the problem and where to search, I use a kernel 3.0.6 and the problem occurs on all asterisk 1.8 and I need 1.8 for the calendar |
| 00:46.44 | WIMPy | Look at the lik infobot gave to get a backtrace. |
| 00:48.31 | autofsckk | p3nguin: i have another issue, im behing a nat, trying to connect to my asterisk at home, behing nat too, well i already connect, i called an extension there and i can here perfectly clear but there cant hear me, i have nat=yes on the peer conected from outside in general i have nat=yes and i have localnet and externhost defined |
| 00:48.57 | p3nguin | autofsckk: Okay. |
| 00:50.39 | autofsckk | p3nguin: okay? any ideas why they cant hear me? i have opened the firewall on 5060-1 and 10000-20000 udp both, and redirected my router to those ports too |
| 00:50.39 | p3nguin | Make sure you also set directmedia to no in general and in every peer that is not on the same LAN as asterisk. |
| 00:51.06 | p3nguin | You don't need 5061, since SIP is 5060. |
| 00:53.41 | sflemming | WIMPy: Was the infobot message for me? How to open the link? |
| 00:53.42 | autofsckk | orale conectate puto |
| 00:53.57 | autofsckk | ups sorry, wrong window :( |
| 00:54.33 | WIMPy | sflemming Yes that was for you. And I'd try to double-klick it. But I don't know your client. |
| 00:55.54 | sflemming | WIMPy: When I click infobot I open a new window and can speak with infobot, but what to enter? sorry, I'm very rarely in IRC |
| 00:56.59 | WIMPy | Don;t klick infobot. Click the url it posted. Or if that doesn't work, copy that you your browser. |
| 00:58.06 | sflemming | WIMPy: Oh thank you, I did not see that, arg |
| 01:01.09 | *** join/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 01:04.09 | shadowapex | Hi everyone, I'm trying to use SendFax using spansdp and I'm having a problem with the dialplan. Whenever the call gets hung up by the receiving end, dialplan execution stops. I need the dialplan to continue so I can get information about that fax and whether or not it was successful. Is there any way to make Asterisk continue executing the dialplan after a call hangs up? |
| 01:04.57 | WIMPy | 'core show application Dial' |
| 01:06.04 | shadowapex | I never use the Dial application in the dialplan |
| 01:06.12 | shadowapex | Would that still apply? |
| 01:06.32 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 01:06.33 | WIMPy | So how do you make a call? |
| 01:06.43 | shadowapex | exten => s,n,SendFax(${faxlocation}/${faxfile},f) |
| 01:07.20 | shadowapex | This is the full context: http://pastebin.com/r7Ymj72c |
| 01:08.04 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 01:08.07 | shadowapex | Essentially, if the call exits prematurely, the dialplan execution stops after SendFax. |
| 01:08.08 | WIMPy | I'd expect you to call some phone number in order to send a fax. |
| 01:08.38 | shadowapex | I'm using AMI to originate the call. |
| 01:09.19 | WIMPy | Swap the two call legs. |
| 01:09.42 | shadowapex | What I'm sending to AMI: http://pastebin.com/qBhvqyDP |
| 01:09.47 | shadowapex | How do you mean? o.O |
| 01:10.23 | WIMPy | You let the destination number call your fax extension. |
| 01:10.35 | WIMPy | Make your fax extension call the destination number. |
| 01:10.57 | WIMPy | Or try to use the h extension. |
| 01:11.37 | shadowapex | How do I use the h extension? Just put "Exten: h" in the origination action? |
| 01:11.52 | WIMPy | In your dialplan. |
| 01:12.09 | WIMPy | h is called on hangup. But I'm not sure that works in that scenario. |
| 01:12.20 | shadowapex | hmm |
| 01:12.35 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 01:15.31 | shadowapex | Hmmm.. How can I let the destination number call my fax extension if I'm trying to originate the call? |
| 01:19.52 | shadowapex | Actually, I think using the h extension just may have done the trick :D |
| 01:20.06 | shadowapex | Thank you WIMPy! |
| 01:22.03 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 01:31.34 | *** join/#asterisk coppice (~chatzilla@m121-203-221-104.smartone.com) |
| 01:37.25 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 01:42.01 | autofsckk | p3nguin: do the remote sip client has to move something on the router so he can connect to my * ? |
| 01:54.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 02:07.16 | *** join/#asterisk luckman212_phone (~luckman21@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 02:21.10 | *** join/#asterisk james_zhu (~Administr@211.162.113.248) |
| 02:22.24 | *** part/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 02:28.30 | p3nguin | autofsckk: Not usually, no. |
| 02:29.04 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 02:30.05 | autofsckk | i did the changes about directmedia=no on general and sip clients connecting from the outside and it still doesnt work |
| 02:31.31 | autofsckk | let me put the sip.conf so you can see it, gimme a sec |
| 02:33.13 | *** join/#asterisk bulletrtr (48af881f@gateway/web/freenode/ip.72.175.136.31) |
| 02:33.43 | *** join/#asterisk coppice (~chatzilla@m121-203-203-52.smartone.com) |
| 02:33.59 | bulletrtr | TOPIC #asterisk |
| 02:39.33 | p3nguin | autofsckk: Fix line 7 to say externhost, not eternhost. Delete line 9. |
| 02:39.55 | p3nguin | canreinvite has been replaced with directmedia -- you don't need both. |
| 02:40.59 | bulletrtr | Hello p3nguin! Hey, I wanted to thank you for the originate channel direction to get the paging call to simultaneously dial with a cell phone dial command. It works great! |
| 02:41.12 | p3nguin | Lines 95 and 96 are jacked: disallow=all allow=all ? Really? |
| 02:41.39 | autofsckk | ha |
| 02:42.23 | autofsckk | p3nguin: theres another canreinvite=no should i delete it too? |
| 02:42.28 | p3nguin | You disallow all, then you allow only the codec(s) you want to be available. |
| 02:42.41 | p3nguin | Change any canreinvite to directmedia. |
| 02:43.07 | p3nguin | Every peer that is on the other side of a NAT should have nat=yes and directmedia=no in the peer entry. |
| 02:43.40 | p3nguin | If you have a phone behind a NAT somewhere, it needs to have nat=yes and directmedia=no |
| 02:43.55 | autofsckk | ok |
| 02:44.06 | p3nguin | If you have a phone on the same LAN as asterisk, you can use directmedia as long as there is no NAT involved on the other side of the call. |
| 02:44.48 | p3nguin | For your ITSP, it should also have directmedia=no, but they are not behind NAT, so nat=no too. |
| 02:45.28 | p3nguin | Maybe I can summarize what I was trying to say... |
| 02:45.37 | *** part/#asterisk talntid (~t@c-67-168-114-26.hsd1.wa.comcast.net) |
| 02:45.54 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 02:45.57 | autofsckk | what does Using SIP RTP Cos mark 5 means? |
| 02:45.58 | p3nguin | If there is any NAT between the peer and asterisk, directmedia=no. |
| 02:46.39 | p3nguin | If Asterisk is behind NAT and the peer is also behind another NAT, nat=yes. |
| 02:47.06 | p3nguin | If Asterisk and the peer are on the same LAN, nat=no. |
| 02:47.19 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 02:48.24 | autofsckk | what does directmedia mean? when is it useful? |
| 02:56.11 | p3nguin | directmedia is the ability for RTP to go directly between the endpoints, leaving asterisk out of the path. |
| 02:56.43 | p3nguin | If NAT is involved, asterisk needs to remain in the path to act like a proxy. |
| 02:56.54 | p3nguin | "like a proxy" ... not a proxy. |
| 02:58.43 | SeRi | yawns |
| 02:58.47 | SeRi | good morning guys |
| 02:58.49 | SeRi | :) |
| 03:01.47 | autofsckk | oh i see |
| 03:03.26 | autofsckk | a friend of mine is trying to connect, well it is connected to the box now, but when i try to dial his extension it sounds busy, he es trying to dial too and busy, could it be because of the modifications i did? we did connect before the only thing is that i couldnt hear a thing but he could hear me |
| 03:03.35 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:06.18 | p3nguin | Which extension are you calling? |
| 03:07.18 | autofsckk | } |
| 03:07.29 | autofsckk | 4001 |
| 03:07.33 | autofsckk | from 5000 |
| 03:08.07 | p3nguin | You don't call from extensions; you dial to extensions. |
| 03:08.15 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 03:09.41 | p3nguin | Extension 4001 says to dial SIP/4001. Does 4001 have an IP address listed if you check "sip show peers"? |
| 03:10.16 | autofsckk | yes it does |
| 03:10.43 | p3nguin | Okay, so the next thing is going to be a sip debug of the call. |
| 03:10.48 | p3nguin | sip set debug on |
| 03:11.08 | p3nguin | Call 4001, watch the info fly by. |
| 03:11.12 | p3nguin | sip set debug off |
| 03:11.29 | p3nguin | Copy *all* the info that went by, and pastebin it. |
| 03:11.45 | p3nguin | And for the love of all that is holy, stop sending it to me in a /notice. |
| 03:12.39 | *** join/#asterisk mintos (~mvaliyav@114.143.161.232) |
| 03:12.40 | p3nguin | Paste the pastebin URLs in the channel so other people can contribute. Otherwise, I'm going to have to start billing for private help. |
| 03:13.08 | autofsckk | ok i will, sorry |
| 03:13.27 | autofsckk | im waiting for my friend to connect so he can dial |
| 03:16.22 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 03:17.17 | autofsckk | and now my friend isnt paying attention to msn :S |
| 03:18.08 | *** join/#asterisk adolfomaltez (~taro@190.62.214.80) |
| 03:21.11 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 03:39.36 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001550.dsl.bell.ca) |
| 03:42.21 | autofsckk | p3nguin: thats the debug http://paste.ttnk.org/25 |
| 03:43.18 | *** join/#asterisk master_of_master (~master_of@p57B53B4D.dip.t-dialin.net) |
| 03:47.34 | p3nguin | What device is at 189.240.83.78? |
| 03:48.30 | autofsckk | an x-lite sip client |
| 03:49.01 | p3nguin | There is a NAT there? |
| 03:49.06 | autofsckk | yes |
| 03:49.26 | autofsckk | hes behind a router |
| 03:50.59 | p3nguin | What is at 69.43.161.160? |
| 03:51.18 | autofsckk | i dont know |
| 03:52.19 | autofsckk | it shouldnt be connected if it is heheheheh not shown on sip show peers |
| 03:52.46 | *** join/#asterisk [1]SnakeDoctor (~mtaylor1@206-188-70-99.cpe.distributel.net) |
| 03:53.40 | autofsckk | p3nguin: where did you take that ip from? |
| 03:53.40 | p3nguin | Contact: <sip:4001@69.43.161.160:5060> |
| 03:53.50 | p3nguin | Something/someone is using 4001 at that address. |
| 03:55.25 | autofsckk | well 4001 is the pap2 |
| 03:55.50 | p3nguin | Is it on the same LAN as Asterisk? |
| 03:56.09 | autofsckk | yes it is |
| 03:56.55 | p3nguin | So you have no idea where that IP address came from? |
| 03:57.01 | autofsckk | not really |
| 03:57.06 | p3nguin | That's not good. |
| 03:57.11 | autofsckk | i know |
| 03:57.38 | autofsckk | thats why i dont like to put paste on channels and public places heheheh |
| 03:58.10 | autofsckk | but well, learning has a price :D |
| 03:58.45 | p3nguin | Don't pastebin your passwords, and you'll have a much better chance at survival. |
| 03:59.02 | autofsckk | its says is an ip from australia |
| 03:59.27 | autofsckk | havent paste any password |
| 04:00.12 | p3nguin | You're going to have to figure out why your INVITE from 4001 has an Australian contact address. |
| 04:01.14 | autofsckk | but where did you see that ip |
| 04:01.16 | autofsckk | i dont gettit |
| 04:01.21 | p3nguin | You're in Mexico, right? |
| 04:01.27 | autofsckk | if i do a sip show peers |
| 04:01.36 | autofsckk | its not there |
| 04:01.38 | autofsckk | yes i do |
| 04:01.50 | p3nguin | It's in the invites, line 220 is one instance of it. |
| 04:02.13 | autofsckk | what are those invites? |
| 04:02.32 | p3nguin | Invites are how a call is established. |
| 04:02.57 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-212.ks.ks.cox.net) |
| 04:03.46 | p3nguin | One phone sends an invite to asterisk, asterisk accepts and that leg of the call is created. Asterisk then invites another peer to have a call, that peer accepts, and that leg of the call is made. Now you have an end-to-end phone call. |
| 04:05.20 | p3nguin | The problem isn't the invite, it's the really strange IP address that probably isn't the right address for the peer involved in the call. |
| 04:05.41 | p3nguin | Anything useful in "sip show peer 4001"? |
| 04:05.45 | autofsckk | so what can that ip do? steal some phone calls? should i be very worried? im really i newbie on asterisk |
| 04:06.59 | p3nguin | This is one reason to not use such simplistic names for phones. Scanners check those basic numbers more often than they check abstracted names, such as the MAC address on the phone. |
| 04:07.23 | ketas | hahaha |
| 04:07.24 | p3nguin | Like a SIP device with the name 123.... TERRIBLE name for a device! |
| 04:08.06 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 04:09.27 | autofsckk | you want to see the info from sip show peer 4001 ? |
| 04:10.05 | p3nguin | You can look it over and see if the correct IP address and other information is there. |
| 04:10.16 | autofsckk | yes it is |
| 04:10.33 | p3nguin | I'm so confused about that other address. |
| 04:10.34 | autofsckk | is the ip from the pap2 in my lan |
| 04:12.53 | autofsckk | p3nguin: what kind of name for a sip device is recommended? |
| 04:13.25 | p3nguin | ~devicenames |
| 04:13.25 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 04:13.56 | SeRi | p3nguin, in order for my brother to recive a call to hes voip.ms sub account he needs a DID? |
| 04:14.03 | p3nguin | For soft phones, I typically use the MAC address of the primary (or only) ethernet device. |
| 04:14.18 | p3nguin | seri: Yes and no. |
| 04:14.30 | p3nguin | You only need a DID to get a call from the PSTN. |
| 04:14.47 | SeRi | bingo |
| 04:14.56 | p3nguin | You can get calls from other accounts via internal voipms extensions. |
| 04:15.10 | p3nguin | and SIP URIs |
| 04:16.34 | SeRi | He will be traveling to Dominic Republic and he needs a customer to call him over via VoIP. |
| 04:16.42 | SeRi | cheaper :) |
| 04:17.31 | SeRi | Thanks for the help. |
| 04:18.44 | autofsckk | so what can i do about that au ip? any idea? |
| 04:18.53 | p3nguin | If you had an attendant, callers could press 1 to reach you, 2 to reach him, and 3 to do something else useful. |
| 04:19.02 | p3nguin | autofsckk: Block it at the firewall if it worries you. |
| 04:19.56 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 04:30.07 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:34.39 | SeRi | p3nguin, You use android? |
| 04:34.45 | p3nguin | No. |
| 04:34.59 | SeRi | ok. |
| 04:35.19 | SeRi | looking for a good android soft phone. |
| 04:35.22 | SeRi | :) |
| 04:36.23 | *** join/#asterisk DerkKo (~afernande@199.255.83.50) |
| 04:36.32 | SeRi | ekiga sucks |
| 04:36.37 | DerkKo | Sup guys |
| 04:36.55 | SeRi | hola |
| 04:37.22 | DerkKo | Hola |
| 04:37.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 04:37.56 | SeRi | hola [TK]D-Fender |
| 04:39.44 | *** part/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 04:40.41 | DerkKo | Im running into a problem, where opensips is a stateless proxy but asterisk behind it continues to send an invite after an ack on an asterisk behind it |
| 04:40.49 | *** join/#asterisk gladier (~gladier@pedobear.molested.me) |
| 04:42.49 | gladier | hey guys... does anyone know of CNG (RFC3389) is supported on asterisk 1.8? I've just noticed that CNG is enabled on my epygi but asterisk isn't bitching about it |
| 04:43.27 | dijib | seri im going for a few smokes and joining my confbridge |
| 04:43.34 | dijib | if your arouns SeRi |
| 04:43.43 | SeRi | cool |
| 04:43.52 | SeRi | almost dont on the phone ill join in a bit |
| 04:43.53 | dijib | u down to join? |
| 04:43.59 | SeRi | sure |
| 04:44.07 | dijib | k well ill be a few min still |
| 04:44.53 | ChannelZ | girl talk? |
| 04:45.49 | DerkKo | gladier: your ep is probably inserting comfort noise packets into the call without letting you know in the SDP that it should. |
| 04:47.19 | gladier | thing is i have the same modem epygi on another asterisk 1.4 install which is bitching nonstop about CNG being enabled even though the epygi has it disabled (mind you i think this is an epygi bug) |
| 04:47.38 | SeRi | dijib, let me know. |
| 04:48.11 | DerkKo | Should not matter you can ignore it, this just means asterisk is probably dropping comfort noise packets, the call experience is not affected at all. |
| 04:49.17 | gladier | yea... it was mroe a curiosity than anything - the 1.4 box is having call quality issues so im just ruling stuff out as i go along |
| 04:49.49 | dijib | im in |
| 04:53.36 | SeRi | dijib, give me a few min. working with my brother |
| 04:54.57 | dijib | yep. you should also buy another DID for like $0.99 to have his own number. |
| 04:55.06 | dijib | im stepping out for SMOKE(s) |
| 05:07.26 | SeRi | why people fear what they do no understand.... |
| 05:14.19 | *** join/#asterisk irroot (~irroot@197.109.39.170) |
| 05:16.55 | p3nguin | Do the qos settings such as tos_audio have default values, or do I have to uncomment them to make them active? |
| 05:18.43 | SeRi | p3nguin, in what> |
| 05:18.45 | SeRi | ? |
| 05:19.09 | dijib | im sleepy. |
| 05:19.11 | dijib | out of conf |
| 05:19.12 | autofsckk | SeRi: cause they feel insecure of not knowing how to deal with something they dont know anything about? |
| 05:19.48 | SeRi | dijib, sorry still on the phone |
| 05:19.54 | SeRi | helping my brother with a soft phone |
| 05:20.15 | dijib | to a sub account. |
| 05:20.19 | SeRi | yes |
| 05:20.24 | SeRi | he is connected now. |
| 05:20.29 | SeRi | but couldnt before |
| 05:20.29 | dijib | got it. you vnc'ing or tsc? |
| 05:20.36 | SeRi | now I am seting up call forwarding for him |
| 05:20.37 | dijib | ah i see |
| 05:20.41 | dijib | what router is he behind? |
| 05:20.48 | SeRi | kerio appliance |
| 05:20.51 | dijib | nevermind |
| 05:20.53 | dijib | your good |
| 05:20.58 | dijib | he have an embedded system? |
| 05:21.09 | SeRi | He has a server appliance |
| 05:21.56 | p3nguin | seri: I'm referring to the qos settings in sip.conf. |
| 05:22.25 | SeRi | p3nguin, I se.. I know nothing about thet :/ |
| 05:22.31 | irroot | morning folks |
| 05:22.39 | SeRi | morning irroot |
| 05:23.19 | p3nguin | Many settings have default values that are applied unless overridden. I'm wondering if the qos settings such as tos_audio have a default value if not specifically overridden. |
| 05:23.32 | SeRi | ah. I see. |
| 05:24.44 | irroot | p3nguin looking at the source the global settings are set |
| 05:25.06 | p3nguin | So if I do not uncomment tos_audio, the default value will be ef? |
| 05:26.02 | irroot | im checking |
| 05:26.44 | SeRi | I now understand the frustration you vet guys have to go threw when some users just dont get it.... |
| 05:27.18 | irroot | p3nguin ok not so simple |
| 05:27.36 | irroot | http://pastebin.com/dbmjX2cY |
| 05:27.50 | SeRi | I love my brother but hes windows mentality is not helping him think out side the box. |
| 05:27.51 | irroot | the defined default is 0 |
| 05:28.48 | p3nguin | I must define it, then. Not a problem, just wanted to make sure. |
| 05:32.44 | SeRi | dijib, kerio appliance can be one pain in the ass. there rule matching just dont make sense. |
| 05:32.57 | SeRi | their* |
| 05:42.33 | SeRi | dijib, you in? |
| 05:43.15 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:43.38 | SeRi | looks like hes out. I am out too. g/n all. |
| 05:46.49 | irroot | gn |
| 05:51.41 | irroot | everytime a call wen to null Q it bombed |
| 05:51.53 | irroot | you know this |
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| 06:01.37 | ChannelZ | Do what now? |
| 06:10.41 | dijib | im in |
| 06:12.20 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 06:17.24 | *** join/#asterisk james_zhu (~Administr@211.162.113.248) |
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| 06:28.02 | p3nguin | Will RTP packets from the internet which are delivered on a network without QoS still have the DSCP tags intact when they arrive? |
| 06:28.44 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 06:30.48 | p3nguin | It seems like traffic shaping on my upstream actually helps audio quality when there's other crap going on, but I'm curious if shaping the downstream will have any effect at all. |
| 06:34.26 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
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| 06:55.01 | ChannelZ | You can't shape downstream traffic |
| 06:58.43 | p3nguin | I can certainly try to queue downstream traffic, but my point was that I didn't know if it would do any good or not. |
| 06:58.44 | kaldemar | sure you can try but it's close to no use if the whole router path (or the bottleneck nodes) doesn't support it. |
| 07:00.19 | p3nguin | I know I can't do policy-based QoS without support from the ISP, so I'm not even going to bother with that. |
| 07:01.32 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
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| 07:14.29 | SeRi | p3nguin, you avail> |
| 07:14.30 | SeRi | ? |
| 07:17.45 | SeRi | any body in that can help me out with a quick context |
| 07:19.42 | *** part/#asterisk nny (~Scott@174.107.223.14) |
| 07:24.08 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 07:25.40 | ChannelZ | yes |
| 07:29.00 | SeRi | ChannelZ, Is this valid context? |
| 07:29.28 | SeRi | http://pastebin.com/37Rj04pn |
| 07:33.57 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 07:34.43 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:34.43 | ChannelZ | arrrgh |
| 07:35.45 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:36.49 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:36.51 | schmidts | good morning |
| 07:36.52 | ChannelZ | SeRi: yes I don't see anything wrong offhand |
| 07:38.04 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:38.25 | ChannelZ | and hi schmidts |
| 07:45.46 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
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| 07:49.34 | *** join/#asterisk fish9370 (~Miranda@fpg.ru) |
| 07:50.29 | fish9370 | Hello |
| 07:50.57 | fish9370 | I have problem with receiving fax |
| 07:51.06 | fish9370 | [Nov 10 11:33:45] WARNING[7434] app_fax.c: Error transmitting fax. result=49: The call dropped prematurely. |
| 07:51.06 | fish9370 | [Nov 10 11:33:45] WARNING[7434] app_fax.c: Transmission error |
| 07:51.52 | fish9370 | please help |
| 07:51.54 | schmidts | fish9370 does this happens every time? |
| 07:51.58 | fish9370 | yes |
| 07:52.36 | schmidts | the error message just says that the opposite fax machine has hangup the call before it was finished, or even before they finish syncing |
| 07:53.02 | fish9370 | how I can fix this? |
| 07:53.25 | ChannelZ | What kind of channel? |
| 07:53.59 | fish9370 | e1->iax2->my_host |
| 07:54.13 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 07:56.42 | fish9370 | if I receive from sip it's work fine |
| 07:57.21 | ChannelZ | hmm.. is the iax2 link using ulaw/alaw? |
| 07:57.32 | fish9370 | alaw |
| 07:59.00 | schmidts | which asterisk version do you use fish9370? |
| 07:59.09 | fish9370 | meybe, I can debug spandsp? |
| 07:59.16 | fish9370 | 1.6.20 |
| 07:59.29 | fish9370 | 1.6.2.20 |
| 07:59.32 | schmidts | i dont had this special problem but i had some problems with sending or receiving faxes over iax2 but not so many problems with sip |
| 08:01.07 | fish9370 | maybe, need do something with channel for receiving fax? |
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| 10:10.40 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 10:11.17 | phpboy | I've been banging my head against the wall for some time now, how can I get the Agent's Name from agents.conf to reflect in the queuelog on ADDMEMBER, is it even possible? |
| 10:14.19 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 10:15.23 | irroot | phpboy how do you add them as members in queues.conf ? |
| 10:16.07 | irroot | you should set the membername in there and it should work there is also a option in trunk to log on membername not interface |
| 10:16.15 | irroot | what version you using ? |
| 10:16.21 | phpboy | exten => login,n,AddQueueMember(${QUEUEDST},SIP/${CALLERID(num)},1,,${AGENT(${AID}:name)}) |
| 10:16.42 | irroot | ok add member name there too |
| 10:16.55 | phpboy | 1.8.5 - I authenticate like this... exten => login,n(auth),Authenticate(${AGENT(${AID}:password)}) |
| 10:17.18 | phpboy | doesn't ${AGENT(${AID}:name)} cover that? |
| 10:18.31 | irroot | yip should do |
| 10:18.31 | irroot | and queue show shows the member name ? |
| 10:18.57 | phpboy | yeah |
| 10:19.39 | irroot | phpboy the option is in trunk [not even 10] |
| 10:19.39 | irroot | you can edit it in quite easily to the source |
| 10:19.39 | irroot | im busy working on cleanups and things too app_queue |
| 10:21.11 | phpboy | irroot: I don't quite follow what ur saying? |
| 10:21.36 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 10:22.21 | irroot | phpboy you can edit in aqm_exec to log the membername |
| 10:22.53 | irroot | in the source code its going to log the interface by default |
| 10:23.09 | phpboy | so I'll have to recompile? |
| 10:23.18 | irroot | in "next years" version of asterisk version 11 there is support for this as a option |
| 10:23.54 | irroot | you can edit it yourself and recompile yes that is the simplest option |
| 10:24.17 | phpboy | I'm little worried about that |
| 10:24.35 | phpboy | I think I'm going to rather write a shell script that gets the name and passes it in |
| 10:25.01 | phpboy | when the Agent logs on, probably a 'safer' way if you will |
| 10:27.16 | irroot | or you can have a custom log event for your own purpose |
| 10:27.31 | phpboy | that may be a better idea even |
| 10:27.38 | *** part/#asterisk james_zhu (~Administr@211.162.113.248) |
| 10:27.50 | irroot | then ignore ADDMEMBER and have a MY_ADDMEMBER if you want |
| 10:27.56 | irroot | can be done in dialplan |
| 10:28.01 | phpboy | that I can do |
| 10:28.24 | phpboy | but, is there a way to not print the ADDMEMBER in the log, kinda overwrite it with my own? |
| 10:28.35 | irroot | look at the QueueLog application |
| 10:28.58 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:29.05 | irroot | phpboy its generated from the code so cant overwrite it |
| 10:30.32 | phpboy | k |
| 10:30.38 | phpboy | thanks a lot for ur help |
| 10:30.51 | irroot | phpboy pleasure |
| 10:31.19 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 10:35.56 | phpboy | Why's the option there if it doesn't actually work? :\ |
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| 10:48.00 | *** join/#asterisk tdi (tdi@91.206.27.15) |
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| 10:48.48 | tdi | hi there, ive got HFC-S BRI card, dahdi_hardware shows it, dahdi_scan and dahdi_tool do not, where can i go (docs etc) to make it work ? |
| 10:51.51 | WIMPy | tdi: Serveral options. Summary at http://voice.yeti.dk/Asterisk_vs_ISDN |
| 10:52.12 | WIMPy | dahdi does not natively suppoty these cards. |
| 10:52.30 | tdi | WIMPy: so what should i do ? |
| 10:52.54 | tdi | WIMPy: elastix detects them, my vanilla debian does not |
| 10:53.11 | tdi | thanks for the link anyways. |
| 10:53.41 | WIMPy | Use the native Linux drivers or get a patch[ed| for] dahdi. |
| 10:54.25 | tdi | native ? zaphfc ? |
| 10:54.39 | WIMPy | native=misdn |
| 10:54.53 | WIMPy | It's all on that page. |
| 10:54.54 | tdi | hfcpci then ? |
| 10:55.00 | WIMPy | Including links. |
| 10:55.04 | tdi | yes thanks, Im on that page |
| 10:55.07 | WIMPy | yes |
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| 11:08.52 | tdi | WIMPy: I just wanted to thank you again, just made it work |
| 11:09.11 | tdi | with dahdi-zaphfc from git |
| 11:09.17 | WIMPy | Great |
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| 11:15.05 | *** join/#asterisk rqan (~rqan@91.103.2.203) |
| 11:16.56 | rqan | Hi All, I have taken over the administration of an office phone network from a VOIP administrator, and I have very little experience, so would be very greatful if somone could assist. We use freepbx. We have to record all calls, this works fine for everyone except one person. As far as I can see there is nothing unique about him, but the monitored calls are really choppy and distorted. He isd using a softphone (but not the only person using one) |
| 11:17.17 | rqan | Would anyone be able to advise about this choppy distorted sound? |
| 11:21.07 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 11:23.13 | WIMPy | Are you saying that only the recording is choppy? |
| 11:23.29 | WIMPy | I.e. the converstaion itself is normal? |
| 11:25.27 | rqan | @WIMPy Yes, it is very strange |
| 11:25.58 | rqan | The person would make a lot of calls, call quality os perfect, but when the recording is checked, it is terrible |
| 11:27.48 | *** join/#asterisk Stese (3eabc605@gateway/web/freenode/ip.62.171.198.5) |
| 11:29.13 | rqan | WIMPy, I use the same softphone - twinkle on ubuntu, and t |
| 11:29.44 | Stese | Hey all!" |
| 11:30.12 | Stese | Can someone look at this pastebin and spot my noob mistake? (new Asterisk 1.8 on Centos6 VM) |
| 11:30.13 | Stese | http://pastebin.com/Y0zzdhXp |
| 11:30.25 | WIMPy | rqan: Same codec everywhere? |
| 11:30.29 | rqan | Hee hee, Boss just said we had to stop using soft phones, the irony of it all |
| 11:31.15 | Stese | ~book |
| 11:31.15 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 11:31.25 | WIMPy | Stese: Remove the spaces after the commas. |
| 11:31.33 | rqan | WIMPy, yea actually, I though that was the problem, so changed mine and his to G.711 A-law |
| 11:31.37 | WIMPy | and before. |
| 11:31.58 | WIMPy | rqan: Sounds very strange. |
| 11:32.46 | rqan | yea, looks like the mystery wont be resolved for a little whle anyway, untill boss changes her mind. (It is like dilbert in here i swear). I really appriciate your help tho mate, thank you |
| 11:33.35 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:33.37 | Stese | WIMPy: Done that, no joy... I thought I was using spaces before on 1.6 without issues ('cept in context names) |
| 11:40.27 | Stese | Any one else? |
| 11:41.04 | WIMPy | Did you reload the config? |
| 11:42.24 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:44.12 | Stese | yeah |
| 11:44.19 | Stese | a few times over :P |
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| 11:46.11 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:55.37 | Stese | this is weird... never had this before... :s, it's not as if I'm trying anything complicated |
| 11:56.55 | WIMPy | Did you also remove the space between = and > ? |
| 11:57.28 | Stese | no |
| 11:57.35 | Stese | but i've never had to before.. |
| 11:59.07 | Stese | ok, that got it... weird! i've not got access to my old 1.6 dialplans, but i'm fairly sure that it let me use = > |
| 11:59.10 | Stese | sorry to be a pain! |
| 11:59.34 | WIMPy | np |
| 11:59.44 | WIMPy | I just know that it can be picky about spaces. |
| 12:00.06 | WIMPy | Even if I'm not sure in what places exactely. |
| 12:00.10 | Stese | :) maybe it's more picky in 1.8 |
| 12:00.27 | WIMPy | possibly |
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| 12:03.56 | Stese | gah, I can't remember how to do templates in the sip.conf... isn't is something like [name](!) for the template and then [sipuser](name) for the user? |
| 12:04.13 | WIMPy | exactely |
| 12:04.21 | Stese | :) |
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| 12:15.03 | itsupport | hi everybody |
| 12:15.21 | itsupport | i'm very new to linux |
| 12:15.49 | itsupport | I'm using pbx in a flash and have few queries |
| 12:16.19 | itsupport | can anybody help me |
| 12:16.53 | Stese | Ask away |
| 12:17.06 | itsupport | i just want to know what is the root mysqluser ? |
| 12:17.40 | Stese | Hmm, I'm guessing PBX in a Flash is a product based on Asterisk? |
| 12:17.57 | itsupport | yes |
| 12:18.54 | Stese | I'd think then that contacting the people who produce that product would be able to help... asterisk itself doesn't have mySQL built in.... it can interface with it when configured to do so |
| 12:19.40 | itsupport | ok |
| 12:19.42 | itsupport | thanx |
| 12:19.46 | itsupport | bye |
| 12:22.44 | *** join/#asterisk snowlord (~snowlord@80.149.125.74) |
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| 12:41.07 | wonderworld | could someone give me a hint on how to send a voice announcement to users of a confbridge conference (triggered by a cron job) |
| 12:47.18 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
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| 12:51.01 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 13:11.48 | *** join/#asterisk dlublink (~david@206-248-174-128.dsl.teksavvy.com) |
| 13:12.37 | dlublink | Is there a tool I can run that will automatically detect one way audio and send an email or log to a log file? All I need is the call that has one way audio and the time. |
| 13:13.12 | Stese | You want to record a call and then email the call out? |
| 13:13.53 | Stese | like a voicemail, or are you looking for a system to detect bad calls? |
| 13:14.16 | Chainsaw | Stese: It sounds like a NAT issue notification system to me. |
| 13:14.52 | Chainsaw | Stese: You can probably get the RTP failures from the main asterisk log I suppose, if you turn verbosity up suitably high. |
| 13:16.11 | Stese | Chainsaw: It's not for me, it's for dlublink :P, but I think you are right, and hope he can find a solution :) |
| 13:16.46 | dlublink | I want to detect bad calls, I have so few of them my users don't report them anymore |
| 13:16.57 | jkroon | dlublink, yes, such tools exist, they are very expensive. |
| 13:17.06 | dlublink | Oh |
| 13:17.11 | jkroon | i've started on a floss one, but lost interest again, and it requires some re-thinking. |
| 13:17.24 | dlublink | Given that I have 2-3 such calls a week, it is probably not worth it |
| 13:19.25 | jkroon | trust me, when you start having 50+ clients it's a GOOD thing to have. you want to pick up problems before they do. |
| 13:19.47 | jkroon | at a few hundred clients I'm surprised I don't get more than two/three such calls per month at the moment. |
| 13:19.50 | dlublink | Which do you recommend ? |
| 13:19.56 | Stese | in my limited experience, all the NAT issues i've ever had, were due to SIP ALG's on routers |
| 13:20.10 | jkroon | call drops seems to be prevalent at the moment though, so far all indications points at upstream providers though. |
| 13:20.22 | jkroon | Stese, agreed. |
| 13:20.54 | jkroon | dlublink, i saw a price or two and decided to not bother testing it, so i can't make a recommendation. |
| 13:26.28 | wonderworld | could someone give me a hint on how to send a voice announcement to users of a confbridge conference (triggered by a cron job) |
| 13:26.47 | jkroon | call origination? |
| 13:27.33 | *** join/#asterisk joelio (~Joel@m.etrono.me) |
| 13:27.47 | wonderworld | yes, but i didn't figure out the mechanism to use. originate local channel that runs an agi. agi connects to the konf and plays the sound? |
| 13:27.49 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 13:28.44 | joelio | Hi all, I'm looking for recommendations for a system that can support video conferencing and that does transcoding. It needs to be open source as we use linphone and have extended it to support 1080p resolution |
| 13:28.55 | jkroon | how about originate Local/ext@conf application Playback announcefile |
| 13:29.05 | jkroon | where ext@conf will get the call connected into the conference? |
| 13:29.31 | wonderworld | jkroon: hey, even better, cool. no agi this way, right? |
| 13:29.36 | jkroon | none. |
| 13:29.43 | wonderworld | nice, i'll try that. thanks |
| 13:30.08 | jkroon | is always amused at the type of concoctions people come up with with AGI ... never did manage to figure out the attraction. |
| 13:30.55 | wonderworld | i never used local channels before, so my problem was a mindtwister for me |
| 13:31.12 | jkroon | they are AWESOME |
| 13:31.19 | jkroon | is abusing them for all kinds of purposes. |
| 13:31.23 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:32.41 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:33.27 | wonderworld | what if a single application is not enough. what if i needed to play multiple sounds for example. would i need agi then? |
| 13:34.25 | WIMPy | Use the dialplan or just give multiple parameters to Playback() |
| 13:35.08 | WIMPy | I haven't tried with the 10er version of ConfBridge, but before the whole thing is impossible AFAIK. |
| 13:35.46 | jkroon | wonderworld, no. |
| 13:36.06 | jkroon | ... Playback sound1&sound2&sound3 etc ... |
| 13:36.33 | jkroon | if you need multiple applications you can either build them using AGI or using asterisk dialplan - whichever you prefer (I prefer the latter) |
| 13:36.56 | jkroon | heck, you can even parameterize it with channel variables, and pass those in via a .call file. |
| 13:37.59 | wonderworld | ok. when using dialplan i would use GoTo as application and just use the dialplan at a place where all my needs are happening? |
| 13:38.54 | WIMPy | Never thought of that. The usual way is a local channel again. |
| 13:40.19 | wonderworld | ok, where would that second local channels in the standard procedure originate from? |
| 13:44.30 | *** join/#asterisk nfi|ermes (~sdff@host54-26-static.93-94-b.business.telecomitalia.it) |
| 13:46.25 | dlublink | hey, I found this "No remote address on RTP instance '0x28a13c8' so dropping frame". might this be the error I am looking for ? |
| 13:47.36 | nfi|ermes | i installed dahdi-linux-complete-2.6.0-rc1+2.6.0-rc1 ma module zaphfc is missing , where can i find it ? |
| 13:48.54 | WIMPy | nfi|ermes: dahdi does not support these cards natively. You've got several options, see http://voice.yeti.dk/Asterisk_vs_ISDN |
| 13:51.02 | *** join/#asterisk clintc (~clintc@n128-227-139-116.xlate.ufl.edu) |
| 13:54.31 | nfi|ermes | so i should use dahdi-hfcs |
| 13:54.42 | nfi|ermes | o zaphfc, according to that article |
| 13:54.48 | nfi|ermes | or |
| 13:55.02 | *** part/#asterisk clintc (~clintc@n128-227-139-116.xlate.ufl.edu) |
| 13:55.08 | WIMPy | Or the drivers included with the Linux kernel. |
| 13:55.10 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 13:55.58 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:56.28 | [TK]D-Fender | wonderworld, Point it to and extension instead of Application |
| 13:58.40 | [TK]D-Fender | an* |
| 14:06.02 | wonderworld | [TK]D-Fender: thanks, i tried that but it didn't work -> http://pastebin.com/MtLyBYsY |
| 14:06.35 | wonderworld | local channel joins the confbridge, but is unable to play sound |
| 14:06.52 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:06.59 | WIMPy | >>I haven't tried with the 10er version of ConfBridge, but before the whole thing is impossible AFAIK. |
| 14:07.01 | wonderworld | lol, type... wait, never mind |
| 14:07.02 | [TK]D-Fender | wonderworld, exten=>1111,1,(SayDigit(123456678)) <--- you put your app in backets |
| 14:07.04 | wonderworld | typo even |
| 14:07.11 | wonderworld | yes, just saw that |
| 14:08.16 | wonderworld | yay, it works. thanks guys, you helped me a lot |
| 14:15.55 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mjeippjnjmqndqqv) |
| 14:18.52 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
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| 14:19.12 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:19.18 | *** join/#asterisk narziss (narziss@gateway/shell/anapnea.net/x-yuimaalivkjxwthp) |
| 14:26.01 | wonderworld | ok, it works, but the "bot" stays in the conference. even if it hangs up itself. "local show channels" shows <unowned> -- 1@bot. how would i terminate the bot? |
| 14:28.27 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:29.31 | wonderworld | i guess it should kick itself from the conference? |
| 14:30.08 | anonymouz666 | coppice: In MFC/R2 is it possible to send second DNIS digit without get the bit from the first DNIS digit? |
| 14:31.34 | coppice | some MFC/R2 variants time out under certain conditions, and carry on sending without receiving the handshake tone |
| 14:32.15 | anonymouz666 | if I send one DNIS digit, the correct handshake is to receive one bit on and other off ? |
| 14:32.21 | eppigy | mornin |
| 14:33.39 | coppice | the handshake to a tone is another tone, not a bit change. those signaling bits only change during the initial seize and acknowledge |
| 14:34.10 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 14:34.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:35.39 | anonymouz666 | coppice: understood, I don't receive any RX after my first DNIS digit sent (TX) and then the mfc/r2 stack keep waiting until Dial timeout. |
| 14:35.48 | anonymouz666 | the equipment of telco guy, just works... |
| 14:35.53 | anonymouz666 | it's bizarre |
| 14:35.56 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 14:36.08 | anonymouz666 | never seen a similiar case in the last 5 years |
| 14:36.55 | anonymouz666 | or the other end does not get my DNIS digit or my Asterisk does not recognize the MF tone |
| 14:37.34 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 14:37.36 | tompaw | Hello. |
| 14:38.42 | tompaw | How can I achieve linear queue strategy since agentcallbacklogin was dropped and the dialplan replacement uses dynamic agents list? |
| 14:40.12 | moy | anonymouz666: u can use dahdi_monitor on the line to see if the MF tone is really there (both, outbound and inbound) |
| 14:42.01 | anonymouz666 | moy: sure, can do it right now |
| 14:46.51 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 14:49.57 | r0m|u | hola. |
| 14:53.09 | r0m|u | last night I was having problems with astrisk accepting calls from my gsm gateway. how can I tell asterisk to accept calls from my gsm gateway on the form of callerid@asterisk? I am trying to pass caller id from the gsm gateway to the asterisk to the context can execute the dial plan accordingly.... |
| 14:53.52 | r0m|u | Asterisk keep telling me no such peer... |
| 14:53.53 | blitzrage | r0m|u: match on IP address? |
| 14:54.11 | blitzrage | which should just be a type=peer |
| 14:54.22 | r0m|u | Thats how I have it. |
| 14:54.30 | r0m|u | but it was a no go. |
| 14:56.06 | wonderworld | is the only way to destroy a confbridge conference to kick all remaining users? |
| 15:00.11 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:00.19 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 15:01.37 | nunne | Does anyone know if asterisk sets any channel variables on successfull blind/attended transfer? Because I need to tear down the original call (using the hangup-extension in a macro).. Not only the caller and the person i transfered to gets teared down correctly. |
| 15:02.22 | Katty | i feel |
| 15:02.29 | Katty | discombobulated this morning |
| 15:03.10 | eppigy | maybe you should have a seat over here |
| 15:03.55 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 15:04.43 | r0m|u | I found my issue. the gsm gateway is adding sip: infront of the caller id. |
| 15:04.58 | r0m|u | Is there a way to strip that in asterisk? |
| 15:05.03 | r0m|u | for incomming calls |
| 15:06.07 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 15:06.30 | Katty | sits by eppigy |
| 15:06.43 | Madkiss | is there an asterisk shell command that tells me whether asteirsk is running in real-time mode or not? |
| 15:07.19 | r0m|u | Madkiss, AMI? |
| 15:07.40 | Madkiss | never heard of her |
| 15:07.48 | r0m|u | ~book |
| 15:07.48 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 15:07.56 | r0m|u | look there ^^ |
| 15:08.05 | r0m|u | It will tell you everything about AMI |
| 15:08.37 | r0m|u | she is cute and nice. she plays well with remote commands. BUT dangerous in an open env..... |
| 15:08.44 | Madkiss | i'm just writing an OCF resource agent for asterisk. I'm not gonna read a book some hundred pages long to figure out a single answer :) |
| 15:09.25 | r0m|u | and I dont have time to explain the complexity of AMI. good luck :) maybe some one else does. |
| 15:09.46 | Madkiss | google returns "asterisk manager api" |
| 15:10.07 | Madkiss | doesn't look what i'm looking for. I'm looking for "asterisk -rx 'core uptime show'", but not for uptime, but for real time mode. |
| 15:10.40 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:12.29 | WIMPy | Something like 'module show like realtime'? |
| 15:13.58 | Madkiss | well, if astcanary is not running, asterisk will switch into "normal" mode even though the realtime module is loaded, won't it? |
| 15:14.39 | WIMPy | NFI. But the use count should tell you. |
| 15:14.47 | eppigy | lenas on Katty |
| 15:14.53 | eppigy | leans on Katty |
| 15:18.41 | *** join/#asterisk jkroon (~jkroon@197.173.4.45) |
| 15:20.43 | *** part/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:21.16 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:21.18 | *** join/#asterisk garymc (~chatzilla@host86-176-88-100.range86-176.btcentralplus.com) |
| 15:24.09 | r0m|u | WIMPy, can you give me a quick hand on an issue I am having with cid passing? |
| 15:24.31 | WIMPy | What issue? |
| 15:24.37 | r0m|u | han_sip.c:24331 handle_request_register: Registration from '<sip:12817043522@10.30.2.53>' failed for '10.30.2.57:5065' - No matching peer found |
| 15:24.49 | r0m|u | my gsm gateway is passing cid to my asterisk |
| 15:24.57 | r0m|u | but asterisk is denying it |
| 15:25.27 | WIMPy | Where's the link between registration and CID? |
| 15:25.53 | WIMPy | 1. don't set callerid for the peer and 2. you may need trustrpid. |
| 15:25.58 | r0m|u | the instructions for the gateway say not to register to asterisk if you are passing cid |
| 15:26.10 | WIMPy | Then the CID should pass. |
| 15:26.38 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 15:26.48 | WIMPy | Do you know the IP of that gateway or is it dynamic? |
| 15:27.05 | r0m|u | I know it. its static and is set static in the sip.conf |
| 15:27.41 | r0m|u | host=lanip |
| 15:27.43 | WIMPy | Then there's no need to register. |
| 15:27.52 | Katty | i read that as host=lamp |
| 15:28.08 | r0m|u | ok so set trustrpid=yes and that should help |
| 15:28.10 | WIMPy | prefers lamb |
| 15:28.18 | nunne | When using attendedtransfer function.. and the first user hangsup before the call transfer hangsup (to mimic the behaviour of blindtransfer).. the first user doesnt enter h (hangup extension) until the call transfer has been answered.. is there anyway around this? |
| 15:28.31 | Katty | never had lamb before. |
| 15:28.36 | Katty | it's spensive. |
| 15:28.36 | WIMPy | I heven't seen what the issue is, yet. |
| 15:28.53 | r0m|u | lamb is good :) |
| 15:29.13 | nunne | and im using asterisk 1.4 :P |
| 15:29.22 | WIMPy | Katty: Get someont to invite you :-) |
| 15:29.40 | Katty | invite me? |
| 15:30.07 | Qwell | Katty: never had lamb? O.o |
| 15:30.07 | Kobaz | nunne: i have a patch i'm working on called hangup handlers, which will run a hangup dialplan for you on xfer |
| 15:30.12 | WIMPy | For some lamb. |
| 15:30.20 | Qwell | Go to any greek place and get a gyro. |
| 15:31.42 | Katty | gyros are yummy |
| 15:31.46 | Qwell | yummier with lamb |
| 15:31.47 | Katty | i guess i have had lamb then |
| 15:32.04 | Qwell | most places do lamb by default. It's possible you haven't though. |
| 15:32.17 | Qwell | s/most/good/ |
| 15:32.51 | Qwell | I went to a new greek place the other day that was good. I don't like cucumbers, and the sauce they used wasn't all super cucumbery. |
| 15:33.32 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:33.38 | Katty | wondderrr boyyyy |
| 15:33.44 | Katty | what is the secret of your power |
| 15:35.18 | The_Boy_Wonder | BECAUSE I EAT LAMB |
| 15:35.29 | Qwell | wat |
| 15:35.39 | Qwell | how do you have context?! |
| 15:35.39 | The_Boy_Wonder | ?! |
| 15:35.43 | Qwell | You weren't here. |
| 15:35.44 | The_Boy_Wonder | context of what |
| 15:35.48 | The_Boy_Wonder | how do you know |
| 15:35.48 | WIMPy | LOL |
| 15:35.58 | r0m|u | WIMPy, I cant get it to accept the damn calls it keeps telling me that "No Matching peer found" |
| 15:36.03 | nunne | Kobaz: that sounds excellent! which version is it for? Im using Asterisk 1.4 |
| 15:36.05 | Qwell | The_Boy_Wonder: was that seriously just a random comment? |
| 15:36.11 | The_Boy_Wonder | yup |
| 15:36.15 | Katty | creepy. |
| 15:36.20 | Qwell | we were *just* talking about lamb,. |
| 15:36.23 | Katty | we were totally just talking about lamb before you walked in |
| 15:36.55 | WIMPy | r0m|u: That's neither related to registering nor CID. |
| 15:37.00 | The_Boy_Wonder | WAT?! |
| 15:37.39 | WIMPy | r0m|u: You have a type=peer with the GWs IP? |
| 15:37.45 | r0m|u | Yes |
| 15:37.51 | Kobaz | nunne: 1.8/trunk |
| 15:38.25 | Qwell | The_Boy_Wonder: http://pastebin.com/TBX7aet6 |
| 15:39.07 | The_Boy_Wonder | wtf? |
| 15:39.07 | nunne | Kobaz: I hope I will be able to change your patch to 1.4 :) Running embedded system here |
| 15:39.33 | *** join/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 15:39.45 | r0m|u | WIMPy, http://pastebin.com/EF4WRTXd |
| 15:40.10 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:40.53 | [TK]D-Fender | r0m|u, allowguest=yes <-- No. This is a [general] setting only. insecure=port,invite |
| 15:40.53 | WIMPy | r0m|u: allowguest is only valid under [general]. |
| 15:41.52 | r0m|u | crap.... :( than this might not work.... I am not turning on guest for all peers :( |
| 15:42.15 | jaytee | all this talk of lamb makes me hungry for lamb vindaloo |
| 15:42.23 | Kobaz | nunne: it's in some shambles at the moment, i just added hangup support for masq'd channels yesterday |
| 15:42.24 | WIMPy | The's exactely what it's good for. And only that. |
| 15:42.33 | jaytee | with mango chutney and lime pickle |
| 15:42.47 | Kobaz | and pine mango? |
| 15:42.58 | Kobaz | throw in pinefrog too |
| 15:43.56 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 15:44.08 | Katty | i want some mango chutney |
| 15:44.18 | Qwell | I want gyros. :( |
| 15:45.23 | [TK]D-Fender | <r0m|u> crap.... :( than this might not work.... I am not turning on guest for all peers :( <- NO. I gave you a peer specific setting |
| 15:45.30 | [TK]D-Fender | r0m|u, to replace the one you had that was wrong |
| 15:45.56 | r0m|u | Yes I just try that and still rejecting it with no matching peer :( |
| 15:47.43 | nunne | Kobaz: Okay, I guess i will have to tell customers just to use attendedtransfer for real attended transfers and then solve blindtrasnfer using ${BLINDTRANSFER} in the mean time :) |
| 15:49.08 | *** join/#asterisk tehrcon (~tercon@mail.imprinters.com) |
| 15:49.49 | r0m|u | [TK]D-Fender, here is my context http://pastebin.com/JgLzxyvr |
| 15:50.20 | [TK]D-Fender | r0m|u, I don';t see a new call.. |
| 15:50.55 | r0m|u | [TK]D-Fender, what do you mean? |
| 15:51.01 | [TK]D-Fender | lok at the call |
| 15:51.42 | r0m|u | sip:1NXXNXXXXXX? |
| 15:52.10 | [TK]D-Fender | r0m|u, no that is dialplan. |
| 15:52.15 | [TK]D-Fender | r0m|u, I said to look at the call |
| 15:52.23 | [TK]D-Fender | And doesn't even loko valid |
| 15:52.36 | WIMPy | We don't get to the dialplan with "no matching peer". |
| 15:52.40 | [TK]D-Fender | not that it matters yet... |
| 15:56.00 | r0m|u | No matter what I do I get no matching peer. and like WIMPy pointed out without no matching peer I cant get to the dial plan. |
| 15:56.34 | WIMPy | Have you tried to restart Asterisk? |
| 15:56.51 | [TK]D-Fender | r0m|u, Where is the call? |
| 15:57.16 | r0m|u | [TK]D-Fender, [Nov 10 09:56:50] NOTICE[2944]: chan_sip.c:24331 handle_request_register: Registration from '<sip:mycid@10.30.2.53>' failed for '10.30.2.57:5065' - No matching peer found |
| 15:57.33 | r0m|u | Thats all I am getting :( |
| 15:57.36 | [TK]D-Fender | r0m|u, that a registration, not a call |
| 15:57.51 | r0m|u | you want me to trun on sip debuging |
| 15:57.51 | WIMPy | registering has nothing to do with incomming calls. |
| 15:58.03 | WIMPy | And as I pointed out, there's no need to register. |
| 15:58.17 | [TK]D-Fender | r0m|u, actually that is part of your problem |
| 15:58.17 | WIMPy | Disable it if you can. |
| 15:58.29 | [TK]D-Fender | r0m|u, it is trying to register and you put a fixed host <- |
| 15:58.29 | r0m|u | WIMPy, I understand but I am not sure why that is coming up! |
| 15:58.34 | WIMPy | It's not supposed to work if you specify a host. |
| 15:58.38 | [TK]D-Fender | r0m|u, You PUT A HOST. |
| 15:58.48 | [TK]D-Fender | Registration is going to be REFUSED. Every single time |
| 15:58.48 | WIMPy | Because yu register, but you can't. |
| 15:59.06 | [TK]D-Fender | r0m|u, "host=dynamic" if the other side is expecting to register |
| 15:59.48 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:00.22 | r0m|u | [TK]D-Fender, the other side should not try to register because is not activated to register. let me try dynamic and see |
| 16:00.44 | [TK]D-Fender | <r0m|u> [TK]D-Fender, [Nov 10 09:56:50] NOTICE[2944]: chan_sip.c:24331 handle_request_register: Registration from '<sip:mycid@10.30.2.53>' failed for '10.30.2.57:5065' - No matching peer found <------- registeation. |
| 16:00.53 | [TK]D-Fender | registration* |
| 16:01.22 | [TK]D-Fender | "not activated to register". I see a giant sign saying "Oh yes I am..." |
| 16:02.50 | r0m|u | [TK]D-Fender, let me check the gateway settings again. |
| 16:04.19 | Kobaz | nunne: yeah, using attended transfers to blind transfers is a training issue |
| 16:07.28 | r0m|u | [TK]D-Fender, its tring to register and I am not sure why. Its all dible |
| 16:07.31 | r0m|u | disable |
| 16:08.24 | r0m|u | I am going to reset it and see |
| 16:15.58 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 16:15.59 | joelio | Hi, I'm wondering what my options are for video transcoding in partnership with Asterisk? |
| 16:16.16 | WIMPy | None so far. |
| 16:17.47 | irroot | joelio you welcome to do it :P |
| 16:17.53 | Stese | Is anyone in the UK using Voiceflex as a SIP provider? |
| 16:18.14 | joelio | WIMPy: really? I've found a project called diastar, but as much as it touts open source, it's commercially licensed with closed source components. Looking for somthing properly open so we can hack in HD video transcoding etc. |
| 16:18.50 | joelio | irroot: I thnk it may come to that! We've already added hd video to linphone and written a v4l2loopback adapter to take hdmi feeds |
| 16:18.56 | joelio | just looking at transcoding now |
| 16:19.59 | r0m|u | no go so I give up. |
| 16:20.26 | r0m|u | Thanks for the help guys. the device no matter what wants to register maybe it was never meant for this purpose |
| 16:20.49 | r0m|u | or just has bugs up the ass |
| 16:21.18 | WIMPy | Or the interface is that shitty that you'd never guess how to get it to do what you want. |
| 16:21.23 | p3nguin | Allow it to register. Make calls. Be happy. |
| 16:22.58 | [TK]D-Fender | lr0So let it register |
| 16:23.57 | r0m|u | p3nguin, If it register it wont pass caller id. |
| 16:24.34 | r0m|u | but its fine now. Its register and the device is happy. I am not though lol |
| 16:25.30 | p3nguin | Registering has nil to do with the ability to do anything with callerid. |
| 16:25.46 | r0m|u | p3nguin, with this devices it does. |
| 16:25.52 | p3nguin | I doubt it. |
| 16:26.27 | r0m|u | p3nguin, http://www.portech.com.tw/data/MV-370%20user%20manual.pdf |
| 16:26.33 | r0m|u | page 21 of the pdf |
| 16:26.48 | r0m|u | to pass caller id the device can not be register |
| 16:26.56 | Qwell | wat |
| 16:27.10 | p3nguin | That's the most ridiculous thing I've heard all day. |
| 16:27.31 | r0m|u | actually page 16 I think |
| 16:28.03 | Qwell | wow |
| 16:28.09 | p3nguin | I was about to tell you that 21 is SMS Reader. |
| 16:28.31 | Qwell | page 20 says page 16 on it |
| 16:29.01 | p3nguin | That's not uncommon with repacked manuals. |
| 16:29.24 | r0m|u | Yeap |
| 16:29.26 | Qwell | TOC |
| 16:29.35 | Qwell | pdfs don't care |
| 16:29.38 | r0m|u | 20 on the pdf app and 16 on the browser lol |
| 16:29.53 | p3nguin | It says you should use Tel/User if you need to register to asterisk. |
| 16:30.07 | r0m|u | so any who its just broken :) |
| 16:30.17 | r0m|u | Yes keep reading |
| 16:30.20 | [TK]D-Fender | <r0m|u> p3nguin, If it register it wont pass caller id. <- show us |
| 16:30.25 | r0m|u | look at passing caller ID |
| 16:30.32 | p3nguin | I'm reading. |
| 16:30.38 | r0m|u | [TK]D-Fender, sure one sec let me test |
| 16:32.20 | nfi|ermes | i can't build zaphfc module in Centos 5.7 (2.6.18-274.7.1.el5) |
| 16:32.37 | p3nguin | They apparently don't know how to read their own manual and put accurate page numbers on anything... |
| 16:33.09 | nfi|ermes | i also tried to build dahdi-hfcs , but dahdi keep asking for zaphfc module |
| 16:33.18 | p3nguin | It says to see page 42 to set up asterisk to accept the callerID, but page 42 has nothing of the likes. |
| 16:34.33 | r0m|u | [TK]D-Fender, chan_sip.c:22081 handle_request_invite: Call from '1002' (10.30.2.57:5065) to extension '1006' rejected because extension not found in context 'inbound-gsm'. |
| 16:34.48 | r0m|u | and thats because is not understanding the context to jump to disa |
| 16:34.49 | r0m|u | exten => _sip:1NXXNXXXXXX,1,GotoIf($["${CALLERID(num):4}" = "12817043522"]?disa,1) |
| 16:35.14 | [TK]D-Fender | r0m|u, Every time I ask you to lok I expect a complete pastebin, not jsut one-lie snippits. And that is a clear dialplan error and doesn't prove chat callerID is coming in |
| 16:35.18 | p3nguin | That says go to extension "disa" priority "1" |
| 16:35.24 | r0m|u | Its passsig the ext and not the cid |
| 16:35.32 | [TK]D-Fender | extension '1006' <------------ |
| 16:35.35 | [TK]D-Fender | r0m|u, No |
| 16:35.44 | r0m|u | ok one sec let me pb |
| 16:35.53 | [TK]D-Fender | r0m|u, We don't see what it's passing because we can't see all of the SIP headers. |
| 16:36.01 | [TK]D-Fender | r0m|u, You aren't showing us the complete call. |
| 16:36.21 | [TK]D-Fender | <[TK]D-Fender> extension '1006' <------------ <- nothing to do with CallerID |
| 16:37.01 | [TK]D-Fender | "sip:" <- what tells you that the call should come in looking like that? |
| 16:37.46 | r0m|u | one sec |
| 16:38.44 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001550.dsl.bell.ca) |
| 16:38.52 | r0m|u | [TK]D-Fender, http://pastebin.com/Ybd9G83V |
| 16:39.37 | *** join/#asterisk afink (~afink@wsip-184-187-15-226.om.om.cox.net) |
| 16:40.01 | p3nguin | Why do you have callcentric set to nat=yes? They are not behind nat. |
| 16:41.31 | r0m|u | [TK]D-Fender, http://pastebin.com/Gw7LYk94 |
| 16:41.41 | r0m|u | p3nguin, probably a mistake |
| 16:41.49 | r0m|u | I thought I had change them long time a go |
| 16:42.11 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 16:42.14 | r0m|u | p3nguin, nat=no |
| 16:42.21 | r0m|u | I am not sure why is sending nat=yes |
| 16:42.32 | r0m|u | I have it set to nat=no |
| 16:42.38 | r0m|u | in the peer settings |
| 16:43.16 | p3nguin | Is _sip:1NXXNXXXXXX even a valid extension pattern? I've never seen anyone do this before. |
| 16:43.26 | [TK]D-Fender | No |
| 16:43.33 | [TK]D-Fender | r0m|u, From: "12817043522" <sip:1002@10.30.2.53>;tag=5fdcdb5c <-- loks like callerid to me |
| 16:43.38 | p3nguin | I guess that's why I've never seen anyone do it before. |
| 16:44.16 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-124.mobile.uci.edu) |
| 16:44.59 | r0m|u | [TK]D-Fender, well fuck. The Taiwanese manual is wrong. |
| 16:45.24 | r0m|u | p3nguin, I did that because before it was sending sip:cid as is doing now with ext |
| 16:45.54 | [TK]D-Fender | r0m|u, Real problem is you say it's not there and we see it. What you believe constitutes "fact" does not appear to. |
| 16:46.29 | r0m|u | Sr Yes Sr. I stand correct in my place! |
| 16:46.45 | p3nguin | Unless it said Call from XYZ to extension 'sip:whatever' rejected because extension not found, it wasn't really doing that. |
| 16:47.22 | p3nguin | And I have NEVER ever seen that as an extension. |
| 16:47.28 | p3nguin | Ever. |
| 16:47.30 | r0m|u | p3nguin, what was that word that you engrave for my saying in to the bot? I want to say it now as things are staring to clear! |
| 16:47.39 | [TK]D-Fender | He seemed to believe that they would dial the callerID as the extension. Nobody does that. |
| 16:47.40 | p3nguin | ~motorboat |
| 16:47.41 | infobot | <r0m|u> its all clear now. like boobs slaping you in the face! |
| 16:47.58 | r0m|u | yes yes it is clear now [TK]D-Fender ! |
| 16:48.02 | r0m|u | :) |
| 16:48.11 | r0m|u | like the bot said :) |
| 16:48.33 | r0m|u | [TK]D-Fender, I guess I belived the manual :( |
| 16:48.47 | r0m|u | WHich was broken to begging with any ways :) |
| 16:49.15 | blitzrage | actually that _sip:1NXXNXXXXXX should work |
| 16:49.24 | [TK]D-Fender | Telephony is not faith-based. Look. Always look. And then show. |
| 16:49.43 | blitzrage | I mean, it would work technically, not necessary in the scenario you're describing |
| 16:49.47 | r0m|u | I seen you said that so many times [TK]D-Fender lol it should be on the boot |
| 16:49.57 | p3nguin | Even if it will match, I've never ever seen asterisk receive that as an extension. |
| 16:50.01 | r0m|u | Thanks for the info blitzrage |
| 16:50.02 | blitzrage | p3nguin: right :) |
| 16:50.28 | blitzrage | I mean, _yousuckNXXNXXXXXX is valid too :) |
| 16:50.44 | p3nguin | I guess your phone would have to make a call to sip:sip:somenumber |
| 16:50.51 | [TK]D-Fender | "technically possible" vs "This would be the first time for anyone ever to receive this, why are you so special?" |
| 16:51.01 | blitzrage | p3nguin: ya, and who the heck knows how that would even end up being parsed before it got to the dialplan |
| 16:51.10 | blitzrage | [TK]D-Fender: +1 |
| 16:51.13 | r0m|u | p3nguin, There is a first time for everything and here is yours handle_request_register: Registration from '<sip:12817043522@10.30.2.53>' failed for '10.30.2.57:5065' - No matching peer found :) |
| 16:51.37 | blitzrage | that has nothing to do with pattern matching |
| 16:51.41 | Qwell | r0m|u: What version is this, exactly? |
| 16:51.42 | p3nguin | I'm not sure what you're trying to saw there. |
| 16:51.52 | p3nguin | say |
| 16:52.05 | p3nguin | So there's no peer for that device. Big deal. Make one. |
| 16:52.22 | blitzrage | ya, man up! |
| 16:52.32 | p3nguin | The SIP port is wrong, too, which could have something to do with it. |
| 16:52.37 | r0m|u | Qwell, 1.8.7.1 |
| 16:52.50 | r0m|u | lol |
| 16:52.55 | r0m|u | I guess nothing lol |
| 16:53.01 | p3nguin | Make a peer for the device. Have the device register. Make calls. Enjoy. |
| 16:53.13 | r0m|u | :) |
| 16:54.32 | r0m|u | ok well its working now. though the dial plan is working |
| 16:54.37 | r0m|u | is not* |
| 16:54.52 | p3nguin | Dial plan is the easy part. |
| 16:55.01 | p3nguin | Make an extension that matches what you're trying to call. |
| 16:55.05 | [TK]D-Fender | r0m|u, It was looking for 1006 and told yuo very clearly. Go make it |
| 16:55.25 | r0m|u | if 1NXXN go to disa |
| 16:55.33 | r0m|u | in disa I should be able to dial ext |
| 16:55.34 | p3nguin | I never saw that. |
| 16:55.52 | p3nguin | I saw _sip:NXXNXXXXXX |
| 16:55.53 | r0m|u | p3nguin, I took the sip: out |
| 16:55.54 | p3nguin | which is crap. |
| 16:56.04 | [TK]D-Fender | <r0m|u> if 1NXXN go to disa <- won't match 1006 |
| 16:56.14 | p3nguin | If you dial 1006, it will not match _1NXXNXXXXXX |
| 16:56.33 | p3nguin | Do you know what N and X mean there? |
| 16:56.59 | [TK]D-Fender | Doesn't even matter. 1006 = short, 1nxx..... blah = long |
| 16:57.16 | [TK]D-Fender | Someone really isn't awak here yet... |
| 16:57.18 | r0m|u | ok ok hold on. You guys are not following what I am trying to do. If I call the gsm the gsm should send me to asterisk and from asterisk it should execute the dial plan |
| 16:57.22 | p3nguin | I have a feeling he's only using those patterns because something suggested it was the thing to do. |
| 16:57.26 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:57.31 | p3nguin | Never bothering to understand what it means. |
| 16:57.58 | r0m|u | when I am sent to asterisk as [TK]D-Fender pointed out 1281 is been passed to asterisk |
| 16:58.01 | p3nguin | If I'm wrong, prove it. |
| 16:58.22 | [TK]D-Fender | <r0m|u> [TK]D-Fender, chan_sip.c:22081 handle_request_invite: Call from '1002' (10.30.2.57:5065) to extension '1006' rejected because extension not found in context 'inbound-gsm'. |
| 16:58.28 | [TK]D-Fender | Looking for 1006 <--------- |
| 16:58.31 | [TK]D-Fender | 4 DIGITS |
| 16:58.32 | r0m|u | so when 1281 comes in it shoud do that no? exten => _1NXXNXXXXXX,1,GotoIf($["${CALLERID(num):4}" = "mycid"]?disa,1) |
| 16:58.37 | p3nguin | "to extension '1006'" |
| 16:58.40 | [TK]D-Fender | 100^ <- |
| 16:58.43 | [TK]D-Fender | 1006 |
| 16:59.01 | [TK]D-Fender | _1NXXNXXXXXX <=- NO. Does this look lie will match 4 digits? It won't |
| 16:59.08 | p3nguin | _1NXXNXXXXXX does not match 1006 |
| 16:59.12 | [TK]D-Fender | _1NXXNXXXXXX <=-this has ntohing to do with callerid |
| 16:59.14 | p3nguin | 1006 does not match _1NXXNXXXXXX |
| 16:59.25 | r0m|u | [TK]D-Fender, ok so its clear. the gsm cant do what I wanted to do. |
| 16:59.26 | p3nguin | _1NXXNXXXXXX is the extension being called. |
| 16:59.31 | [TK]D-Fender | r0m|u, NO] |
| 16:59.34 | p3nguin | I'd bet it can. |
| 16:59.40 | p3nguin | The problem is user error. |
| 17:00.14 | r0m|u | one sec making a change |
| 17:00.21 | [TK]D-Fender | You don't have an exten to match 1006. |
| 17:00.48 | [TK]D-Fender | <PROTECTED> |
| 17:01.01 | p3nguin | I wonder if making an extension for 1006 ever occurred. |
| 17:01.02 | [TK]D-Fender | <PROTECTED> |
| 17:01.19 | dijib | ive having my phones not ring when Dial'ed and i get this in CLI |
| 17:01.20 | dijib | <PROTECTED> |
| 17:01.38 | Kobaz | so |
| 17:01.42 | Kobaz | i have a blf/hints problem |
| 17:01.44 | p3nguin | Turn off DND or call forwarding on the phone. |
| 17:01.44 | Kobaz | <PROTECTED> |
| 17:01.52 | Kobaz | <PROTECTED> |
| 17:01.53 | dijib | i was testing the extensions the other day by calling and then not picking up.. then they stopped ringing |
| 17:02.02 | Kobaz | how do i give it a kick? |
| 17:02.12 | Kobaz | why would it show up as idle? |
| 17:02.21 | [TK]D-Fender | Kobaz, show us [2621] 's conf |
| 17:04.00 | Kobaz | http://pastebin.com/g1K5jJcA |
| 17:04.17 | Kobaz | okay, i narrowed down the problem |
| 17:04.29 | Kobaz | with SIP/2621 calls some other device, it still shows idle |
| 17:04.45 | Kobaz | but if from another device i send a call TO SIP/2621.. then it shows as ringing and then inuse |
| 17:04.49 | dijib | p3nguin, i dont believe i have them enabled |
| 17:05.49 | [TK]D-Fender | <[TK]D-Fender> Telephony is not faith-based. Look. Always look. And then show. |
| 17:06.04 | [TK]D-Fender | You're right. This should be in the bot... |
| 17:06.25 | r0m|u | :) |
| 17:06.38 | Kobaz | [TK]D-Fender: so 'outbound' calls do not update the hint device status status |
| 17:06.43 | Kobaz | [TK]D-Fender: but 'inbound' calls do |
| 17:06.53 | [TK]D-Fender | Kobaz, It should update. Realtime peer? |
| 17:06.59 | Kobaz | static realtime |
| 17:07.02 | [TK]D-Fender | Kobaz, what ver? |
| 17:07.04 | r0m|u | p3nguin, can you msg me? |
| 17:07.08 | Kobaz | 1.something lemme see |
| 17:07.10 | Kobaz | might be old |
| 17:07.21 | Kobaz | 1.6.0.26 |
| 17:07.24 | [TK]D-Fender | 1.something = any version from the past 8 years |
| 17:07.26 | Kobaz | hmm, i should bump this to 1.8 |
| 17:07.35 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:07.43 | [TK]D-Fender | Kobaz, type=friend <--- do it |
| 17:08.16 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 17:08.30 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:08.42 | Kobaz | it's in there |
| 17:08.48 | [TK]D-Fender | is now? |
| 17:08.51 | [TK]D-Fender | wasnt' before |
| 17:08.58 | [TK]D-Fender | be clear... |
| 17:09.25 | Kobaz | what? it's right in the pastebin |
| 17:09.33 | Kobaz | it's the second line |
| 17:09.40 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:09.44 | [TK]D-Fender | Kobaz, No it's not |
| 17:10.09 | Kobaz | yeah it is |
| 17:10.19 | [TK]D-Fender | peer |
| 17:10.21 | [TK]D-Fender | rather |
| 17:10.28 | [TK]D-Fender | dangnammit |
| 17:10.32 | Kobaz | type | friend |
| 17:10.37 | Kobaz | right there, second line |
| 17:10.37 | [TK]D-Fender | Kobaz, change to peer |
| 17:10.40 | Kobaz | oh |
| 17:10.42 | [TK]D-Fender | Kobaz, My bad... |
| 17:10.44 | [TK]D-Fender | typed too fast |
| 17:11.01 | Kobaz | i've never had problems with blf using type friend |
| 17:11.15 | [TK]D-Fender | Kobaz, its chronic in my experience |
| 17:11.17 | Kobaz | i'm switching to 1.8 and i'll see if theres problems |
| 17:11.19 | Kobaz | hmm |
| 17:11.24 | [TK]D-Fender | Kobaz, just test peer in place. |
| 17:11.36 | [TK]D-Fender | "friend" is really deal for the most-part... has been since 1.4 |
| 17:12.02 | Kobaz | deal? |
| 17:12.05 | Kobaz | dead? |
| 17:12.36 | [TK]D-Fender | dead* |
| 17:12.37 | [TK]D-Fender | gah |
| 17:12.48 | [TK]D-Fender | This has been a terrible day so far... |
| 17:12.51 | Kobaz | the l isn't even anywhere near the d |
| 17:12.54 | Kobaz | :P |
| 17:13.20 | [TK]D-Fender | Kobaz, Yeah, I'm not claiming it was a keyboard accident... |
| 17:13.21 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 17:16.25 | Kobaz | okay |
| 17:16.29 | Kobaz | upgraded to 1.8 and it works |
| 17:16.37 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:16.55 | Kobaz | but i'll try with type=peer for new systems |
| 17:17.20 | dijib | p3nguin, ive de-activated call forwarding and DND on the ATA. still does the busy/congested |
| 17:19.45 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 17:20.31 | Kobaz | allrightey |
| 17:20.34 | Kobaz | so.. more blf stuff |
| 17:20.45 | Kobaz | i'm trying the better-style blf on a polycom with a sidecar |
| 17:21.00 | [TK]D-Fender | dijib, Still the device's fault |
| 17:21.00 | Kobaz | attendant.resourceList.1.address="sip:2621@192.168.50.1" attendant.resourceList.1.label="Sam 2621" |
| 17:21.05 | Kobaz | using that type of syntax... |
| 17:21.41 | Kobaz | rather than using directory entries. So now the problem is when the attendent phone gets a call, the screen says: "call from: sip:2650@192.168.50.1.....etc" |
| 17:21.43 | [TK]D-Fender | dijib, Phone left off hook. On a call and no call-waiting enabled, etc |
| 17:21.57 | Kobaz | anyone know how to make the polycom display just a name and number, not the sip address? |
| 17:22.26 | [TK]D-Fender | Kobaz, Normally only happens whent he domain on the end doesn't amtch what the phone uses to register to (cross LAN segments, etc) |
| 17:22.54 | Kobaz | it only shows sip:... when getting a call to a monitored device |
| 17:24.14 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 17:24.19 | Kobaz | calls to 2630 direct (the phone that has the sidecar), display regular callerid |
| 17:27.52 | Kobaz | i'll see if i can rewrite the sip header in chan_sip to make the polycom screen display nicely |
| 17:28.03 | Kobaz | the current behavior is just really stupid |
| 17:28.56 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:28.58 | *** join/#asterisk Chaz6 (~chaz@2001:470:1f09:11d4:38::1) |
| 17:30.19 | Kobaz | here's the bit |
| 17:30.20 | Kobaz | <?xml version="1.0"?><dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="5" state="full" entity="sip:2621@192.168.50.1"><dialog id="2621" direction="recipient"> |
| 17:30.46 | Kobaz | there's a sip notify sent out with that in the body, to the sidecar phone |
| 17:31.05 | Kobaz | so if i change that in chan_sip to just show the user part... |
| 17:35.19 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:36.09 | dijib | phone is on hook. no call currently. im thinking from me calling and not answering is switched on DND or something but deactivate DND doesnt seem to change anything |
| 17:38.40 | *** join/#asterisk jkroon (~jkroon@dsl-241-253-176.telkomadsl.co.za) |
| 17:40.03 | *** join/#asterisk garymc (~chatzilla@host86-176-88-100.range86-176.btcentralplus.com) |
| 17:50.11 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 17:50.15 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 17:53.23 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:53.24 | *** join/#asterisk joshaidan (~brianj@216-211-66-197.block.tbaytel.net) |
| 17:56.46 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:05.36 | [TK]D-Fender | dijib, What phone, and how are you trying to deactivate? |
| 18:06.31 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 18:07.50 | dijib | scoop |
| 18:07.51 | dijib | http://i.imgur.com/Yz2jb.jpg |
| 18:07.59 | dijib | using the ATA's feature codes |
| 18:08.14 | dijib | ive used all the pertinenct Deact features |
| 18:15.03 | saisoma | hey guys, I have a config question with my SIP trunks. I know it's something simple, but I'm stumped. |
| 18:15.29 | saisoma | two asterisk servers, one originates a call to the second, the second sends the call back to the first for vm. I'm getting errors. More details: http://pastebin.com/3fpu5jFq |
| 18:16.08 | *** join/#asterisk BlackBishop (~dexter@ipv6.d3xt3r01.tk) |
| 18:19.51 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 18:20.52 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:24.55 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 18:26.53 | [TK]D-Fender | dijib, Make sure that when you use the feature that it isn't passing it on as a call to * |
| 18:27.23 | dijib | asterisk doesnt do anything when i enter them so its all called in the pap2t-na but i still cant get it working |
| 18:28.08 | [TK]D-Fender | dijib, Look with SIP DEBUG enabled to be sure. |
| 18:28.11 | [TK]D-Fender | ^^^ |
| 18:32.07 | *** join/#asterisk r0t0r00t3r (~www.acq.o@66-214-250-113.dhcp.gldl.ca.charter.com) |
| 18:38.43 | *** join/#asterisk garymc (~chatzilla@host31-53-157-167.range31-53.btcentralplus.com) |
| 18:42.34 | *** join/#asterisk umay (~chris@67-6-158-37.hlrn.qwest.net) |
| 18:43.04 | *** join/#asterisk joshaidan_ (~brianj@216-26-202-160.dynamic.tbaytel.net) |
| 18:43.07 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 18:43.36 | ruben23 | hi guys..anyone have link for howto or any idea on impelementing fail2ban on asterisk |
| 18:44.07 | umay | current fail2ban wiki page a little out of synch for 1.8 |
| 18:44.24 | umay | but a good start |
| 18:44.46 | umay | need to update patterns to include port after <HOST> |
| 18:46.59 | blitzrage | umay: ya run into that a few times. Luckily it's a wiki, and you can help keep it up to date. |
| 18:48.29 | umay | yeah i suppose the 5 minutes to set up yet another login wouldnt kill me |
| 18:50.48 | Kobaz | this polycom blf attendant phone stuff is so retarded |
| 18:51.34 | [TK]D-Fender | Kobaz, The standard presence feature seems to fine for me... |
| 18:52.39 | dijib | [TK]D-Fender, 100% certain the ATA is catching the feature |
| 18:52.44 | Kobaz | well it doesn't show you proper callerid when a call is ringing a monitored extension |
| 18:53.26 | Kobaz | [TK]D-Fender: is the presence you're using is different than attendant uri? |
| 18:53.29 | *** join/#asterisk jkroon (~jkroon@dsl-241-253-176.telkomadsl.co.za) |
| 18:53.47 | *** join/#asterisk r0t0r00t3r (~www.acq.o@66-214-250-113.dhcp.gldl.ca.charter.com) |
| 18:54.53 | r0t0r00t3r | has anyone played around with asterisk & sms? |
| 18:54.56 | Kobaz | [TK]D-Fender: do you have a sample config of the setup you're using |
| 18:56.26 | *** join/#asterisk zurdo13 (~Jose@201.130.0.10) |
| 18:57.58 | [TK]D-Fender | Kobaz, standard directory w/ buddy-watch |
| 18:58.17 | *** join/#asterisk pietro (~pietro@88-149-224-154.dynamic.ngi.it) |
| 18:58.23 | r0t0r00t3r | robaz, it's php... not an asterisk config |
| 18:58.41 | BlackBishop | whish I could write extensions.conf in php |
| 18:58.42 | BlackBishop | :) |
| 18:58.47 | p3nguin | umay: Are you saying change '<HOST>' to say '<HOST>:5060' or something else? |
| 18:59.05 | umay | Kobaz: i think theres a setting i am missing in polycom config file for "legacy" buddy option, but this patch is what i am currently using to get it to work (plus get a Pickup soft key lit) http://pastebin.com/Z6GV2vZn |
| 18:59.55 | Kobaz | [TK]D-Fender: yeah the problem is with the buddy watch, you don't know who is calling who |
| 19:00.00 | *** join/#asterisk jkroon (~jkroon@dsl-241-253-176.telkomadsl.co.za) |
| 19:00.00 | umay | p3nguin: '<HOST>:.*' |
| 19:00.02 | Kobaz | there's no callerid presented on the screen |
| 19:00.16 | Kobaz | yeah pickup is broken too, by default |
| 19:00.32 | p3nguin | umay: Oh. Why do we need to use that instead of the explicit port number? |
| 19:02.21 | r0t0r00t3r | http://pastebin.com/PybacnmM here's the code |
| 19:02.51 | blitzrage | BlackBishop: you can if your extensions.conf just does a FastAGI() call |
| 19:03.22 | Kobaz | umay: ah that looks like a good patch |
| 19:03.35 | Kobaz | umay: that looks like it might fix my problem |
| 19:03.46 | p3nguin | umay: And is the filter.d/asterisk.conf the only file that needs to be updated? |
| 19:03.51 | umay | yeah but how could it be needed ? thats what i keep thinking -- must be missing something else in polly configs |
| 19:04.23 | umay | i mean how is picking up a watched buddies call not working ? when this is something key systems have done for XX years |
| 19:04.44 | p3nguin | Nevermind on the question regarding the port number. I see how some devices don't use the SIP port. |
| 19:04.48 | umay | p3nguin: yeah heres mine http://pastebin.com/B5FTdYd3 |
| 19:04.49 | r0t0r00t3r | lol @ umay |
| 19:05.28 | Kobaz | umay: asterisk doesn't send the right xml data to the phone |
| 19:05.32 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 19:06.02 | umay | i've had a form of that patch in production for almost a year |
| 19:06.17 | umay | was not a laughing matter during deployment |
| 19:07.03 | p3nguin | umay: That looks pretty much the same as how I editing mine. Thanks. |
| 19:07.35 | r0t0r00t3r | i ifeel your pain, umay... |
| 19:08.05 | p3nguin | I don't have the one NOTICE.* .*: Failed to authenticate user .*@<HOST>.* |
| 19:08.08 | p3nguin | but I'll add it. |
| 19:08.12 | r0m|u | back |
| 19:08.22 | r0m|u | sorry server puked a lun |
| 19:09.57 | p3nguin | You're using .*@<HOST>.* ... but should it not be .*@<HOST>:.* ? |
| 19:10.36 | umay | que ? |
| 19:10.44 | umay | i gots some colons in there |
| 19:10.58 | p3nguin | On the last line of the patterns |
| 19:10.58 | umay | all about the colon in those middle 4 lines |
| 19:11.10 | umay | oh dunno |
| 19:11.18 | p3nguin | For the Failed to authenticate user ... pattern |
| 19:11.19 | r0m|u | que? = what? |
| 19:11.21 | umay | "No matching peer found" was the 99% |
| 19:11.31 | p3nguin | Yeah, I see that a bunch. |
| 19:11.35 | umay | r0mju: si |
| 19:11.48 | r0m|u | llo entiendo :P |
| 19:11.50 | r0m|u | :) |
| 19:12.03 | dijib | typical that i have a conjestion issue |
| 19:12.08 | dijib | the nuub that i am |
| 19:12.08 | zurdo13 | ahora se la tienes que cromar |
| 19:12.25 | r0m|u | calmate :) jajajaja |
| 19:14.33 | *** join/#asterisk rdahlin_1_ (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 19:17.08 | r0m|u | p3nguin, you avail? |
| 19:18.23 | r0t0r00t3r | hey penguin, you the same penguin that used to run a conf? |
| 19:19.01 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:19.07 | *** join/#asterisk becca_r (~IceChat77@72.165.148.230) |
| 19:21.09 | r0t0r00t3r | had anyone coded an exchange scanner (sequential) like the one i have by chance? example: http://www.projectstreamer.com/users/r0t0r00t3r/scan-demo/scan-demo.mp3 |
| 19:21.19 | r0t0r00t3r | would like to collab if anyone has... |
| 19:21.40 | autofsckk | hello, im new to asterisk, yesterday i was testing and it worked at least an echo extension, now i try to connect to it and it hangs bye itself, did a sip debug here http://pastebin.com/MfFmxZKU can somebody help me please to find out what is going on? |
| 19:23.20 | _Corey_ | r0t0r00t3r: What are you trying to accomplish with these sequential calls? |
| 19:23.30 | autofsckk | theres an ip there 187.132.70.12 that i dont have anyidea from whom could it be, i dont understand how to interpret the debug, i saw a hangupcausecode 111 |
| 19:24.23 | _Corey_ | r0t0r00t3r: What you describe sounds a lot like what used to be called a "demon dialer" or something like that... In any case, it's not legal in the USA |
| 19:24.32 | p3nguin | r0m|u: I'm here again. |
| 19:24.42 | r0t0r00t3r | scanning blocks of addresses... method in which to pentest/audit the pstn... |
| 19:24.53 | r0m|u | p3nguin, msg. |
| 19:25.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:25.15 | r0t0r00t3r | it's legal if you have authorization from the company to perform the audit on. |
| 19:25.28 | r0t0r00t3r | furthermore, i really don't need a lecture in ethics. k thx. |
| 19:27.19 | umay | ethics != law |
| 19:28.07 | r0t0r00t3r | can you please cite the law for me? |
| 19:28.15 | *** join/#asterisk joshaidan (~brianj@S0106602ad06b3517.tb.shawcable.net) |
| 19:28.46 | umay | its not possible -- the laws grow by 3,000,000 pages a day -- need hadoop to keep em straight |
| 19:29.58 | _Corey_ | r0t0r00t3r: We're not lawyers... Just check the TCPA, etc. This has definitely been around for a while so you should be able to find some info. |
| 19:30.30 | umay | TCPA is w/o jurisdiction for pure intra-state traffic |
| 19:30.59 | umay | inter-state only |
| 19:31.46 | *** part/#asterisk pietro (~pietro@88-149-224-154.dynamic.ngi.it) |
| 19:32.01 | umay | even then its tenuous - nothing in the constitution about telephone calls |
| 19:32.04 | *** part/#asterisk Chaz6 (~chaz@2001:470:1f09:11d4:38::1) |
| 19:33.30 | r0t0r00t3r | lolz |
| 19:33.42 | autofsckk | help please to interpret a sip debug? http://pastebin.com/MfFmxZKU i found that my call to echo test just shuts befor 1 minute, but trying to see de sip debug i found an ip address that shouldnt be there, this box isnt receiving connections from the outside, could it be still hacked? |
| 19:34.51 | r0t0r00t3r | sorry i brought it up, was just a question. in any event, i won't share any of the more "nefarious" of such caoabilities. |
| 19:35.00 | r0t0r00t3r | nefarious uses* |
| 19:35.10 | r0t0r00t3r | capabilities* |
| 19:35.18 | r0t0r00t3r | can't type for crap today. lol |
| 19:36.31 | *** join/#asterisk leroybuckingham (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 19:37.29 | leroybuckingham | Hey I'm not positive, but I think I'm experiencing a bug with the ajam XML output for all Command actions in version 1.8.6-cert1-rc1 |
| 19:37.39 | autofsckk | Contact: <sip:600@187.132.70.12:5060> that ip isnt from here, and anybody i know |
| 19:37.55 | becca_r | That IP is from Mexico |
| 19:38.02 | leroybuckingham | It seems to be putting an extra single quote inside one of the attributes |
| 19:38.40 | autofsckk | becca_r: yes it is, im in mexico too, but its not from anybody i know, im testing with a spa3102 |
| 19:39.37 | becca_r | then you either have someone connecting to your box, or you've provisioned that IP on one of your devices. |
| 19:40.02 | autofsckk | Via: SIP/2.0/UDP 187.132.70.12:5060;branch=z9hG4bK28d28ec6;rport |
| 19:41.17 | autofsckk | becca_r: can you tell me please how can i see if someone is connecting to my box? |
| 19:41.42 | *** join/#asterisk peep_ (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 19:42.16 | peep_ | any 1.8 AMI experts around? |
| 19:44.13 | peep_ | More specifically Im making the switch over to 1.8 right now and it seems to me that when using the AMI action 'command' asterisk is returning invalid XML |
| 19:44.56 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:45.27 | wdoekes2 | peep_: pb that you get and what you expect |
| 19:45.31 | wdoekes2 | *what you get |
| 19:46.47 | peep_ | wdoekes2: Sure thing, 2 seconds |
| 19:47.22 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 19:50.55 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 19:51.15 | peep_ | wdoekes2: two examples here - http://pastebin.ca/2093638 |
| 19:51.49 | peep_ | wdoekes2: There are seemingly random extra single quotes near the "end command" text |
| 19:52.00 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 19:52.24 | McBoingBo | finally found a decent method for me to take a peak into the call traffic, using tcpdump and wireshark, but for g729 calls, how do I get the same level of information? I thought at first that only playing back the RTP stream would be fuxxored, but the actual analysis seemed out of whack as well, can anyone help? |
| 19:53.26 | wdoekes2 | and your current version is? (and it worked properly with which version?) |
| 19:54.37 | peep_ | wdoekes2: Currently running 1.8.6 , has worked in 1.6 and lower. I mean technically it still works but its just outputting bad XML |
| 19:54.42 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 19:56.52 | wdoekes2 | ok.. I'll take a short peek, see if I can find anything |
| 19:58.33 | *** join/#asterisk afink (~afink@wsip-184-187-15-226.om.om.cox.net) |
| 19:59.13 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 20:01.22 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 20:01.39 | treborsux | where can i print out the * commands like *98 for voicemail ect |
| 20:02.23 | [TK]D-Fender | treborsux, No such thing |
| 20:02.54 | treborsux | suprise no one made one |
| 20:03.37 | [TK]D-Fender | treborsux, No.. there is no such thing as "*98 for voicemail". I looked and ai have no such thing. So what you are asking about... does not exist at all |
| 20:03.52 | [TK]D-Fender | treborsux, Not that there isn't a map ... it's that there is not TO map. |
| 20:03.58 | [TK]D-Fender | nothing* |
| 20:04.08 | *** join/#asterisk eZz (~ez@195.3.244.4) |
| 20:04.24 | treborsux | i hit *98 and my voicemail comes up |
| 20:04.27 | treborsux | on my phone |
| 20:04.38 | treborsux | welcome to mail! |
| 20:04.50 | [TK]D-Fender | treborsux, It's your dialplan.. YOU map it... |
| 20:04.51 | treborsux | another * command shuts off call waiting ect |
| 20:05.01 | treborsux | i know but there is a default |
| 20:05.07 | [TK]D-Fender | no there isn't |
| 20:05.15 | treborsux | for freepbx |
| 20:05.19 | treborsux | sorry wrong channel |
| 20:05.22 | [TK]D-Fender | Don't ask HERE then |
| 20:05.39 | [TK]D-Fender | :p |
| 20:08.07 | autofsckk | http://pastebin.com/iVxMBXkV please can somebody give me a hint? that 187.132.70.12 i dont have any idea who could it be, tha paste means that is trying to connect or is already connected? |
| 20:09.01 | autofsckk | or what should i read so i can understand how to know to do that debubing? |
| 20:09.48 | becca_r | http://en.wikipedia.org/wiki/Wireshark |
| 20:11.00 | [TK]D-Fender | autofsckk, that is a response to an OPTIONS (qualify) packet. It means ... pretty much nothing. |
| 20:11.14 | wdoekes2 | peep_: ok.. this manager over http is new to me, but I can reproduce your problem with latest svn |
| 20:11.45 | vader-- | I know i ask this every so often but does anyone have asterisk running in a Vmware ESXi enviroment yet? |
| 20:11.49 | carrar | 127.0.0.1 is hacking you!! :) |
| 20:11.56 | peep_ | wdoekes2: Hrm I hadnt thought about whether or not it was a problem with AJAM...Did you try AMI direct? |
| 20:13.41 | [TK]D-Fender | carrar, Dorothy is at it again..... |
| 20:13.57 | autofsckk | thanks |
| 20:14.07 | jaytee | that IP address, 187.132.70.12, is from Mexico. That can only mean one thing.....THEY'RE STEALING OUR JOBS!!! :-) |
| 20:14.45 | umay | DEY TUK UR JOBZ |
| 20:14.48 | autofsckk | i know sometimes is a little annoying when new people ask you things, i have been in support channels helping too, and all i can say is thank you for your patience |
| 20:15.30 | umay | vader--: yes asterisk in vmware, limited sip only |
| 20:15.34 | wdoekes2 | peep_: ami over 5038 should work fine |
| 20:17.41 | peep_ | wdoekes2: Ahh nevermind AMI over 5038 responds with plaintext so that doesn't help. |
| 20:17.57 | peep_ | wdoekes2: It's gotta be whatever is generating the XML responses for AJAM |
| 20:18.07 | wdoekes2 | exactly |
| 20:18.25 | peep_ | wdoekes2: Lame. Sounds like its time to fill out a bug report |
| 20:18.40 | peep_ | wdoekes2: Thanks for confirming it on your end at least |
| 20:18.44 | wdoekes2 | :) |
| 20:21.54 | vader-- | umay how many phones? did it work well? |
| 20:22.19 | eZz | autofsckk: try to install fail2ban |
| 20:22.33 | umay | hundreds of phones but just for MWI blinking |
| 20:22.47 | umay | no voice traffic |
| 20:23.05 | eZz | autofsckk: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk |
| 20:24.10 | autofsckk | eZz: thanks, i will read that |
| 20:24.14 | p3nguin | "connect" is such an inaccurate term in VoIP. |
| 20:24.19 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:24.30 | p3nguin | It's UDP; there is no connection. |
| 20:24.45 | eppigy | well it is managed by rtp |
| 20:24.50 | eppigy | which is sessionized |
| 20:24.59 | eZz | autofsckk: np, btw that system helps me against an attacks from china |
| 20:25.01 | eppigy | ^____________^ |
| 20:25.10 | p3nguin | Still no connection, since it's UDP. |
| 20:25.20 | eppigy | but the connection is rtp |
| 20:25.22 | eppigy | not udp |
| 20:25.41 | p3nguin | You don't think RTP is UDP? |
| 20:26.00 | eppigy | rtp is a stateful transport method |
| 20:26.20 | p3nguin | Why would we have people forwarding their UDP 10000-20000 for RTP if it isn't UDP? |
| 20:26.32 | p3nguin | UDP can have state, too. |
| 20:26.35 | eppigy | http://en.wikipedia.org/wiki/Real-time_Transport_Protocol#Sessions |
| 20:26.37 | p3nguin | iptables says so. |
| 20:26.50 | eppigy | that is my closing argument |
| 20:26.54 | eppigy | enjoy |
| 20:27.38 | p3nguin | "RTP and RTCP typically use unprivileged UDP ports (1024 to 65535)" |
| 20:28.13 | eppigy | that isnt the arugment |
| 20:28.17 | eppigy | it is stateful |
| 20:28.19 | eppigy | and a connection |
| 20:28.25 | eppigy | lets try to stay on track |
| 20:29.16 | [TK]D-Fender | UDP doesn't have a state. However on an application level, RTP may. |
| 20:29.31 | [TK]D-Fender | Though it isn't maintained by the stack itself |
| 20:30.50 | eppigy | lets just hug it out |
| 20:31.00 | *** join/#asterisk spditner (~simon@206-248-134-157.dsl.teksavvy.com) |
| 20:31.23 | spditner | anyone else seeing ConfBridge failing to mute users before the marked user arrives? |
| 20:32.07 | spditner | using bridge_softmix |
| 20:32.56 | McBoingBo | how can I analize a pcap file that was from a g729 call? |
| 20:34.07 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 20:34.46 | eZz | McBoingBo: as I remember wireshark can do that |
| 20:36.26 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 20:37.30 | *** join/#asterisk garymc (~chatzilla@host31-53-157-167.range31-53.btcentralplus.com) |
| 20:37.50 | McBoingBo | eZz, well it doesnt play nice with g729 codec |
| 20:38.56 | r0m|u | p3nguin, you avail? |
| 20:38.59 | McBoingBo | I just want to know if the wireshark analysis is accurate for the g729 codec and that it only has issues playing back the stream |
| 20:39.21 | eZz | McBoingBo: ok thank you for update. I didn't launched this one during year |
| 20:39.29 | McBoingBo | but I think even the analysis is not right, the only answer so far I got was "bend over" lol |
| 20:40.01 | eZz | McBoingBo: what about g729 debug inside * ? |
| 20:40.15 | McBoingBo | eZz, I dont follow.... |
| 20:40.54 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 20:41.13 | *** join/#asterisk NeoLobster (~NeoLobste@snugglenets.com) |
| 20:41.27 | eppigy | oh hai NeoLobster |
| 20:42.14 | NeoLobster | hi |
| 20:54.32 | r0t0r00t3r | neolobster -- now that's a handle. |
| 20:57.13 | p3nguin | ~r |
| 20:57.13 | infobot | r is probably The "r" option to Dial will override any sounds you should be hearing and provide a fake ringing sound to the caller. You generally want the caller to hear the sounds they are supposed to hear, not a fake ringing sound. The caller will hear ringing without the "r" option. Using the "r" option is an edge case and should not normally be used or needed. |
| 20:57.17 | p3nguin | r0m|u: ^^^ |
| 20:59.42 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 21:00.03 | r0m|u | ah!!! |
| 21:00.08 | r0m|u | its all clear now! |
| 21:00.10 | r0m|u | :D |
| 21:13.44 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 21:17.41 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 21:22.42 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 21:25.08 | spditner | anyone know offhand what modules are needed for confbridge to work in 10? Getting a 'No provider found, checking channel drivers for SIP', and the call being dropped. |
| 21:30.58 | wonderworld | spditner i compiled 10 without adding modules and it worked out of the box |
| 21:32.22 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 21:33.26 | voxter | I've got an issue with a Dial() string being clipped at 80 characters, I believe on behalf of chan_local, as evidenced in this output: http://pastebin.ca/2093662 - Running asterisk 1.8.5.0 |
| 21:33.32 | voxter | Anyone seen this happen or know of a fix? |
| 21:37.43 | p3nguin | If a call comes from a peer as From: "Caller Name" <sip:devicename@deviceaddress> rather than From: "Caller Name" <sip:CallerIDnumber@deviceaddress> ... what setting controls that? I thought trustrpid might, but it doesn't appear to. |
| 21:38.20 | *** join/#asterisk pietro (~pietro@88-149-224-154.dynamic.ngi.it) |
| 21:38.20 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 21:38.20 | ironm | hello Ihor |
| 21:40.54 | *** join/#asterisk PoWeRKiLL (~powerkill@109.66.209.253) |
| 21:47.10 | *** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz) |
| 21:50.45 | p3nguin | No one? |
| 21:54.49 | WIMPy | Controls what? What that other device sends? |
| 21:55.30 | p3nguin | If the device is the one in charge of that information, then I guess I have to find out where the setting is... |
| 21:55.47 | p3nguin | But I was thinking more along the lines of what asterisk sees. |
| 21:56.07 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 21:56.18 | WIMPy | trustrpid only controls if the rpid is used, if present. |
| 21:58.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:00.01 | *** join/#asterisk [1]SnakeDoctor (~mtaylor1@206-188-70-99.cpe.distributel.net) |
| 22:00.47 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 22:04.20 | dijib | anybody around? |
| 22:04.20 | dijib | http://i.imgur.com/Yz2jb.jpg |
| 22:04.48 | dijib | ive confirmed that the *XX code is being caught by the ATA and not hitting asterisk using sip debug |
| 22:05.03 | dijib | yet still only one ring and then goes to VM |
| 22:07.58 | p3nguin | Okay, so the device decides if it sends From sip:devicename or From sip:calleridnumber? |
| 22:08.56 | WIMPy | Who else could do that? |
| 22:10.48 | dijib | Hook State:OnRegistration State:Online |
| 22:23.08 | dijib | [Nov 10 17:07:02] -- Got SIP response 480 "Temporarily not available" back from 10.0.0.11:5060 |
| 22:23.09 | dijib | [Nov 10 17:07:02] -- SIP/100-00000017 is circuit-busy |
| 22:27.03 | [TK]D-Fender | dijib: imagebin a full status screenshot, and an PB a complete call w/ SIP debug.... |
| 22:31.19 | dijib | [TK]D-Fender, get my PM? |
| 22:35.05 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 22:35.58 | *** join/#asterisk ruied (~ruied@po-217-129-154-119.netvisao.pt) |
| 22:37.49 | [TK]D-Fender | Ok, indeed strange.... |
| 22:42.25 | dijib | i noticed on the *XX page there was a Accept Last Deact Code, so i entered it after the *79 & *79 but that still doesnt work |
| 22:47.45 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-109-66-209-253.red.bezeqint.net) |
| 22:53.15 | *** join/#asterisk PoWeRKiLL (~powerkill@109.66.209.253) |
| 22:53.57 | dijib | p3nguin, what do you make of the PM i just sent? |
| 22:56.16 | p3nguin | For one, your dial options are shit. |
| 22:56.36 | p3nguin | Don't use m and don't use r, and certainly don't use m and r together. |
| 22:56.51 | [TK]D-Fender | dijib: I suggest you factory reset it and reconfigure.... |
| 22:57.05 | dijib | THE ATA.. yeah i was thinking that |
| 22:57.57 | p3nguin | For another, why are you using NAT settings for phones on the same LAN as asterisk? |
| 23:02.09 | dijib | its always worked |
| 23:04.02 | p3nguin | That doesn't make it right. |
| 23:04.09 | p3nguin | It just makes you lucky. |
| 23:05.22 | WIMPy | You get used to that. |
| 23:05.35 | dijib | and i can remoe the sip, hook it up behind a nat. and it will connect without reconfiguration |
| 23:06.21 | p3nguin | I guess it'll be hard for it to make calls if you remove the SIP from it. |
| 23:06.58 | dijib | not if its reconnected to another sub-internet-network |
| 23:07.32 | p3nguin | That doesn't even make sense. If you take out the SIP, it's done for. Finished. |
| 23:07.53 | dijib | this is true |
| 23:07.57 | p3nguin | SIP is what controls calling. Without it, there is no call from or to that device. |
| 23:07.58 | dijib | ok back back to the issue |
| 23:08.05 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 23:08.15 | dijib | yeah but the device responds approprietly |
| 23:09.08 | dijib | guess im factory defaulting.. let you know how it goes |
| 23:10.30 | vader-- | Any of you guys deal with POS equipment? specifially things that require OPOS drivers? |
| 23:12.14 | WIMPy | We can't do data except for fax with some luck. |
| 23:13.53 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 23:15.52 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 23:21.15 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 23:25.59 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 23:26.04 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mjeippjnjmqndqqv) |
| 23:29.20 | p3nguin | What function do I need to look at for checking if a caller ID name contains a certain string? |
| 23:30.52 | p3nguin | I guess REGEX |
| 23:31.02 | ChannelZ | guess it depends on how hard you need to look |
| 23:32.45 | p3nguin | I just want to check if a CID name contains certain words, for example CREDIT or INTEREST. |
| 23:37.21 | p3nguin | I'm thinking something like ${REGEX("CREDIT" ${CALLERID(name)})} would do it. |
| 23:37.44 | p3nguin | I could be mistaken, so I'll test in a minute. |
| 23:45.36 | *** join/#asterisk Janos (~Janos@186.32.185.5) |
| 23:54.26 | Janos | hi, question, is there any way to know remotely which echo canceler is installed in a digium card so i can find out which firmware does it need and finally figure out why it is not been loaded ? |
| 23:54.48 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:58.24 | dijib | [TK]D-Fender, p3nguin, after the Factory Reset && Reconfiguration everything seems to have returned to normal |
