| 00:03.17 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 00:16.15 | DrDamnit | p3nguin: Got it working without the @, but this dial statement still needs help: exten => _29XX,1,dial(IAX2/oshea:[hphinc]/${EXTEN}) |
| 00:16.54 | p3nguin | What's the oshea:[hphinc] part for? |
| 00:16.58 | DrDamnit | If I do not include the rsa key in it, the call goes through (unencrypted), but when I put the key in there, it doesn't do anything, and tcpdump doesn't even show a connection attempt. |
| 00:17.28 | DrDamnit | oshea is the resource, and [outkey] is the RSA key to use for encrypting the call. |
| 00:17.51 | DrDamnit | but I cannot find good docs on it, so it may be wrong. I think that it may be causing asteirsk to send the call to nowhere, which is why tcpdump shows nothing. |
| 00:19.11 | *** join/#asterisk Alborracho (c819d8de@gateway/web/freenode/ip.200.25.216.222) |
| 00:19.16 | Alborracho | hi everybody |
| 00:19.34 | Alborracho | does anyone know how to capture packets form a digium card with tshark? |
| 00:20.10 | Alborracho | im trying to get signalling from my card, ss7 dump does not work properly |
| 00:20.13 | IsUp | DrDamnit: http://www.voip-info.org/wiki/view/Asterisk+iax+rsa+auth |
| 00:20.32 | DrDamnit | Alborracho: tcpdump -i eth0 -s 65535 -w /tmp/dump.pcap |
| 00:20.36 | DrDamnit | Then open the capture with wireshark. |
| 00:20.46 | IsUp | Alborracho: are you running chan_ss7 or libss7? |
| 00:21.02 | Alborracho | but i want the packets from my digium card not my if card |
| 00:21.06 | Alborracho | chan ss7 |
| 00:21.07 | DrDamnit | Oops. Sorry Misread your question. |
| 00:21.54 | IsUp | Alborracho: i am running chan_ss7 too and dump works fine. |
| 00:22.36 | IsUp | Alborracho: also i am using Sangoma card, unable to capture with tcpdump. but i dont know if its different with Digium |
| 00:23.02 | Alborracho | the problem is that im using T1s and ss7 dump shows a lot of errors in wireshark |
| 00:23.15 | Alborracho | so i was told that if i get the packets with tshark they are clean |
| 00:23.36 | IsUp | Alborracho: what kind of 'errors'? |
| 00:24.03 | Alborracho | the signalling is ANSI |
| 00:24.21 | Alborracho | the error is "malformed packet" |
| 00:26.05 | IsUp | Alborracho: well, no idea at all. if you are having problems with your linkset, maybe you can talk to your telco. so they can run capture on their end |
| 00:26.45 | Alborracho | too bad, the telco takes like two years for such a thing, if anyone knows wolud be apreciated |
| 00:27.04 | Alborracho | ill continue searching in google |
| 00:27.15 | IsUp | Alborracho: whats your issue? |
| 00:27.31 | Alborracho | i cant get the call to complete |
| 00:27.39 | Alborracho | i have the CICs in idle |
| 00:27.46 | Alborracho | link is INSERVICE |
| 00:27.56 | Alborracho | but the call falls in timeout |
| 00:28.07 | Alborracho | always, so i wanted to get a trace to see the problem |
| 00:28.09 | IsUp | Alborracho: great. you are talking about outgoing calls, right? |
| 00:28.17 | Alborracho | yes, outgoing calls |
| 00:30.00 | IsUp | Alborracho: double check with your provider that they are enabled "two-way channels". i had similiar issue before. and ive realized that my telco just enabled linkset and NOT configured CICs as two-way |
| 00:30.19 | IsUp | Alborracho: are you able to upload your capture file? |
| 00:30.35 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 00:30.45 | Alborracho | this was working 3 hours ago, dont know waht changed, sure, in baste bin? |
| 00:30.56 | Alborracho | *paste |
| 00:31.20 | IsUp | Alborracho: upload anywhere doesnt matter. also pastebin your CLI output |
| 00:31.23 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 00:31.32 | IsUp | Alborracho: was it working on same CICs? |
| 00:31.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jxxcfrsszfkbtkgi) |
| 00:32.27 | Alborracho | yes, it was working |
| 00:32.41 | Alborracho | http://www.2shared.com/file/rZtcbViW/dump20120105v2.html |
| 00:34.32 | IsUp | Alborracho: okay and pastebin your CLI output for a failed call |
| 00:34.44 | Alborracho | 1 sec pls |
| 00:34.54 | DrDamnit | p3nguin: new dial statement: exten => _29XX,1,dial(IAX2/hph:[hphinc]@oshea/${EXTEN}) gives me this error on the other end: Rejected connect attempt. No secret present while force encrypt enabled. Thoughts? |
| 00:34.57 | IsUp | Alborracho: oh and please include your ss7.conf |
| 00:35.20 | Alborracho | ok |
| 00:36.44 | Alborracho | http://pastebin.com/6jseMHx7 |
| 00:36.55 | p3nguin | Why not put the user, key, and secret in the peer entry? |
| 00:37.08 | p3nguin | Then you use the regular Dial format. |
| 00:37.59 | Alborracho | IsUp: the Auto fallthrough comes after a moment |
| 00:38.11 | DrDamnit | p3nguin: I did: http://pastebin.com/6Q7CKfCr |
| 00:38.14 | *** part/#asterisk LeLutin (~lelutin@vps-1036922-2706.manage.myhosting.com) |
| 00:38.46 | p3nguin | okay |
| 00:39.01 | DrDamnit | Is there something wrong with that config? |
| 00:41.37 | p3nguin | I didn't see anything wrong, as long as what you specified is what you really meant. |
| 00:42.17 | DrDamnit | p3nguin: Understood. I am as confident in it being correct as you can be without it working. :-) |
| 00:43.41 | IsUp | Alborracho: try with this config: http://pastebin.com/raw.php?i=KeYSDBa2 and restart your asterisk |
| 00:44.36 | Alborracho | IsUp: ok |
| 00:46.07 | Alborracho | IsUp: got an error http://pastebin.com/d4qLQx0p |
| 00:46.47 | IsUp | Alborracho: okay, change channels => 1-24 to channels => 1-23 under l3 |
| 00:47.13 | Alborracho | got it :) |
| 00:48.00 | Alborracho | IsUp: same result Everyone is busy/congested at this time (1:0/1/0) bus faster |
| 00:49.23 | Alborracho | IsUp: wait, the CICs are blocked, going to reset them |
| 00:49.34 | DrDamnit | The dial statement (I think) is not sending the secret to authenticate with the remote end. How do I debug this? |
| 00:49.55 | IsUp | Alborracho: okay if it fails revert your config back. ive found that in readme of chan_ss7, http://pastebin.com/raw.php?i=A7pFgdkJ you can try tshark with that |
| 00:50.24 | p3nguin | You don't need to send a secret in the dial command when you have configured the peer entry correctly. |
| 00:50.35 | Alborracho | IsUp: ok, ill try taht |
| 00:54.18 | DrDamnit | p3nguin: Would you mind looking at the two entries? I would appreciate a second set of eyes to see the mistake I am not seeing: http://pastebin.com/Jq8C1iaB |
| 00:54.51 | Alborracho | IsUp: got a files and see the trace but i cant copy the file, "mtp2fifo.pcap: not a regular file" |
| 00:56.40 | IsUp | Alborracho: maybe you need use a paramater on tshark to write capture to a file |
| 00:56.47 | p3nguin | There is no "nat" setting in iax. |
| 00:57.12 | p3nguin | And I don't see a register statement in the remote side. |
| 00:57.31 | IsUp | Alborracho: tshark -V -i /tmp/mtp2fifo.pcap -w ss7.pcap |
| 00:57.41 | Alborracho | IsUp: ok |
| 00:57.48 | DrDamnit | Sorry, I only put the peer in there: register => oshea:[oshea]@voip.myserver.com |
| 00:58.24 | IsUp | /tmp/mtp2fifo.pcap -> this is just interface, as you specify in tshark with '-i', '-w' write capture to a file |
| 00:58.54 | p3nguin | Assuming the info is all correct, nothing really sticks out as being the problem. |
| 00:59.18 | DrDamnit | p3nguin: I was afraid you'd say that. :-/ |
| 00:59.41 | DrDamnit | This is the error I keep getting: Rejected connect attempt. No secret present while force encrypt enabled. |
| 00:59.45 | Alborracho | IsUp: got this "tshark: Unrecognized libpcap format" |
| 00:59.46 | p3nguin | And here I always thought that no news was good news. |
| 01:00.03 | DrDamnit | Which, to me, says: "You're not sending me anything I can use to authenticate you." |
| 01:00.39 | p3nguin | What happens if you actually configure a secret? |
| 01:00.54 | p3nguin | Neither entry has one. |
| 01:01.04 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 01:01.13 | DrDamnit | I assumed that with key authentication (like ssh) a secret password was not required. |
| 01:01.15 | DrDamnit | I'll try it. |
| 01:01.47 | p3nguin | I think it will still use the secret to authenticate, but use the keys to encrypt the transaction. |
| 01:02.57 | DrDamnit | Added secret=password (awesomely secure, I know), and got this: [Jan 5 20:02:13] NOTICE[1017]: chan_iax2.c:7855 authenticate_verify: Call Terminated, Incoming call is unencrypted while force encrypt is enabled.[Jan 5 20:02:13] NOTICE[1017]: chan_iax2.c:10911 socket_process: Host x.x.x.x failed to authenticate as hph |
| 01:03.39 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 01:03.40 | IsUp | Alborracho: probably ss7 dumps not in proper format. so just use output, you dont need a capture file for 1 call actually. |
| 01:05.36 | p3nguin | You should probably specify the username for each side, as well. |
| 01:07.05 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 01:08.06 | Alborracho | IsUp: can i get the output with ">>" into a file? |
| 01:08.13 | IsUp | Alborracho: yes |
| 01:08.27 | DrDamnit | add usernames, same result. |
| 01:09.41 | IsUp | DrDamnit: how about pastebin your iax configs for both sides? |
| 01:10.03 | p3nguin | http://pastebin.com/Jq8C1iaB |
| 01:10.04 | DrDamnit | prepares pastebins. |
| 01:10.19 | p3nguin | I guess you've made a couple changes since that one, though. |
| 01:14.26 | Alborracho | IsUp: got his output http://pastebin.com/KNCasMYx |
| 01:14.27 | DrDamnit | My side is 499 lines long with multiple other working non-encrypted IAX peers. So it just has general and the peer I have an issue with. |
| 01:14.34 | DrDamnit | This should be a complete view: http://pastebin.com/0CpBr63U |
| 01:15.29 | DrDamnit | I have verified that the outkeys are the private keys for that box with which that box will use to authenticate. The keys are loaded and initialized. |
| 01:15.37 | DrDamnit | and the sums are the same. |
| 01:15.58 | IsUp | Alborracho: pastebin output of 'ss7 show channels', your CIC looks invalid on capture. also OPC/DPC too. |
| 01:16.56 | Alborracho | IsUp: http://pastebin.com/ZJbPbTYD |
| 01:18.20 | IsUp | Alborracho: okay, at this point you should share your capture with telco. also who supplied 'stp' value? your telco? |
| 01:18.44 | Alborracho | IsUp: yes, they did |
| 01:19.20 | IsUp | Alborracho: okay, one thing to test. use this format: Dial(SS7/telco/18296490618... or Dial(SS7/70/18296490618 |
| 01:19.39 | DrDamnit | Does it matter that I have the private and public keys in /var/lib/asterisk/keys together? (hphinc.key and hphinc.pub are both there, might it be using the wrong key to authenticate?) |
| 01:20.41 | Alborracho | IsUp: ok, ill give it a try, do you give any paid consultory? |
| 01:20.42 | p3nguin | Wouldn't you have hphing.key and oshea.pub? |
| 01:20.59 | p3nguin | And on the other side, oshea.key and hphinc.pub? |
| 01:21.46 | DrDamnit | on the oshea side, I have hphinc.pub, oshea.pub and oshea.key. On my side, I have hphinc.pub, hphinc.key, and oshea.pub. |
| 01:21.54 | DrDamnit | myside = hphinc, btw. |
| 01:22.14 | IsUp | Alborracho: no i am a volunteer :p just lemme know if you need anything else. i am running chan_ss7 on different mobile telcos |
| 01:22.41 | DrDamnit | Looking at the tcpdump in wireshark, it looks like oshea is asking hph to authenticate as user hph (this is correct), but the RSA response is obviously not what it is expecting because it rejects the call. This, to me, would indicate a key issue? |
| 01:24.48 | *** join/#asterisk Neptu (~Neptu@213.88.195.203) |
| 01:25.24 | Alborracho | IsUp: didnt work, thanks a lot for your help, tomorrow ill try to contact someone in netfors, we work with chan ss7 in a lot of telcos too, but always with E1s, this is the only one with T1s... |
| 01:25.54 | IsUp | Alborracho: good luck |
| 01:26.05 | Alborracho | thank you, see you |
| 01:26.30 | DrDamnit | Here's a look at the keys: http://pastebin.com/hm7ddkr0 |
| 01:29.38 | IsUp | DrDamnit: can you run 'keys show' in CLI |
| 01:30.42 | DrDamnit | IsUp: Here's a look at the keys: http://pastebin.com/hm7ddkr0 |
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| 01:31.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:31.39 | IsUp | DrDamnit: ok can you try 1 way call with this config http://pastebin.com/raw.php?i=ZUkpqFKD |
| 01:32.53 | DrDamnit | DrDamnit follows directions... |
| 01:36.52 | *** join/#asterisk Neptu (~Neptu@213.88.195.203) |
| 01:40.47 | DrDamnit | IsUp: Give me a minute. You're assuming the remote box is acting like the telco, but my box is acting like the telco. |
| 01:40.56 | DrDamnit | have to swap some things. |
| 01:41.11 | IsUp | DrDamnit: okay |
| 01:42.16 | DrDamnit | IsUp: I keep getting: sked to authenticate to x.x.x.x with an RSA key, but they don't allow RSA authentication |
| 01:46.51 | DrDamnit | On my server: No registration for peer 'oshea' (from x.x.x.x) |
| 01:47.45 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:51.08 | IsUp | DrDamnit: i think its same issue with yours: http://lists.digium.com/pipermail/asterisk-users/2005-May/099056.html - im not sure if its a bug or missing configuration but ive found that example http://www.panoramisk.com/121/iax-and-rsa-authentication/1/en/ |
| 01:53.36 | DrDamnit | This article says that it works abyway, but iax2 show registry shows that the iax trunk is unregistered. |
| 01:54.10 | DrDamnit | This post is from 2005, also, I would imagine they would have gotten it fixed by now? |
| 01:56.03 | DrDamnit | If I use the other configs (the one from pastebin) it registers. The one way trunks do not even register. What could be different? |
| 01:56.18 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 01:58.15 | DrDamnit | Testing the theory. |
| 02:00.06 | DrDamnit | IsUp: Completely strange: I used the originally pastebin'd configs, but changes them to a user / peer relationship. When the remote side tried to dial my extension I got: "I don't know how to authenticate [SOME OTHER PEER NOT RELATED TO THIS PROJECT] to x.x.x.x. Thoughts? |
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| 04:22.26 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 04:22.46 | Yourname` | Hello. Who is available to have a casual telephony related basic discussion over PM? |
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| 04:47.55 | *** join/#asterisk high-rez (~gus@carrera.bourg.net) |
| 04:48.53 | high-rez | I'm trying to use GotoIf to send a user directly to their own voicemail if they dialed themselves. It doesn't seem to be working - is this statement right? exten => _XXXX,2,GotoIf($["${CALLERID(num)}" = ${EXTEN}]?ownvm) |
| 04:51.39 | p3nguin | high-rez: VoiceMailMain(${CALLERID(num)}@default) |
| 04:51.50 | p3nguin | Forget the GotoIf. |
| 04:52.22 | high-rez | I'm trying to figure out how that woudl work in my dial context :) |
| 04:53.03 | high-rez | I onlt want to dial VoiceMainMain *if* they are calling themselves. How would I check if they are calling their own extension ? |
| 04:56.13 | p3nguin | You aren't using a defined "check your voice mail" extension? |
| 04:56.46 | p3nguin | I use *86 (*VM) to check own voice mail. I thought that was what you meant. |
| 04:57.15 | p3nguin | You're trying to see if a phone is trying to call itself? And, if it is, bypass the Dial() and go straight to VoiceMail()? |
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| 04:59.08 | p3nguin | With regard to the line you provided before, the reason it does not match is because of the quotations. "${CALLERID(num)}" isn't likely to ever equal ${EXTEN} |
| 04:59.29 | p3nguin | "${CALLERID(num)}" = "${EXTEN}" maybe |
| 04:59.30 | high-rez | p3nguin: No my setup is really quite weird. :) Asterisk is connected to a GSM Base Station via openbsc and lcr |
| 04:59.39 | high-rez | Yeah, the quote was missing on EXTEN - that fixed it. |
| 05:00.11 | *** join/#asterisk zh-home (~zerohalo@74.61.196.236) |
| 05:05.32 | high-rez | p3nguin: Thanks for your help. It's all working like a real GSM network would with regards to voicemail entry. I just need to get voicemail to kick off an SMS to the phone for indication and it'll be done. :D |
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| 07:17.07 | ChannelZ | hmm. Is there a simple linux command that can do a simple test on its arguments and return a code depending on the evaluation? |
| 07:19.01 | florz | nope, there are only commands that can do complex tests |
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| 07:24.03 | ChannelZ | actually 'test' might do it |
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| 08:06.59 | cadey | Hi guys,quick question about the queue with autofill. If if we had 5 attendents (avaliable channels to ring) and 5 people in the queue and autofill is set to yes then each channel attendent would ring for each person in the queue however what happens if one attendent does anser the call, does that poor sole on the other end of the line potential go from possition 2 to 5? |
| 08:07.22 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 08:07.28 | cadey | or is it intelegent enough to just match the next pick up to the next on in the queue even though they are ringing in parralel? |
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| 08:37.04 | FluxiFlax2022 | hi all I have an issue in that I use asterisk with a2blling, I have a problem in that my carrier sent me a bill for 10 000 mins for 200 calls, I have the 200 calls on record but only for 1400 mins, most of the calls the carrier sent are 20 minutes sharp no my systems they dont show for more than a few mins...I know this is not strictly an Asterisk Q. ..any ideas ? |
| 08:40.14 | *** join/#asterisk Assalino (53f49842@gateway/web/freenode/ip.83.244.152.66) |
| 08:40.24 | irroot | FluxiFlax2022 dont look at A2B records look at the CDR's |
| 08:40.33 | Assalino | Morning! :) |
| 08:43.21 | Assalino | Guys, quick question: to get Asterisk up and running in a way that it can make calls to any given number and run them through a 3rd party speech recognition engine, would I need special hardware? |
| 08:43.46 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:43.48 | Assalino | they mention some connection hardware for some solutions, but I'm not sure if that'd be more for PBXs, etc |
| 08:44.27 | Assalino | Could I just boot a server on Amazon Cloud (for example), install asterisk, configure packages, etc and it would work? |
| 08:44.41 | Assalino | or would it have to be a dedicated server where I'd have to install some special hardware? |
| 08:46.04 | WIMPy | If you need to interface to hardware, you need something to plug it in. If you only use virtual (IP) connections, that's all you need. |
| 08:47.32 | Assalino | I don't have any external hardware at the moment, so there'd be no dependencies |
| 08:47.52 | Assalino | so in theory, I'd need no hardware and could make the calls via VoIP, correct? |
| 08:48.24 | WIMPy | yes |
| 08:48.37 | Assalino | awesome |
| 08:48.50 | Assalino | so if I were to go ahead, should I get ASterisk 1.8 or 10.0? |
| 08:48.55 | Assalino | I'm not sure what the differences are |
| 08:49.12 | irroot | WIMPy o/ |
| 08:49.39 | WIMPy | Hi irroot |
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| 09:09.25 | IsUp | hi |
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| 09:11.26 | ChannelZ | 10 is TEH FUTURE! |
| 09:11.33 | FluxiFlax2022 | irroot, A2B does create the CDRs, you mean Asterisk generated CDRs ? |
| 09:12.07 | ChannelZ | If you're just building a box, no reason not to go with 10 really |
| 09:12.15 | irroot | FluxiFlax2022 indeed eliminate discrepencies |
| 09:12.31 | FluxiFlax2022 | irroot, I double checked with the engineers acutally CDRs and A2B are showing same values..which are different from carrier though |
| 09:13.15 | FluxiFlax2022 | irroot, another thing to note is that only certain routes have this issue..but on the same carrier |
| 09:13.33 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 09:14.28 | Assalino | thanks for your help guys. I'm out :) |
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| 10:02.07 | aberrios | anyone using Asterisk 1.8 with Lumenvox SRE? |
| 10:03.58 | KNERD | nope |
| 10:08.21 | Yourname` | nope |
| 10:09.21 | WIMPy | No one |
| 10:09.27 | WIMPy | ever |
| 10:09.30 | aberrios | :( |
| 10:22.56 | KNERD | never heard of it |
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| 10:29.43 | *** join/#asterisk Beznalogov (~Tantor@46.47.91.113) |
| 10:29.47 | Beznalogov | Hello |
| 10:30.28 | Beznalogov | I use asterisk 1.6 and I have made some changes to my extensions.conf file and some files that are being included in extensions.conf |
| 10:31.10 | Beznalogov | Obviously something is wrong as the config files are not being read and I have no contexts, except the ones that are automatically being generated (by fe meetme) |
| 10:31.24 | Beznalogov | How can I find out where the errors are? I don't get any message at all |
| 10:32.04 | IsUp | Beznalogov: start asterisk with 'asterisk -vvvvvvvvvvvgfc' |
| 10:33.27 | Beznalogov | Thanks |
| 10:54.26 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 10:54.34 | Beznalogov | This is a very strange problem. When I use the command that you just gave, it loads everything perfectly. When I use 'sudo service asterisk start', no extensions are being loaded |
| 10:56.33 | IsUp | Beznalogov: okay tail -f /var/log/asterisk/full or wherever your log file is. and then start asterisk with 'service asterisk start' |
| 10:56.51 | IsUp | Beznalogov: so you can see actually whats going on when you start asterisk |
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| 12:33.21 | jacco_7564 | Hello folks |
| 12:33.26 | IsUp | hi |
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| 12:59.08 | DanfromUK | Hi, this is not strictly an asterisk questions, but... ive got a client with an old digital pbx phone systems connected to an isdn30e line. I've been shown a sip gateway that can connect to asterisk on one side, and has a socket that the old phone system can connect to. But its a bit pricey. |
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| 12:59.56 | DanfromUK | Is there such a thing as an ISDN30e PCI card that can be used with a copy of Asterisk, that can act like a voip gateway between the old phone system, and our asterisk box? |
| 13:00.26 | DanfromUK | Or can anyone recommend a gateway that isnt too expensive? |
| 13:00.39 | DanfromUK | they use 8 channels of the isdn30e |
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| 13:09.58 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:11.45 | dunlingli | I need send all the test messages stored in database when a phone is registered to Asterisk. I created a macro in dialplan to do this. How can I invoke this macro from a python codes? |
| 13:13.28 | leifmadsen | dunlingli: you likely need to originate a call to the phone -- probably via AMI |
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| 13:15.53 | garymc | Hi can anyone help me with my call file. |
| 13:16.02 | garymc | any ideas what im doing wrong with the call file. Here is an example of my call file. http://pastebin.com/yPxDWWeK |
| 13:16.05 | garymc | So when the calls come through to the phones in the office the phones ring with the internal ring instead of the external |
| 13:16.15 | garymc | ring tones |
| 13:16.58 | [TK]D-Fender | garymc, Don't ask FreePBX questions in here... |
| 13:17.14 | wdoekes2 | garymc: aren't those ring tones decided by the callerid? then I suggest you set the callerid appropriately |
| 13:17.15 | [TK]D-Fender | garymc, especially unprefixed |
| 13:17.35 | garymc | oh i thought call files where asterisk, possibly |
| 13:17.58 | [TK]D-Fender | garymc, You want to change dialplan processing that sets all these rules. |
| 13:18.04 | [TK]D-Fender | garymc, that = FreePBX |
| 13:18.26 | [TK]D-Fender | garymc, Answers in that channel |
| 13:29.51 | dunlingli | Leifmadsen: I use "CLI command originate", in my python code, something like self.command("channel originate sip/1 extension %s@MessageContext" %(extension)"). it can sent the messages, but I have to answer the first phone in order to reach dial plan contained my macro. Is there anyway this can work without answer the phone? Also, in AMI command originate, what shall I use as <Channel>? can I use my macro as Application? |
| 13:30.49 | leifmadsen | dunlingli: if you don't answer the phone, what are you playing audio to? |
| 13:31.02 | leifmadsen | you have to answer the phone for there to be another channel to do something with |
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| 13:35.10 | *** join/#asterisk awclin (~alinford@85.88.137.50) |
| 13:36.11 | dunlingli | leifmadsen: My goal is to send messages from Asterisk without any human interaction since the messages are stored in Asterisk database. I don't want to create a voice channel for communication. Is this possible for Asterisk to handle? |
| 13:37.16 | leifmadsen | dunlingli: then I'm confused as to what you're trying to send if you're not sending a voice message |
| 13:37.34 | leifmadsen | dunlingli: make the other channel a Local channel then that answers the call, but just does a Wait() or something |
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| 13:38.11 | leifmadsen | Local/msg-centre@mycontext |
| 13:38.18 | leifmadsen | exten => msg-centre,1,NoOp() |
| 13:38.23 | leifmadsen | same => n,Answer() |
| 13:38.27 | leifmadsen | same => n,Wait(20) |
| 13:38.29 | leifmadsen | <PROTECTED> |
| 13:38.31 | leifmadsen | soemthing like that |
| 13:43.14 | dunlingli | leifmadsen: it is text messages, not voice mail. |
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| 13:51.20 | *** join/#asterisk portnov (~portnov@v-20678-unlim.vpn.mgn.ru) |
| 13:51.28 | portnov | hi all. |
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| 14:17.33 | dunlingli | leifmadsen: I tried with your suggestions, it seems that I have to answer the first phone. I tried to use queue, but it seems only create voice channels. I am new here, is there a way to use queue for MessageSend? I look through AMI commands, there is no MessageSend, Note that Sendtext doesn't work in my case. Can I send text messages through AMI? |
| 14:19.10 | [TK]D-Fender | * is not a messaging platform. |
| 14:19.52 | IsUp | :) |
| 14:19.57 | jim_ec2 | I thought there was stuff like SIMPLE for * |
| 14:20.38 | jim_ec2 | to push messages |
| 14:20.51 | [TK]D-Fender | jim_ec2, Where do you see this? |
| 14:22.10 | jim_ec2 | http://en.wikipedia.org/wiki/SIMPLE |
| 14:23.19 | jim_ec2 | one of the rfc's related to that was written by an engineer at digium |
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| 14:23.39 | file | haha... like we have time to write RFCs |
| 14:24.29 | jim_ec2 | wish I had my laptop with me today, still have that rfc open |
| 14:24.52 | jim_ec2 | oh wait nevermind, that rfc was aix |
| 14:29.23 | *** join/#asterisk awclin (~alinford@85.88.137.50) |
| 14:29.44 | [TK]D-Fender | jim_ec2, No, what tells you that * supports this? |
| 14:30.41 | leifmadsen | you mean iax I think |
| 14:30.53 | jim_ec2 | [TK]D-Fender: I can't remember, I was looking at a bunch of stuff about * and came across SIMPLE |
| 14:30.56 | leifmadsen | if you were looking at the aix rfc, you were lookin at the wrong one :) |
| 14:30.58 | jim_ec2 | http://www.rfc-editor.org/rfc/rfc5456.txt |
| 14:31.07 | [TK]D-Fender | leifmadsen, Digium wrote IBM's UNIX? OMG! |
| 14:31.10 | jim_ec2 | yeah iax :) |
| 14:31.28 | leifmadsen | I love how people write out full sentences and then replace typing asterisk with * which is super obfuscated when reading a sentence |
| 14:31.55 | jim_ec2 | leifmadsen: your brain builds a regex for this channel |
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| 14:32.39 | [TK]D-Fender | leifmadsen, Only you and complete newbs seem to have any issue with * = asterisk :) |
| 14:32.54 | [TK]D-Fender | leifmadsen, That's a very perculiar contrast... |
| 14:33.18 | [TK]D-Fender | "peculiar" |
| 14:33.23 | jim_ec2 | can't remember how to escape * in vim regex |
| 14:33.31 | jim_ec2 | s/\*/asterisk |
| 14:33.45 | Faustov | gives jim_ec2 a cracker |
| 14:34.36 | jim_ec2 | om nom nom nom |
| 14:35.17 | jim_ec2 | uuuggghhhh sql server is so slow |
| 14:37.33 | Faustov | slow query language |
| 14:38.35 | [TK]D-Fender | Some Questions Linger |
| 14:38.56 | jim_ec2 | sit quietly longtiem |
| 14:39.41 | jim_ec2 | is blink a good *nix softphone? |
| 14:40.25 | *** join/#asterisk portnov (~portnov@v-20678-unlim.vpn.mgn.ru) |
| 14:40.54 | portnov | I can't set up Asterisk to receive incoming Jingle calls :/ |
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| 14:47.45 | dunlingli | Even though * is not messaging platform, I have to pass text messages through AMI for my project. Could you point me which part of * source codes I need to modify in order to add MessageSend command to AMI? |
| 14:49.12 | portnov | * receives series of |
| 14:49.12 | portnov | JABBER: asterisk INCOMING: <iq from='... |
| 14:49.12 | portnov | messages, but seems it does not interpret it as jingle calls :/ |
| 14:49.21 | IsUp | . |
| 14:49.48 | [TK]D-Fender | dunlingli, Where does "MessageSend" exist? |
| 14:49.58 | leifmadsen | dunlingli: which version are you using? |
| 14:51.28 | leifmadsen | [TK]D-Fender: that's what she is wanting to add |
| 14:51.36 | leifmadsen | so it obviously doesn't exist anywhere |
| 14:51.36 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:52.25 | [TK]D-Fender | leifmadsen, I'm sensing something is being poorly communicated and I'd really prefer a confirmation... |
| 14:52.46 | leifmadsen | "Could you point me which part of * source codes I need to modify in order to add MessageSend command to AMI?" |
| 14:52.52 | dunlingli | leifmadsen: Asterisk 10: SVN-10-r348517M |
| 14:53.06 | [TK]D-Fender | leifmadsen, I think this was following that Originate attempt to use a Dialplan app for this, no? |
| 14:53.22 | [TK]D-Fender | leiTo which I'm sure your advice for use of a local channel was never comprehended or followed |
| 14:53.58 | [TK]D-Fender | dunlingli, what "MessageSend" code/app/function/whatever are you referring to? |
| 14:54.41 | n3hxs | found a reference to Asterisk and AMI in Google. |
| 14:55.01 | [TK]D-Fender | n3hxs, SMRT |
| 14:55.11 | n3hxs | And found http://asterclick.drclue.net/Documentation.php very interesting. |
| 14:55.29 | n3hxs | but it may not be related at all.... |
| 14:56.20 | [TK]D-Fender | n3hxs, well it does relate to 2 other words used in that request... but no mention of what this other entity is. |
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| 14:58.17 | dunlingli | In my dialplan, I use new application in *10, messageSend (https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend) |
| 14:58.44 | [TK]D-Fender | dunlingli, then use the Orignate method you were working on before. |
| 14:58.49 | [TK]D-Fender | Originate* |
| 14:59.57 | [TK]D-Fender | And that doc is tragically vague as to what protocols it is even related to. |
| 15:02.24 | [TK]D-Fender | Also the instructions are contradictory on syntax. |
| 15:02.25 | [TK]D-Fender | Send a text message. The body of the message that will be sent is what is currently set to MESSAGE(body). |
| 15:02.27 | dunlingli | The problem is that I have to answer the first phone using originate. I might did something wrong, here is my command "channel originate sip/1 ex (my fake phone) extension ${EXTEN}@message_context" |
| 15:02.38 | [TK]D-Fender | Syntax : MessageSend(to[,from]) |
| 15:03.07 | [TK]D-Fender | dunlingli, You were told to call a LOCAL channel that will simply wait (kill time) so the extension can do it's business |
| 15:03.07 | portnov | does Jingle support in Asterisk 1.8 work at all? :/ |
| 15:03.14 | [TK]D-Fender | its* |
| 15:03.58 | leifmadsen | dunlingli: change sip/1 to a Local channel |
| 15:03.58 | [TK]D-Fender | [8:38] <leifmadsen> Local/msg-centre@mycontext exten => msg-centre,1,NoOp() same => n,Answer() same => n,Wait(20) |
| 15:04.07 | dunlingli | D-Fender: I am able to use MessageSend in dialplan, but I need use it from my python codes through AMI |
| 15:04.07 | leifmadsen | ya... over an hour ago |
| 15:04.16 | [TK]D-Fender | dunlingli, He gave you this half an hour ago |
| 15:04.23 | [TK]D-Fender | hour & half* |
| 15:04.26 | leifmadsen | dunlingli: use a Local channel; answer is the same as it was earlier this morning |
| 15:04.56 | [TK]D-Fender | dunlingli, "sip/1" <- don't do this. You were already told what to do instead |
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| 15:06.14 | dunlingli | what should I put to replace "sip/1" in originate command? |
| 15:07.29 | [TK]D-Fender | [8:38] <leifmadsen> Local/msg-centre@mycontext |
| 15:07.31 | [TK]D-Fender | ^^^^66 |
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| 15:14.55 | mjordan | [TK]D-Fender: just as an aside, the MessageSend documentation is vague wrt the protocol because it is protocol agnostic. It makes the determination as to the protocol based on the information you provide it in the parameters |
| 15:15.40 | [TK]D-Fender | mjordan, And provides no sample for what the syntax is for even one of these hypothetically valid protocols :) |
| 15:15.53 | [TK]D-Fender | mjordan, Nor a list so you could even start guessing |
| 15:16.04 | mjordan | nope, which has caused some confusion. It might be a good idea for us to write an article on the wiki :-) |
| 15:16.16 | [TK]D-Fender | s/some/total |
| 15:16.17 | mjordan | Right now, its SIP MESSAGE and Jabber (IIRC) |
| 15:16.42 | [TK]D-Fender | mjordan, So you're "informing" me and you're not even sure? ;) |
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| 15:17.07 | mjordan | I'm sure of the first one, give me a second on the second one :-P |
| 15:17.46 | mjordan | and I was right |
| 15:17.50 | mjordan | hooray for my brain working in the morning |
| 15:18.04 | leifmadsen | mjordan: huzzuh! |
| 15:18.39 | [TK]D-Fender | mjordan, And you might want to clarify the discrepency between MessageSend([from,]to) and (MessageSend(body) which is contradictory |
| 15:19.06 | mjordan | Hmm... I don't see MessageSend(body) |
| 15:19.14 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend |
| 15:19.21 | [TK]D-Fender | Unless you're expected to actually call the app twice and once it has everything it will actually do it's job. |
| 15:19.35 | [TK]D-Fender | Send a text message. The body of the message that will be sent is what is currently set to MESSAGE(body). |
| 15:19.43 | [TK]D-Fender | ^ 1st line of Description |
| 15:19.45 | mjordan | that's a function, not the application |
| 15:19.51 | [TK]D-Fender | AH |
| 15:19.56 | mjordan | :-) |
| 15:19.59 | leifmadsen | rookie mistake :) |
| 15:20.08 | [TK]D-Fender | Fair enough.... ish... |
| 15:20.23 | mjordan | but if its causing this much confusion, it might behoove us to write an example |
| 15:20.32 | dunlingli | Thanks lefmadsen and D-Fender. I misunderstood your solution. It works now |
| 15:20.57 | [TK]D-Fender | Could be worded better, but no, the name isn't the same and that it is a function is something you could reasonably guess. But you'd still have to guess... |
| 15:21.20 | leifmadsen | well, I wouldn't say "guess" |
| 15:21.33 | leifmadsen | all uppercase implies dialplan function |
| 15:21.45 | [TK]D-Fender | leifmadsen, Maybe they're just yelling :) |
| 15:23.21 | [TK]D-Fender | I'm wondering why the MESSAGE() isn't simply a parameter of MessageSend() instead of as a superfluous (?) function... |
| 15:24.00 | jim_ec2 | anonymous inner? |
| 15:25.08 | jim_ec2 | oh hrm no gitweb or the like, I was wondering why the wiki didn't link to source |
| 15:25.27 | leifmadsen | superfluous yes |
| 15:27.24 | jim_ec2 | hah, https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend |
| 15:27.56 | jim_ec2 | i'm working with some weird commercial fork of * and most of the tables comes from that company and not * |
| 15:27.57 | dunlingli | As for function MESSAGE (https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_MESSAGE), it seems that parameter to and from are writable instead of read only |
| 15:27.57 | [TK]D-Fender | leifmadsen, silly people :) |
| 15:28.08 | jim_ec2 | now I know that at least the CDR table is * |
| 15:31.09 | jim_ec2 | oh hot sauce |
| 15:31.20 | jim_ec2 | dialplan config in lua :) |
| 15:31.31 | jim_ec2 | huggles mod_magnet from lighttpd |
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| 15:41.19 | jim_ec2 | is there something like the cdr table but for calls or is that a vendor specific thing? |
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| 15:42.35 | [TK]D-Fender | jim_ec2, CDR : Call Detail Reporting |
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| 15:42.43 | [TK]D-Fender | jim_ec2, How is that not "for calls"? |
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| 15:46.06 | jim_ec2 | [TK]D-Fender: oh no I know what cdr is now, I have a table named calls as well |
| 15:46.33 | jim_ec2 | [TK]D-Fender: trying to ascertain what is * and what is from the vendor |
| 15:47.08 | mattsqz | fonality's pbxtra web interface limits recordings to 100 per extension. i have been trying to figure out how to make it record all calls. turns out, they use a file called user_recordings.xml to control that. doh. ive been trying to do things the normal asterisk way. |
| 15:51.45 | [TK]D-Fender | jim_ec2, *'s configs tell you what you configured * to access.... |
| 15:52.26 | jim_ec2 | [TK]D-Fender: i don't have access to *, just mysql creds |
| 15:52.44 | jim_ec2 | no documentation either, i've been reverse engineering the schema |
| 15:53.36 | jim_ec2 | i'm just working on an old php script that is pulling from the calls table to an in house app |
| 15:53.55 | jim_ec2 | it's working for the most part, just having trouble trying to get inbound missed calls |
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| 15:55.21 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 15:56.54 | jim_ec2 | i wish this script had been written against the cdr table, it has much more useful information than this silly calls table |
| 15:58.42 | [TK]D-Fender | jim_ec2, If you are running blind then you're pretty much screwed. Don't ask for an autopsy without providing a body... |
| 16:00.47 | jim_ec2 | [TK]D-Fender: yeah that's pretty much what I've been saying for a week now |
| 16:01.08 | [TK]D-Fender | And that is only a suggested table... |
| 16:01.21 | jim_ec2 | [TK]D-Fender: i asked our "phone guy" and he said that company isn't done writing docs on their new warez |
| 16:01.23 | [TK]D-Fender | and format.field names may vary, etc depending how they set things up |
| 16:02.33 | jim_ec2 | yeah, imo the in house rails app should just connect to the mysql box and lookup relevant calls in the cdr table |
| 16:02.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:02.57 | jim_ec2 | then this call result sync process wouldn't have to exist at all |
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| 16:21.05 | *** join/#asterisk gaetronik (~gaetan@ks370400.kimsufi.com) |
| 16:21.08 | gaetronik | Hi |
| 16:21.34 | gaetronik | i've a strange behaviour of a sip account |
| 16:22.20 | gaetronik | i set up an exten to receive call with Answer=>MusicOnHold and another with Answer=>Echo |
| 16:22.36 | gaetronik | when i call the number to reach musiconhold it works fine |
| 16:22.42 | gaetronik | the one with echo it failed |
| 16:22.58 | gaetronik | but if i insert a playback before the echo it works great |
| 16:27.21 | [TK]D-Fender | gaetronik, Because you have to Answer a channel before calling Echo |
| 16:27.32 | [TK]D-Fender | gaetronik, Playback just automatically answers |
| 16:29.12 | gaetronik | [TK]D-Fender: i answer before echo |
| 16:29.36 | [TK]D-Fender | Show us |
| 16:29.49 | gaetronik | w8 |
| 16:30.46 | gaetronik | exten => 0733,1,Answer() |
| 16:30.46 | gaetronik | exten => 0733,n,PlayBack(hello) |
| 16:30.46 | gaetronik | exten => 0733,n,Echo() |
| 16:30.53 | gaetronik | it works |
| 16:31.13 | [TK]D-Fender | ~pb |
| 16:31.14 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:31.15 | [TK]D-Fender | ^^^ |
| 16:31.36 | gaetronik | yes but for 3 lines... |
| 16:31.48 | gaetronik | without the Playback line it fails |
| 16:32.49 | gaetronik | might it be the sdp negociation working differently? |
| 16:34.24 | [TK]D-Fender | I'm not seeing actual call atempts with debug. |
| 16:34.36 | gaetronik | ah ok |
| 16:34.37 | gaetronik | sorry |
| 16:39.58 | aberrios | anyone know of a channel I could get help with Speech Recognition Grammars? |
| 16:40.02 | aberrios | done a quick search |
| 16:40.27 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v004-093.mobile.uci.edu) |
| 16:42.10 | gaetronik | http://pastie.org/private/s9nnafhz0mvmjjevhqitbw |
| 16:42.21 | gaetronik | [TK]D-Fender: _/^ |
| 16:43.01 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 16:43.08 | BenC[UK] | Hi Guys |
| 16:43.30 | BenC[UK] | it seems by default asterisk is trying to create SIP channels with slin |
| 16:43.31 | BenC[UK] | Asked to create a SIP channel with formats: 0x40 (slin) |
| 16:43.35 | BenC[UK] | how can I change this? |
| 16:43.45 | [TK]D-Fender | BenC[UK], Set codes in your peers |
| 16:44.10 | BenC[UK] | peers as users.conf ? |
| 16:45.32 | gaetronik | BenC[UK]: sip.conf |
| 16:45.34 | [TK]D-Fender | BenC[UK], As in where you configure your sip peers |
| 16:46.02 | BenC[UK] | I have already set them there, disalllow all, then allow g729, ulaw |
| 16:46.12 | [TK]D-Fender | BenC[UK], There is no "default". There is only what you put there. |
| 16:46.20 | [TK]D-Fender | BenC[UK], PB it |
| 16:46.21 | [TK]D-Fender | ~pb |
| 16:46.21 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:46.44 | *** join/#asterisk razu (~razu@2001:ad0:1:1:202:b3ff:fe36:921c) |
| 16:47.50 | BenC[UK] | http://pastebin.com/8MFXp8pa |
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| 17:06.43 | gaetronik | no one have idea for my strange sound issue |
| 17:06.46 | gaetronik | ? |
| 17:11.00 | *** join/#asterisk minaguib (~mina@modemcable132.139-59-74.mc.videotron.ca) |
| 17:11.46 | minaguib | Hey folks. If I have an authenticated user (iax2 in this case) in some context in the dialplan - is there a way/some variable that tells me the "mailbox" assigned to them in iax2.conf ? |
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| 17:21.03 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:24.48 | BenC[UK] | if I want to use this.... |
| 17:24.49 | BenC[UK] | ${HASH(SIP_CAUSE,<channel-name>)} |
| 17:25.10 | BenC[UK] | do I replace the <channel-name> with the channel name? or is it auto worked out? |
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| 17:32.03 | BenC[UK] | its nor working either way |
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| 17:58.57 | *** join/#asterisk pecenipicek (~vrga@cpe-109-60-73-242.zg3.cable.xnet.hr) |
| 17:59.25 | pecenipicek | hello folks. tell me, whats the proper way to determine the state of a phone via AMI? |
| 17:59.42 | pecenipicek | IE, is the phone registered or not, for the basics. |
| 18:00.01 | pecenipicek | ExtensionState action always returns -1 for a phone that is online actually. |
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| 18:10.20 | k3asd` | hi |
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| 18:52.47 | [TK]D-Fender | pecenipicek, phone != extension |
| 18:53.07 | pecenipicek | yeah, i fg |
| 18:53.09 | pecenipicek | derp |
| 18:53.14 | pecenipicek | i figured it out. |
| 18:53.38 | pecenipicek | i'm trying to get my script to not take a second to check every phone -.- |
| 18:53.50 | pecenipicek | (PHP is involved) |
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| 19:23.42 | cmendes0101 | When sql is used in a dialplan some reason, some reason after high use connections down seem to close. I'm getting this with lsof TCP server1:59569->server2:mysql (CLOSE_WAIT) |
| 19:23.56 | cmendes0101 | dont* |
| 19:24.31 | *** join/#asterisk timahvo1 (~rogue@41.90.249.214) |
| 19:24.41 | cmendes0101 | I've had to increase the ulimit -n or else it gets hit with connect create port and too many files open in asterisk |
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| 21:08.39 | *** join/#asterisk kannan (kannan@123.238.235.77) |
| 21:09.58 | kannan | hello all, is it possible to setup asterisk behind a NAT and have IP phones register from a remote network, the phones are also behind a NAT ? Also, if so , can anyone suggest a good VOIP service in USA that will provide DIDs for the asterisk behind the NAt |
| 21:12.49 | [TK]D-Fender | kannan, * works fine behind NAT and no carrier has anything to do with that |
| 21:12.51 | [TK]D-Fender | ~sipnat |
| 21:12.52 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 21:14.13 | kannan | [TK]D-Fender , thank you , i will read up ; and how can the asterisk have the ip phones to register to it ? |
| 21:14.23 | [TK]D-Fender | l? |
| 21:14.38 | [TK]D-Fender | Please rephrase that. It does not parse. |
| 21:15.34 | kannan | the Ip phones are requiring the outbound proxy IP address or domain ; but the asterisk server has only a NAT'd Ip addrewss |
| 21:16.12 | kannan | is it possible for the phones as well as the asterisk box to be behing NAT (not the same NAT) |
| 21:16.28 | [TK]D-Fender | read the guide |
| 21:16.58 | kannan | [TK]D-Fender , thanks again , i will |
| 21:16.58 | [TK]D-Fender | port forward to your server as directed, let * know what its WAN IP is by one of the means listed and then it is public to the world. |
| 21:21.02 | kannan | [TK]D-Fender , thanks , it works that the phone is registered, but i cannot hear anything of audio |
| 21:24.19 | [TK]D-Fender | Patient..... niiiiice...... |
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| 21:33.08 | *** join/#asterisk VoipForces (~Adium@69.165.204.66) |
| 21:33.43 | VoipForces | Anyone knows if it's possible to have Asterisk dial through a MagicJack adapter or a MagicJack account ? |
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| 21:39.43 | [TK]D-Fender | BBIAB |
| 21:40.01 | VoipForces | BBIAB ?? LOL |
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| 21:51.12 | keith4 | VoipForces: yes. i recall seeing someone write it up. have to extract the SIP creds from the device, somehow |
| 21:51.33 | keith4 | just google "asterisk magicjack". it's been written |
| 21:53.03 | VoipForces | keith4: yes had seen that, was wondering if there was a less hackish way |
| 21:53.20 | keith4 | nope |
| 21:53.34 | keith4 | not that i've seen, anyway |
| 21:53.49 | VoipForces | Thanks. |
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| 23:23.02 | tech | evening all... |
| 23:23.09 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v004-093.mobile.uci.edu) |
| 23:23.34 | tech | i have s8800 from avaya with signal group and trunk configured correctly on avaya end.. however ic an't see to get it down on my freepbx/asterisk side. |
| 23:23.39 | tech | there error i am getting: |
| 23:23.40 | tech | [2012-01-06 18:14:19] ERROR[3494]: chan_ooh323.c:668 ooh323_request: Call to undefined peer 192.168.23.3[2012-01-06 18:14:19] WARNING[3494]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'OOH323' (cause 0 - Unknown) |
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| 23:24.30 | hardwire | anybody using the asterisk manager interface ala HTTP? (AJAM)? |
| 23:24.53 | hardwire | I'm not entirely sure if once you log in via it.. events queue up waiting for WaitEvent commands to come in. |
| 23:24.59 | hardwire | and what thresholds I'm dealing with. |
| 23:26.04 | hardwire | it sounds like a much better method than using AMI via the TCP socket if it doesn't let events slip through the cracks. |
| 23:26.25 | hardwire | but I'm also not sure how it affects Asterisks performance. |
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