| 00:44.17 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 00:46.00 | saxa | WIMPy: hi, you mean current sqlite or current asterisk ? |
| 00:48.34 | WIMPy | sqlite from slackware-current |
| 00:50.41 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 01:04.51 | saxa | oh WIMPy ok, I got it. I can make my own package then. |
| 01:05.24 | WIMPy | ? |
| 01:05.47 | WIMPy | Just get the current package and upgrade. |
| 01:07.29 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 01:23.29 | *** join/#asterisk navaismo (~navaismo@189.241.66.140) |
| 01:25.15 | navaismo | So first time using TLS/SRTP, Running the sip debug on asterisk im still seeing the exten to dial is that normal? |
| 01:25.46 | navaismo | Using wireshark I cant see the call at all, so im not sure if this works or not. Im using asterisk 11.4.0 and Yealink T20P |
| 01:27.15 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 01:31.47 | WIMPy | Would be bad if Asterisk wasn;t able to decrypt the messages, wouldn't it? |
| 01:32.03 | WIMPy | Are you watching the right TCP port? |
| 01:34.12 | navaismo | on wireshark? Well im capturing all traffic but I can only see options message to others phones. In the sip debug i can see the <--- SIP read from TLS:10.0.1.110:21143 ---> |
| 01:34.30 | navaismo | port must be 5061 right? but always is registering on random port |
| 01:34.52 | navaismo | How can i know if the call is really secured? |
| 01:35.03 | WIMPy | 5061 is th default, yes. |
| 01:35.25 | WIMPy | But you have to see the traffic in wireshark. Are you sure you're not using any filter? |
| 01:35.34 | navaismo | yep sure |
| 01:36.27 | WIMPy | The traffic may be unreadable, but it doesn't disappear. |
| 01:36.46 | navaismo | ok let me restart wireshark |
| 01:36.57 | navaismo | some errors about bus has been displayed |
| 01:42.55 | navaismo | updating wireshark |
| 01:49.28 | navaismo | WIMPy, now im seeing packets with TLSv1 from yealink to asterisk and then a bunch of udp packets(i guess the rtp?) |
| 01:50.06 | navaismo | and the wireshark feature to find voip calls and show it with the audio is not working |
| 01:50.15 | navaismo | so my TLS/SRTP setup is working? |
| 01:52.00 | navaismo | And most important how is this possible without importing any certificate to the Yeaklink phone? |
| 01:55.21 | navaismo | Ok the bunch of udp packages in wireshark has the same ports used in the SDP transaction so i guess the rtp is encrypted too |
| 01:55.25 | navaismo | Hmm so im happy now |
| 01:55.32 | navaismo | ???? |
| 01:55.52 | navaismo | Yealink and asterisk rocks on TLS/srtp? |
| 01:56.05 | navaismo | feels very paranoid now |
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| 02:49.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:12.34 | WIMPy | has some doubts wireshark would find the rtp steam without having seen the sdp messages. |
| 03:25.03 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 03:25.55 | hebber | Scratching my head - v11.4 - queue application - how can i turn off the users in a queue dropping out when they press a single key extension? |
| 03:27.29 | WIMPy | By not turning that feature on. |
| 03:27.48 | [TK]D-Fender | don't give them any to exit with |
| 03:27.54 | hebber | so in queue - I have added a context - is that the reason? |
| 03:27.59 | [TK]D-Fender | yes |
| 03:28.30 | hebber | ok, propably a nice feature configured correctly :) |
| 03:35.37 | hebber | hmm - I use realtime queues - the field context is empty (not NULL) - did a core reload - the problem still exist. queue.conf does not have a any context defined. |
| 03:37.35 | [TK]D-Fender | perhaps tre config is cached.... |
| 03:37.44 | [TK]D-Fender | the* |
| 03:38.57 | hebber | could be - I will do a full stop - start later and see if that resolves it |
| 03:39.22 | WIMPy | Yeah, sip also has a very bad habbit of easily adding new configuration but not deletong removed entries or modifying changed ones on reload. |
| 03:39.37 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 03:40.00 | WIMPy | Sometimes a 'core reload' seems to do more than specific reloads. |
| 03:40.17 | WIMPy | If that doesn't work either, 'core restart when convenient'. |
| 03:40.54 | hebber | I specific queue reload, then core reload, will try core restart soon |
| 03:41.31 | WIMPy | That's what the "when convenient" part is for. |
| 03:43.41 | hebber | learned something new there - did a " restart when convenient" and when it came back online - QUEUES works as intended :) |
| 03:43.42 | hebber | thanks |
| 03:44.41 | WIMPy | Without "core"? How old is your Asterisk? |
| 03:44.51 | WIMPy | nvm |
| 03:45.03 | hebber | 11.4 sorry - sorry its me which is broken :) |
| 03:45.20 | WIMPy | ok :-) |
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| 05:04.06 | igcewieling | It looks like maybe someone broke into one our CPE media GWs, made a bunch of international calls, then restored the config back so we didn't know what they actually did. |
| 05:04.39 | igcewieling | I have some very very bad words for such people. |
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| 06:09.31 | spindritf | is there a way to dismiss the bar at the bottom of wiki.asterisk.org? |
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| 06:13.04 | spindritf | hm, I could use a few yes/nos before diving into the asterisk docs anyway: |
| 06:13.33 | spindritf | I can connect asterisk to a regular VoIP provider over SIP, right? and then in turn connect my phone(s) to the asterisk instance? |
| 06:13.43 | [TK]D-Fender | yes |
| 06:13.48 | [TK]D-Fender | that is what * is |
| 06:13.52 | [TK]D-Fender | ~b2bua |
| 06:13.52 | infobot | methinks b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 06:14.13 | spindritf | and make calls phone → asterisk → another phone OR phone → asterisk → voip provider → regular lines? |
| 06:14.21 | [TK]D-Fender | yes |
| 06:15.05 | spindritf | and it will work with two different voip providers, and multiple phones on my side? |
| 06:15.21 | [TK]D-Fender | as much as you want |
| 06:16.03 | spindritf | and I can use it to receive regular SIP calls, at myname@myserver.tld? |
| 06:16.10 | [TK]D-Fender | sure |
| 06:16.13 | spindritf | awesome, thanks |
| 06:17.02 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-anoznxfawpncxvce) |
| 06:17.32 | spindritf | one more, is there a way to have a reasonably secure asterisk instance on the internet? I want to be able to connect from my mobile on random wifis to the server |
| 06:18.05 | spindritf | the guide suggests limiting access only to known networks |
| 06:19.43 | [TK]D-Fender | many ways to secure a system |
| 06:20.02 | [TK]D-Fender | Including VPNing you mobile |
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| 06:23.32 | spindritf | ok, thanks |
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| 06:40.25 | b7 | Hello everyone. I have a question: I'm going to use Asterisk as an office IP-PBX for my SIP-phones. Does it means that the iax module of Asterisk wouldn't be used and can be safely disabled? |
| 06:40.46 | [TK]D-Fender | sure |
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| 06:45.15 | b7 | [TK]D-Fender, thank you. |
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| 07:47.16 | peetaur | I want to connect asterisk to a regular phone network. First question, do I connect my asterisk box just like a phone, so if you dialed its extension, you'd get the asterisk menu? Second question, how do I know what sort of hardware to buy? If they tell me the lines are ISDN, how do I know if they are PRI or BRI? |
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| 07:59.15 | bulkorok | peetaur: BRI => 2 channels, PRI => 30/24 channels (depends on location) |
| 07:59.48 | bulkorok | you can use internal interface-cards (like the ones from digium or beronet) |
| 08:00.36 | bulkorok | or use an external appliance that converts the ISDN to SIP and connect the appliance via IP |
| 08:00.57 | bulkorok | patton smartnode or sth like that |
| 08:01.32 | bulkorok | peetaur: usually you KNOW when you have a PRI connection |
| 08:04.08 | peetaur | so then it's probably BRI ...? |
| 08:04.40 | peetaur | and there is no trouble putting the phone line right into the interface card, like the voltage coming out of the card won't go on the phone line and disrupt things? |
| 08:06.06 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 08:09.19 | bulkorok | nope |
| 08:10.25 | bulkorok | you must configure the card for endpoint usage... |
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| 08:14.27 | peetaur | are there any caveats for what I'm trying to do? Will I fail if I'm a noob ;) |
| 08:16.09 | Chainsaw | peetaur: Well, it is important to know what signalling you have. We'll ask it differently. How many lines do you have on that connector? |
| 08:16.34 | Chainsaw | peetaur: Dozens or only a few? |
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| 08:19.12 | peetaur | I don't know how to gather the right info. I assume it's only 1 or 2, probably 1. |
| 08:19.51 | peetaur | they use little 4 pin connectors |
| 08:20.42 | peetaur | I'm in Germany. They look like the analog connectors I had in Canada which had 1 line per 2 wires. |
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| 08:22.57 | peetaur | the phones we have now are this model, if it helps http://www.ebay.de/itm/6x-TB13-14D-Bosch-Avaya-Tenovis-Integral-schwarz-TB-13-14-D-/181189531971?pt=Telefonanlagen&hash=item2a2fbcd943 |
| 08:24.32 | bulkorok | peetaur: that seem to be "system-phones" that are NOT compatible to ISDN! |
| 08:25.18 | bulkorok | they work probably only with "Bosch Avaya oder Tenovis Integral 33/33x oder 55/55 Office" pbx's |
| 08:25.28 | Chainsaw | peetaur: So if you are to replace this, you will not be interfacing with the digital RJ11 connectors to the Avaya handsets, but replacing the box in the basement (PABX) that is connected to the actual telephone network. |
| 08:26.12 | Chainsaw | peetaur: You could have the Asterisk box between the telephone network and the PABX to do smarter call routing, but it would help to know what you're trying to accomplish first. |
| 08:27.05 | peetaur | all I want is to keep the existing phone network here, and attach one asterisk box to it so it can connect to another asterisk box in another country to make VOIP calls to the extension(s) there |
| 08:27.16 | peetaur | (for when I'm back in Canada... a very small scale thing) |
| 08:27.22 | *** join/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 08:28.14 | WIMPy | peetaur: http://voice.yeti.dk/Asterisk_vs_ISDN/ |
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| 08:44.14 | peetaur | bulkorok: so I got a report back about the phones, and think you're right that it's something strange, and they said I need a phone gateway rather than interface card, and it has to be attached to their PBX rather than any phone line. |
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| 08:44.59 | WIMPy | Didn't you say yourself that you wanted to connect to the PBX? |
| 08:46.15 | WIMPy | YOu probably want to add Asterisk as another line. |
| 08:48.38 | peetaur | well I was hoping it would be as simple as adding the asterisk box as an extension and then you can call in like it's a phone bot |
| 08:49.05 | WIMPy | You can do it that way as well, off course. |
| 08:49.30 | bulkorok | peetaur: but you need a connection that is compatibel to the PBX and asterisk |
| 08:49.38 | WIMPy | Up0 is also just a form of a BRI. YOu can get NTs for that from all PBX vendors. |
| 08:49.46 | peetaur | and I'd lose things like call conferencing, right? |
| 08:50.33 | WIMPy | Connecting as an extension only makes sense for incomming calls. For outgoing, you can;t route via an extension. |
| 08:50.45 | WIMPy | Unless you want to do it call-through style. |
| 08:50.47 | *** join/#asterisk mads (5ab8319e@gateway/web/freenode/ip.90.184.49.158) |
| 08:52.26 | mads | Hello, i keep getting some strange 3klang messages in my asterisk can someone help me? |
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| 08:55.27 | wdoekes | mads: ~ask |
| 08:55.31 | wdoekes | ~ask |
| 08:55.31 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:56.47 | mads | well the cdr-csv file shows these "","104","s","3klang","""104"" <104>","SIP/x.x.x.x-0063c9d0","","PlayTones","!950/330|!1400/330|!1800/330|0","2013-08-06 10:48:19","2013-08-06 10:48:19","2013-08-06 10:48:19",0,0,"ANSWERED","DOCUMENTATION" all the time and i dont know what it is |
| 08:57.33 | mads | and then the number 104 changes to 105 after and then 106 and so on ... |
| 08:59.44 | wdoekes | I'm not familiar with the cdr record order |
| 09:00.39 | wdoekes | but increasing 104, 105, 106 calls look like brute force scanning for accounts |
| 09:01.07 | bulkorok | I'd say that too.... .check asterisk and fail2ban |
| 09:07.11 | mads | yeah i was affraid of brute force, we already have a mechanism checking for people who is trying to log on with different account and password mismatches and then locking them out after X attempts, but this thing that only shows with a 3klang in the csv is new to me |
| 09:08.12 | WIMPy | Wahtever it is, it succeeded. |
| 09:09.02 | WIMPy | But isn't that the destination column? |
| 09:09.36 | WIMPy | Like the destination of a Goto(), probably of a failed call? |
| 09:10.33 | mads | as far as i remember 1. field is billing account if you have a billing system, then 2. field is the caller (source) and 3. field is destination |
| 09:10.57 | mads | but i'm not 100% sure for about that |
| 09:12.57 | WIMPy | Yes, the call ended in extension s in context 3klang. |
| 09:13.03 | WIMPy | So it's your dialplan. |
| 09:14.54 | mads | yes but the source 104, 105, 106... is not actual accounts on our phone system so i dont understand why this is comming up or if it is some kind of nasty way to scan for accounts |
| 09:15.31 | WIMPy | 1. You let them in. |
| 09:15.49 | WIMPy | 2. Only if you named your accounts the same as your extensions. |
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| 10:49.10 | duchman | hi all |
| 10:49.46 | duchman | have hangup on asterisk calls immediately the receiver picks up........any help on this pls? |
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| 10:55.29 | ABerrios | duchman, can you pastbin the cli verbose output? |
| 10:55.40 | duchman | ok |
| 10:55.43 | duchman | pls hold |
| 10:56.42 | duchman | login as: root |
| 10:56.43 | duchman | root@192.168.169.60's password: |
| 10:56.43 | duchman | Last login: Tue Aug 6 10:14:37 2013 from 192.168.169.14 |
| 10:56.43 | duchman | [root@localhost ~]# asterisk -rvvvvvv |
| 10:56.43 | duchman | Asterisk 11.2.1, Copyright (C) 1999 - 2012 Digium, Inc. and others. |
| 10:56.43 | duchman | Created by Mark Spencer <markster@digium.com> |
| 10:57.25 | ABerrios | !pastebin |
| 10:58.02 | floren | ~pastebin |
| 10:58.02 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 10:58.09 | ABerrios | stupid keyboard |
| 10:58.14 | floren | :) |
| 10:58.22 | ABerrios | duchman, pastebin...otherwise you get kicked |
| 10:58.42 | ABerrios | ~pastebin |
| 10:58.42 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 10:58.55 | duchman | ok |
| 11:00.38 | ABerrios | nope |
| 11:01.43 | ABerrios | oh dear |
| 11:02.00 | ABerrios | facepalms |
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| 11:02.15 | ABerrios | duchman, paste into pastebin then paste the link here |
| 11:02.39 | duchman | ok |
| 11:03.57 | duchman | it is not giving me any link...... |
| 11:04.12 | duchman | I am in pastebin now |
| 11:04.43 | ABerrios | duchman, is it not in your browser url when you've submitted the paste? |
| 11:05.23 | duchman | no just www.pastebin.com... |
| 11:06.11 | ABerrios | submit the paste |
| 11:07.00 | duchman | not seeing any submit link/button |
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| 11:08.23 | duchman | let me check for something similar to pastebin |
| 11:08.27 | duchman | pls hold |
| 11:19.06 | duchman | http://pastebin.ca/2428746 |
| 11:22.25 | *** join/#asterisk andrewya_ (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 11:31.18 | ABerrios | duchman, can you do the same with sip debug enabled |
| 11:31.29 | duchman | ok |
| 11:32.54 | duchman | http://pastebin.ca/2428747 |
| 11:39.13 | *** join/#asterisk cneb3000 (~cneb3000@90.192.191.93) |
| 11:40.54 | cneb3000 | hi everyone. if I were to hypothetically be providing you a sip trunk, and on receipt of your invites respond with a 302 redirect every time in order for us to load balance. would that be a pain in the arse to you? I'm reviewing different methods of load balancing SIP traffic, wanted to hear asterisk suers thoughts. |
| 11:41.04 | cneb3000 | asterisk users* |
| 11:50.52 | duchman | hi.....u there? |
| 11:53.37 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 12:06.06 | *** join/#asterisk Draecos (~Draecos@124.150.57.219) |
| 12:06.35 | *** join/#asterisk izbushka (~izbushka_@193.23.225.11) |
| 12:06.44 | izbushka | hi |
| 12:06.57 | *** join/#asterisk andrewyager (~andrewyag@11-104-141-114.static-dsl.realworld.net.au) |
| 12:07.37 | izbushka | is there any way to MessageSend to user on another asterisk connected by IAX2? |
| 12:07.39 | *** join/#asterisk andrewya_ (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 12:19.01 | *** join/#asterisk duchman (~paulo@41.190.3.104) |
| 12:20.34 | duchman | have hangup on asterisk calls immediately the receiver picks up........any help on this pls? |
| 12:20.59 | izbushka | duchman, codecs? look at debug |
| 12:21.23 | duchman | sip debug? |
| 12:21.28 | duchman | allow=ulaw |
| 12:21.42 | duchman | for both phones |
| 12:21.44 | *** join/#asterisk pensmit (~pensmit@nc-184-3-96-113.dhcp.embarqhsd.net) |
| 12:22.06 | pensmit | Is AsteriskNow better than the FreePBX distribution? |
| 12:22.07 | duchman | http://pastebin.ca/2428747 |
| 12:22.54 | duchman | <izbushka>http://pastebin.ca/2428747 |
| 12:25.22 | duchman | <izbushka>both phones have ulaw enabled |
| 12:26.09 | izbushka | duchman, core set debug 4, core set verbose 4, call and see what happens |
| 12:26.26 | duchman | ok |
| 12:28.40 | duchman | <izbushka>http://pastebin.ca/2428770 |
| 12:29.53 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 12:30.04 | duchman | <izbushka> same result......asterisk hangs up as soon as the receiver picks the call |
| 12:30.48 | izbushka | duchman, sorry, don't know |
| 12:31.54 | duchman | <izbushka>ok thanks |
| 12:31.56 | ABerrios | duchman, have you tried a different soft phone? |
| 12:32.32 | *** join/#asterisk Nickinator (~user@123-243-142-239.static.tpgi.com.au) |
| 12:32.58 | duchman | <ABerrios>tried 3cx and linphone |
| 12:33.46 | ABerrios | duchman, do a sip show peers |
| 12:33.57 | duchman | ok |
| 12:35.22 | duchman | http://pastebin.ca/2428780 |
| 12:36.29 | duchman | <ABerrios, http://pastebin.ca/2428780 |
| 12:38.58 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 12:45.38 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:45.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:51.08 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ac78:d36a:e6dc:5da2) |
| 12:51.54 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 12:54.44 | WIMPy | isn't canreinvite=nonat supposed not to try to reinvite if one party is in a localnet and the other outside? |
| 12:56.53 | WIMPy | somehow ends up with MOH on one end. |
| 12:57.14 | [TK]D-Fender | what are you running |
| 12:57.53 | WIMPy | The new bridging stuff. |
| 12:59.04 | [TK]D-Fender | that parameeter was deprecated for directmedia long ago |
| 13:00.08 | WIMPy | It's still in the current sample config. |
| 13:00.49 | [TK]D-Fender | try what it should use |
| 13:02.11 | WIMPy | "No" is ok, but so far "nonat" seemed to be the best option. |
| 13:04.04 | *** join/#asterisk modesto916 (~modesto@189-90-192-72.isimples.com.br) |
| 13:05.40 | WIMPy | Argh. |
| 13:05.58 | duchman | hi all |
| 13:06.01 | WIMPy | )&!%*!%&!£% |
| 13:06.08 | duchman | have hangup on asterisk calls immediately the receiver picks up........any help on this pls? |
| 13:06.13 | WIMPy | The ITSP uses RFC1918 addresses :-( |
| 13:06.32 | WIMPy | That should make sure you get undesired effects. |
| 13:07.16 | WIMPy | duchman: Likely the same thing. Do you have directmedia enabled? Try to set it to "no". |
| 13:08.49 | duchman | <WIMPy>ok ....let me check |
| 13:09.04 | duchman | <WIMPy>in sip.conf? |
| 13:09.15 | WIMPy | yes |
| 13:12.08 | CeBe | WIMPy: what version of asterisk are you running |
| 13:12.37 | WIMPy | trunk |
| 13:17.23 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 13:19.21 | *** join/#asterisk banane_ (528bc51a@gateway/web/freenode/ip.82.139.197.26) |
| 13:21.20 | banane_ | could anyone please tell me how i can determine which extension / sip account has picked up when using dial M? |
| 13:23.29 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 13:24.06 | jmetro | asterisk variables probably have something in there. |
| 13:27.01 | banane_ | at least until now i couldn´t find anything. do you know if there´s a listing of all variables @jmetro? |
| 13:29.09 | WIMPy | DIALEDPEERNAME / DIALEDPEERNUMBER |
| 13:30.18 | *** join/#asterisk duchman (~paulo@41.190.2.124) |
| 13:30.33 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 13:30.39 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 13:31.12 | banane_ | thank you very much WIMPy |
| 13:33.18 | jmetro | oh, the asterisk variables |
| 13:33.33 | spindritf | so my options for securing an instance of asterisk on the 'net are: a) strong passwords and a hopeful attitude, b) tls which is not widely supported, or c) VPN? |
| 13:33.41 | *** join/#asterisk upp (928c103e@gateway/web/freenode/ip.146.140.16.62) |
| 13:33.55 | jmetro | VPN is a+ |
| 13:35.07 | upp | hello, why i can't recive calls?? http://pastebin.com/sgpmXaTJ |
| 13:35.18 | spindritf | vpn it is then |
| 13:35.47 | jmetro | upp: you have a flaw in your keyboard, try switching the F and P keys. |
| 13:36.45 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 13:36.47 | upp | jmetro: sorry i don't understand you? |
| 13:37.54 | ABerrios | yay for vpn |
| 13:38.52 | *** join/#asterisk italorossi (~italoross@mde5036d0.tmodns.net) |
| 13:39.12 | Greenlight | upp: Try providing some information which might suggest the problem isn't just your keyboard then. |
| 13:39.22 | jmetro | ^ |
| 13:39.36 | [TK]D-Fender | huh? |
| 13:40.06 | upp | Greenlight: do you mean my computer keyboard? |
| 13:40.15 | [TK]D-Fender | upp: show an actual call attempt at verbose 10 |
| 13:40.24 | [TK]D-Fender | ~pb |
| 13:40.24 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:40.26 | [TK]D-Fender | ^^^ |
| 13:40.44 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 13:42.20 | duchman | hi all,my asterisk hangs up as soon as the receiver picks the call...any help on this pls. Tried directmedia=no still no joy....... |
| 13:42.35 | asteriskmonkey | mm |
| 13:42.39 | asteriskmonkey | try and answer() |
| 13:42.42 | asteriskmonkey | before your dial string |
| 13:43.09 | duchman | ok |
| 13:43.15 | asteriskmonkey | I found in asterisk 1.11 and dialing a linksys i needed this for unknown reasons :/ |
| 13:43.33 | [TK]D-Fender | makes no sense |
| 13:43.42 | asteriskmonkey | i know :) |
| 13:44.15 | asteriskmonkey | I was debugging someone elses box , i was tempted to recompile since i never even seen that sort of behaviour |
| 13:45.13 | *** join/#asterisk colinwielga (~colinwiel@static-72-64-129-133.tampfl.fios.verizon.net) |
| 13:47.56 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 13:48.42 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 13:49.50 | *** join/#asterisk generalhan_ (~generalha@about/windows/staff/generalhan) |
| 13:53.31 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:02.10 | duchman | <asteriskmonkey,calls not connecting anymore now |
| 14:02.11 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:02.11 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:02.28 | jmetro | but at least its not hanging. |
| 14:02.33 | jmetro | PB a failed call |
| 14:02.45 | duchman | ok |
| 14:03.36 | duchman | http://pastebin.ca/2428804 |
| 14:07.58 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 14:11.52 | [TK]D-Fender | duchman: your first call is not matching any account on your system. your second LOOKS like it is but is poiunting to a bad context from Freebx |
| 14:12.58 | duchman | <[TK]D-Fender> I have an extension 5555 registered to from-sip-external context |
| 14:13.38 | [TK]D-Fender | duchman: you need to have SIP DEBUG enabled while looking at these calls. It is almost assured that due to not matching peers on your system that you are allow codecs you can't actually transcode as well which would explain the immediate drop on answer |
| 14:13.48 | [TK]D-Fender | Why there? |
| 14:14.38 | duchman | testing with an unregistered phone....allowed ulaw/alaw on phones and asterisk |
| 14:14.51 | [TK]D-Fender | NEW CALL |
| 14:14.57 | duchman | ? |
| 14:15.04 | [TK]D-Fender | with SIP DEBUG |
| 14:15.04 | duchman | ok pls hold |
| 14:15.21 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:16.48 | duchman | <[TK]D-Fender,http://pastebin.ca/2428809 |
| 14:19.40 | [TK]D-Fender | show us your dialplan |
| 14:19.56 | duchman | ok pls hold |
| 14:23.58 | *** join/#asterisk floren_ (~floren@unaffiliated/floren) |
| 14:25.57 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 14:27.35 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:28.54 | *** join/#asterisk serafie (~erin@nat/digium/x-fnobchtahjryhnjr) |
| 14:29.54 | duchman | pastebin.ca |
| 14:29.55 | duchman | Sorry, it appears that Pastebin.ca is having some technical difficulties at the moment. It should be back soon. |
| 14:29.55 | duchman | In the meantime, perhaps you'd like to take a look at a place to share your files or the URL shortener. |
| 14:29.55 | duchman | If in doubt, check Slepp's page for contact information. |
| 14:29.56 | *** join/#asterisk serafie1 (~erin@nat/digium/x-bskjgiuwvhihdbxm) |
| 14:30.18 | duchman | any other pastebin link? |
| 14:30.25 | *** join/#asterisk upp (928c103e@gateway/web/freenode/ip.146.140.16.62) |
| 14:31.16 | upp | [TK]D-Fender:i'm back here it's http://pastebin.com/CAKKCzrm |
| 14:31.50 | *** join/#asterisk deegen (~deegen@S01060023bee90320.gv.shawcable.net) |
| 14:33.08 | [TK]D-Fender | upp : 001 1011 27 Destination out of order |
| 14:34.27 | [TK]D-Fender | DAHDI/g0/0662118389 <- is this number supposed to be valid via your telco? |
| 14:34.52 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:34.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:36.09 | upp | [TK]D-Fender: yes, my Problem is i can't receive calls |
| 14:36.34 | *** join/#asterisk duchman (~paulo@41.190.3.33) |
| 14:36.59 | [TK]D-Fender | upp: then why have you wasted my time showing me an OUTBOUND CALL? |
| 14:37.45 | upp | because you asked me it befor |
| 14:38.36 | [TK]D-Fender | [09:40][TK]D-Fenderupp: show an actual call attempt at verbose 10 |
| 14:39.04 | [TK]D-Fender | I want to see the call in the direction you WERE ASKING ABOUT |
| 14:39.05 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:39.42 | [TK]D-Fender | I did not say to change what you were dubugging |
| 14:41.12 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 14:41.46 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:42.39 | qakhan | hi all. i am getting records from MSSQL in dialplan. i am getting top 5 records from DB according to the callerid. how can i get each record in each row |
| 14:43.37 | qakhan | here is my dial plan exten => 3000,n,Read(callerid) |
| 14:43.37 | qakhan | exten => 3000,n,SET(${callerid}=${ODBC_sql(${vPhoneNo})}) |
| 14:43.43 | [TK]D-Fender | read 5 rows |
| 14:43.59 | [TK]D-Fender | get the next record. |
| 14:44.24 | qakhan | like this? exten => 3000,n,SET(${callerid}=${ODBC_sql(${vPhoneNo})}) |
| 14:44.32 | [TK]D-Fender | you should already know the command to have read the first. |
| 14:44.39 | [TK]D-Fender | so read MORE |
| 14:46.52 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 14:47.49 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:48.50 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 14:55.38 | qakhan | [TK]D-Fender You know, you never helped me. you always pushed me to learn myself. Therefore i like you. |
| 14:55.48 | qakhan | Thanks for your help :) |
| 14:56.30 | [TK]D-Fender | your set is wrog |
| 14:56.46 | [TK]D-Fender | you do not put ${} on the LEFT side |
| 14:56.49 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:57.04 | [TK]D-Fender | youyou put ONLY the variable name there |
| 14:57.19 | [TK]D-Fender | you don't seem to understand basic variables yet |
| 14:57.24 | [TK]D-Fender | ~book |
| 14:57.24 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:57.42 | qakhan | i did put ${} |
| 14:58.08 | [TK]D-Fender | learn th e difference between passing the name to set ... and referncing the VALUE held by the variable |
| 14:58.17 | [TK]D-Fender | DO NOT DO THAT |
| 14:58.28 | [TK]D-Fender | NO ${} ON THE LEFT SIDE |
| 14:59.58 | qakhan | you mean callerid |
| 15:00.15 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:00.25 | [TK]D-Fender | YES |
| 15:00.56 | [TK]D-Fender | and you weren't showing the funtion that READS the record in the first place |
| 15:02.57 | qakhan | which one is that function? |
| 15:05.37 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 15:06.04 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:07.50 | duchman | <[TK]D-Fender>, here is the dialplan http://pastebin.ca/2428821 |
| 15:19.24 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 15:19.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:23.34 | wdoekes | duchman: you shan't copy-paste your editor window |
| 15:23.45 | wdoekes | use: cat /etc/asterisk/extensions.conf |
| 15:23.53 | duchman | ok |
| 15:27.18 | duchman | <wokes> http://pastebin.ca/2428823 |
| 15:30.23 | *** join/#asterisk HannibalS (~Red@84.118.2.206) |
| 15:31.31 | [TK]D-Fender | duck that is freepbx dialplan and only a tiny fraction of it and a waste of time. |
| 15:32.04 | [TK]D-Fender | duchman: linphone seems to be deciding to hang up immediately after, not asterisk |
| 15:32.12 | [TK]D-Fender | get another client |
| 15:32.41 | duchman | ok let me get 3cx |
| 15:33.00 | [TK]D-Fender | qakhan: YOU are the one using the SQL calls. how do you not knoe ehich one is the READ RECOD one? |
| 15:33.21 | [TK]D-Fender | record* |
| 15:35.18 | duchman | <[TK]D-Fender> http://pastebin.ca/2428825 is the link for 3cx |
| 15:35.42 | duchman | hangs up as soon as receiver picks up |
| 15:36.11 | [TK]D-Fender | that is not a complete call |
| 15:36.21 | [TK]D-Fender | get the WHOLE CALL |
| 15:37.17 | duchman | ok pls hold |
| 15:39.27 | Chainsaw | *tinny Cisco music* |
| 15:40.01 | duchman | <[TK]D-Fender> http://pastebin.ca/2428828 is the link from asterisk full message log(1000 lines) |
| 15:40.12 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:40.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:41.23 | [TK]D-Fender | it isn't |
| 15:41.27 | [TK]D-Fender | t all |
| 15:41.40 | [TK]D-Fender | tALL OF IT. |
| 15:42.35 | [TK]D-Fender | fcrom the fikrst leg cominng in throught the dialplan prossing throu the Dial() that should issue all th way to then end, |
| 15:42.45 | [TK]D-Fender | processing |
| 15:46.04 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:46.07 | *** join/#asterisk _zerick_ (~eocrospom@190.187.21.53) |
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| 15:55.53 | pensmit | AsteriskNow or FreePBX distro? Which is better and why? |
| 15:57.06 | jmetro | pensmit: are you trying to learn asterisk? |
| 15:57.43 | pensmit | Not at this point. I'm trying to get a distribution that works well with conferences and updates easily |
| 15:58.18 | jmetro | freePBX seems to be the preferred GUI, but if you ever want to do anything custom or learn asterisk, stop thinking about GUI's and just start with vanilla |
| 15:59.49 | pensmit | I'm trying to get a good take on one distribution over the other. |
| 16:00.10 | pensmit | AsteriskNOW and FreePBX distro are both distributions that don't require compiling |
| 16:00.27 | pensmit | The update process is hopefully painless and simple. |
| 16:00.50 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:01.53 | igcewieling | pensmit: We don't support AsteriskNOW nor FreePBX here, so it may not be the best place to ask. |
| 16:02.02 | jmetro | FreePBX seems to be the most popular |
| 16:02.22 | [TK]D-Fender | This is already answered in #freepbx |
| 16:02.29 | jmetro | and from what i've seen of both, i'd rather go with freePBX if i really wanted to gimp my possibilities. |
| 16:03.06 | *** join/#asterisk navaismo (~navaismo@189.241.66.140) |
| 16:03.18 | Greenlight | Doesn't asteriskNow just package up freepbx anyway? |
| 16:03.21 | igcewieling | FreePBX isn't too terrible. They made some design decisions I feel are totally wrong, but it does work well. |
| 16:03.44 | igcewieling | Greenlight: then why use it instead of FreePBX |
| 16:04.07 | [TK]D-Fender | pensmit: and you should get VERY specefic about your needs for "works well with conferences" |
| 16:04.07 | Greenlight | Ease of install |
| 16:04.27 | [TK]D-Fender | same thing really |
| 16:04.30 | Greenlight | It installs everything. For a non-linux person, I guess it can get them started playing with Asterisk quickly. |
| 16:04.55 | jmetro | trixbox for the ultimate in quick installs! Lol |
| 16:04.59 | Greenlight | ... |
| 16:05.03 | ABerrios | icks |
| 16:05.44 | *** join/#asterisk skorzen (~skorzen@192.199.18.73) |
| 16:05.47 | skorzen | Hello guys. |
| 16:05.56 | skorzen | Anyone here using nagios to monitor asterisk remote SIP peer? |
| 16:08.54 | *** join/#asterisk modesto916 (~modesto@189-90-192-72.isimples.com.br) |
| 16:12.07 | igcewieling | Why would a non-linux person expect they could manage a PBX? |
| 16:12.40 | Greenlight | Perhaps it would be an incentive to go and learn :) |
| 16:12.51 | igcewieling | Greenlight: you don't some GUI for that. |
| 16:12.52 | Greenlight | A "hook" to get them in |
| 16:13.20 | Greenlight | Small steps ;) |
| 16:13.44 | igcewieling | In order to properly install, configure, and secure a VoIP PBX you need to know linux, networking, SIP, NAT, RTP, in addition to the actual PBX |
| 16:14.15 | jmetro | ^ |
| 16:14.49 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:15.05 | igcewieling | Not knowing those things makes about as much sense as skiing mount Hood with no training at all |
| 16:16.10 | Greenlight | People have got to start somewhere. I've always thought of AsteriskNOW as a simple way to do that. |
| 16:16.40 | jmetro | I started with the book |
| 16:16.45 | jmetro | \_o.o_/ |
| 16:16.47 | Greenlight | It's not going to make you able to deploy a production ready machine overnight, no, but if it gets more people using linux and asterisk, and encourages people to learn, that's a good think imo |
| 16:17.05 | [TK]D-Fender | I'd accept : expect to have to learn things fast ; don't bitch abaout that inevitabilkity |
| 16:17.51 | [TK]D-Fender | ESPECIALLY the "don't bitch" part. |
| 16:18.00 | Greenlight | I started with AsteriskNOW - I just wanted to get in there, get my hands dirty and see what it was all about |
| 16:18.08 | jmetro | phones are hugely complicated and theres a lot of things that are still mysteries to me, but i know computers and taught myself how to make it work cause thats what i do |
| 16:18.30 | jmetro | its all programming in one way or another. |
| 16:21.28 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:22.00 | *** join/#asterisk serafie (~erin@nat/digium/x-uiicotqiffjrijyy) |
| 16:23.41 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 16:24.27 | navaismo | someone need to add english subs to this-->http://vimeo.com/68252593 |
| 16:26.31 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:32.10 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 16:40.33 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:44.29 | *** join/#asterisk serafie (~erin@nat/digium/x-gekokzubzybfvenh) |
| 16:45.11 | *** join/#asterisk iGZo (~noc@ip72-207-28-75.sd.sd.cox.net) |
| 16:48.22 | iGZo | Are there any known issues with 11.5? We have hundereds of phones offline after an upgrade (11.4 -> 11.5). Certain extensions still register, but other simply cannont register at all. |
| 16:49.11 | jmetro | iGZo: what are they doing? I had an issue where i set my reg time to 0 on accident and DDOS'd my aastras, they would register but then eventually freeze. |
| 16:50.20 | iGZo | jmetro: we have tons of VMs running Elastix, and the ones that updated lastnight to 11.5 are offline now. |
| 16:51.32 | jmetro | I dont know anything about elastix, is that a gui? Can you see the error output in the asterisk console? |
| 16:52.00 | iGZo | Yeah, Elastix is just a gui. There are no messages. |
| 16:52.15 | jmetro | did you core set debug 999 core set verbose 999 |
| 16:52.21 | navaismo | maybe its about the res_rtp engine as usual with 11.5 |
| 16:52.44 | iGZo | yeah? |
| 16:52.53 | iGZo | (trying the debug now) |
| 16:53.32 | jmetro | I know D-fender prefers "sip set debug on" but i cant wade through debug logs all day. You should be able to at least see if the phone is touching asterisk with debug and verbose set high |
| 16:54.25 | *** join/#asterisk zeroschism (ajs07635@147.134.4.74) |
| 16:54.31 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 16:55.34 | *** join/#asterisk NicoR (5046d082@gateway/web/freenode/ip.80.70.208.130) |
| 16:55.41 | Greenlight | I had this the other day, was missing uuid-devel iirc |
| 16:55.46 | Greenlight | With 11.5 |
| 16:55.48 | iGZo | yeah, absolutely nothing. (except for the Remote Unix connection messages) |
| 16:56.06 | iGZo | uuid-devel? |
| 16:56.09 | Greenlight | "core show module like rtp" |
| 16:56.13 | Greenlight | What does that say |
| 16:57.13 | iGZo | no such command |
| 16:57.35 | iGZo | there is no "module" listed after show |
| 16:57.41 | Greenlight | sorry, "modules show like rtp" |
| 16:57.50 | iGZo | ok |
| 16:57.51 | Greenlight | "module show like rtp" |
| 16:57.52 | Greenlight | even |
| 16:57.57 | qakhan | [TK]D-Fender can you please look into my dialplan |
| 16:57.58 | qakhan | http://pastebin.com/n820h1Fe |
| 16:57.59 | Greenlight | I need caffine |
| 16:58.19 | iGZo | chan_multicast_rtp.so Multicast RTP Paging Channel 0 |
| 16:58.22 | iGZo | res_rtp_asterisk.so Asterisk RTP Stack 5 |
| 16:58.25 | iGZo | res_rtp_multicast.so Multicast RTP Engine 0 |
| 16:58.28 | iGZo | res_srtp.so Secure RTP (SRTP) 0 |
| 16:58.59 | Greenlight | Hmm it's not the issue I observed then ... as you have res_rtp_asterisk |
| 16:59.10 | Greenlight | So, youe phones register or not? |
| 16:59.24 | iGZo | 2 out of 20 are registered |
| 16:59.27 | igcewieling | qakhan: did you have a specific question about your dialplan? |
| 17:00.02 | Greenlight | HOw odd - and the 18 that aren't registering.. seeing any errors ? |
| 17:00.22 | iGZo | Greenlight: that's the thing, it's like they are being ignored. |
| 17:00.51 | [TK]D-Fender | <PROTECTED> |
| 17:01.22 | igcewieling | iGZo: have you used tcpdump or tshark to verify the packets are even arriving on the server? |
| 17:01.30 | iGZo | I basically get a 408, request timed out when trying to connect. |
| 17:01.40 | qakhan | what happened? |
| 17:01.43 | iGZo | igcewieling: I can, let me check. |
| 17:05.09 | [TK]D-Fender | qakhan: you aren't even looking at what you are doing. |
| 17:06.17 | [TK]D-Fender | qakhan: You use a Read() to get user input into a variable an then you KILL that value in the very next line wit that Set() |
| 17:06.45 | iGZo | igcewieling: so it looks like we are getting "401 unauthorized" |
| 17:07.30 | igcewieling | iGZo: you will always get a 401 on the first packet of a transaction, the 2nd request will contain the encrypted password. |
| 17:07.36 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 17:07.55 | igcewieling | check sip debug and make sure you don't have some silly NAT issue which is making Asterisk reject the packets |
| 17:07.57 | *** join/#asterisk emilv (emilv@emilv.lowend.io) |
| 17:08.16 | iGZo | igcewieling: Ill check |
| 17:09.06 | emilv | hello, i'm having problem regarding call forward, i need to forward calls to my mobile after certain time of day. the problem is that when it happens it sends the mobile number back to the trunk which he doesnt allow, how can i get this to work? |
| 17:10.21 | igcewieling | emilv: I doubt that is the case. I suspect the original callerid is being passed, which makes it reject. Set the CALLERID(num) to something your carrier accepts before dialing. |
| 17:10.25 | [TK]D-Fender | then forward to an extension that will dial out another provider |
| 17:11.23 | emilv | should i replace this: ${CALLERID(num)} with an actual phonenumber? |
| 17:11.35 | emilv | or put the number inside the (num)? |
| 17:12.15 | igcewieling | emilv: apparently your first step is to learn Asterisk. |
| 17:12.17 | igcewieling | ~book |
| 17:12.18 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:12.45 | *** join/#asterisk navaismo (~navaismo@189.241.46.165) |
| 17:13.18 | jmetro | I honestly cant find any good documentation on musiconhold.conf |
| 17:14.20 | igcewieling | jmetro: musiconhold.conf.sample ? |
| 17:14.40 | qakhan | [TK]D-Fender ok i got you. i changed it. but it is still not working. |
| 17:14.41 | qakhan | http://pastebin.com/89pNRTmc |
| 17:15.02 | jmetro | igcewieling: I found that yes, but there is something up with my MOH for a specific client that is causing the files to always play in the same order |
| 17:15.07 | jmetro | which means they only ever hear the first track |
| 17:15.20 | igcewieling | jmetro: there is a setting for that. |
| 17:15.26 | jmetro | is it sort=random? |
| 17:16.04 | igcewieling | jmetro: no. |
| 17:17.33 | igcewieling | There is some option to make Asterisk restart MoH from the beginning for each call .vs. one MoH process for all callers |
| 17:18.39 | jmetro | well i know there is the "stream file" |
| 17:18.55 | jmetro | but i have other clients setup with 5-10 files in their MOH folder and asterisk randomly chooses one |
| 17:19.37 | igcewieling | ;cachertclasses=yes ; use 1 instance of moh class for all users who are using it, decrease consumable cpu cycles and memory disabled by default |
| 17:20.11 | jmetro | cachertclasses ? like.. "cache realtime classes" ? |
| 17:20.26 | igcewieling | that is what I was thinking of. I don't know if it would apply or not, but worth checking |
| 17:20.49 | [TK]D-Fender | qakhan: that error says it alll... |
| 17:21.12 | navaismo | what is better srtp or zrtp |
| 17:21.31 | igcewieling | navaismo: yes. |
| 17:21.43 | [TK]D-Fender | qakhan: [Aug 6 13:09:53] WARNING[4316]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '<=', expecting $end; Input: |
| 17:21.45 | [TK]D-Fender | <PROTECTED> |
| 17:21.47 | [TK]D-Fender | <PROTECTED> |
| 17:22.21 | [TK]D-Fender | qakhan: there in nothing on the LEFT side of youe expression so it fails |
| 17:22.35 | [TK]D-Fender | your* |
| 17:23.12 | qakhan | exten => 3000,n,While($[${COUNTER} <= ${ODBCROWS}]) |
| 17:23.20 | navaismo | so my question is wrong? |
| 17:23.47 | navaismo | which is better to use with asterisk zrtp or srtp? |
| 17:23.50 | *** join/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 17:24.03 | btcquant | Hello. Trouble with outbound e-mail of voicemaill. Asterisk is sending them from <asterisk@host> instead of <asterisk@host.com> . This is causing my mail service to reject the e-mail. I cant' find anywhere in the config to change that. (Using freepbx, but also looking at raw config files in /etc/asterisk) Any suggestions on where to update this? |
| 17:25.01 | [TK]D-Fender | qakhan: and whent that EVALUATES it comes back BLANK |
| 17:25.39 | qakhan | i didnt get you |
| 17:25.44 | [TK]D-Fender | qakhan: look at your call and what is actually getting set,. you are not even looking |
| 17:25.54 | [TK]D-Fender | COUNTER IS EMPTY |
| 17:26.57 | igcewieling | navaismo: which is better SIP or RTP? They do different things. STRP and ZRTP do different things. ZRTP CANNOT EVER be better unless BOTH your endpoints support ZRTP, for example. |
| 17:28.18 | igcewieling | SRTP is better because you only need the client and the server to support SRTP. No! ZRTP is better because bypasses the server for encryption so the server can't be used to comprimize the connection. |
| 17:28.20 | igcewieling | etc. |
| 17:29.00 | navaismo | ok thx |
| 17:29.06 | skorzen | btcquant, /etc/postfix/main.cf |
| 17:29.11 | skorzen | check that file for SMTP configuration. |
| 17:29.19 | *** part/#asterisk emilv (emilv@emilv.lowend.io) |
| 17:29.39 | btcquant | skorzen Don't think that's it. The e-mail comes form <asterisk@host> That should be set by asterisk when sending mail?? |
| 17:30.20 | skorzen | btcquant, which mailer are you using then? |
| 17:30.25 | btcquant | postfix |
| 17:30.29 | igcewieling | btcquant: e-mail has MULTIPLE from addresses |
| 17:30.30 | skorzen | So, look there. |
| 17:30.57 | qakhan | Thanks [TK]D-Fender i got my mistake |
| 17:31.03 | skorzen | You should have a parameter called mydomain. |
| 17:31.17 | qakhan | exten => 3000,n,Set(COUNTER=$[${COUNTER + 1}]) |
| 17:31.47 | [TK]D-Fender | qakhan: again you have a real problem understanding braces... |
| 17:32.00 | btcquant | skorzen That worked. Last place I would have thought to look. THANKS!! |
| 17:32.09 | [TK]D-Fender | ${} is fot the variable name only |
| 17:32.12 | qakhan | this line was wrong } should be next to COUNTER |
| 17:32.22 | [TK]D-Fender | You do not put math junk in there |
| 17:32.36 | qakhan | anyhow Thanks alot |
| 17:32.38 | qakhan | :) |
| 17:32.58 | skorzen | great, btcquant :) |
| 17:33.11 | qakhan | you helped me and we resolved the issue :) |
| 17:34.00 | *** join/#asterisk mitchrodrigues (~mitchrodr@38.111.144.81) |
| 17:34.08 | qakhan | here is another question |
| 17:35.09 | qakhan | currently we are getting records in COUNTER veriable, can i save each record in differnet veriable? |
| 17:37.13 | btcquant | Noticed an interesting bug. When e-mailing voicemails, they come through, but the link to visit is set to "http://AMPWEBADDRESS/recordings/index.php " AMPWEBADDRESS is set in the config files. So it looks like Asterisk isn't substituting that in correctly. |
| 17:39.35 | navaismo | ampwebaddress is a freepbx stuff not asterisk |
| 17:41.31 | [TK]D-Fender | qakhan: I don't see your current code and your current call. |
| 17:43.02 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:43.11 | *** join/#asterisk skorzen (~skorzen@188.140.95.251) |
| 17:43.39 | qakhan | ok |
| 17:43.44 | qakhan | let me show you |
| 17:44.03 | [TK]D-Fender | qakhan: and look at it YOURSELF |
| 17:46.14 | qakhan | http://pastebin.com/2dNdC5N4 |
| 17:46.15 | qakhan | here |
| 17:47.08 | btcquant | Anybody have experience with SNOM phones? I think I killed mine. |
| 17:47.31 | [TK]D-Fender | so look at what you are setting... |
| 17:48.11 | [TK]D-Fender | qakhan: you can't have a variable name start with a number. that is illegal. |
| 17:48.21 | *** join/#asterisk skorzen (~skorzen@192.199.18.73) |
| 17:49.22 | igcewieling | [TK]D-Fender: you are writing it for him. |
| 17:49.39 | [TK]D-Fender | igalmost... not quite there... |
| 17:49.48 | [TK]D-Fender | igcewieling: almost... not quite there... |
| 17:50.39 | qakhan | [TK]D-Fender where i used number instead if name in veriable? |
| 17:51.10 | [TK]D-Fender | [13:48][TK]D-Fenderqakhan: you can't have a variable name start with a number. that is illegal. |
| 17:51.55 | qakhan | thats what i am askig |
| 17:52.00 | qakhan | i didnt use it |
| 17:52.36 | [TK]D-Fender | look at your call. you are setting illegal variables |
| 17:53.15 | [TK]D-Fender | qakhan: Whick doesn't really matter in the end because you aren't even doing anything with them |
| 17:53.22 | [TK]D-Fender | which |
| 17:54.36 | qakhan | can you plz point out where is that, i dont see |
| 17:54.46 | [TK]D-Fender | look at your call. you are setting illegal variables |
| 17:54.47 | [TK]D-Fender | ^ |
| 17:54.53 | [TK]D-Fender | look at your SETS |
| 17:55.20 | igcewieling | qakhan: line 32 in your pastebin |
| 17:55.24 | igcewieling | sorry, 38 |
| 17:56.45 | qakhan | 3=Welcome BIJAYA MISHRA 92712 Your Pickup address is 4409 FORBES BLVD Lanham MD 20706" |
| 17:56.48 | qakhan | this one? |
| 17:57.23 | igcewieling | and the others which do the same thing. you are setting the variable named "3" to the value "Welcome BIJAYA MISHRA 92712 Your Pickup address is 4409 FORBES BLVD Lanham MD 20706" and YOU CANNOT DO THAT |
| 17:57.39 | qakhan | its coming from exten => 3000,n,Set(${COUNTER}=${ODBC_FETCH(${ODBC_ID})}) |
| 17:57.57 | qakhan | there is while loop |
| 17:58.02 | igcewieling | Yes, I know. It is doing exactly what you are telling it. |
| 17:58.06 | [TK]D-Fender | qakhan: you can't even tell which line is line # 38? how is that a question? |
| 17:58.15 | [TK]D-Fender | qakhan: 3= is BAD |
| 17:58.37 | [TK]D-Fender | qakhan: 3 is NOT a valid name to use for a variable. |
| 17:58.50 | qakhan | you guys can see my dialplan |
| 17:58.58 | [TK]D-Fender | qakhan: How many moe times will it take for you to understand? |
| 17:59.08 | [TK]D-Fender | qakhan: more* |
| 17:59.19 | [TK]D-Fender | [13:48][TK]D-Fenderqakhan: you can't have a variable name start with a number. that is illegal. |
| 17:59.21 | [TK]D-Fender | ^^^^^^^^^^^ |
| 17:59.29 | qakhan | 3 is coming from while loop |
| 17:59.31 | [TK]D-Fender | 3= <--------------- BAD |
| 17:59.39 | qakhan | i m not setting it up |
| 18:00.07 | [TK]D-Fender | no, you are trying to use "3" as a variable name. YOU CANNOT DO THIS, IT IS ILLEGAL |
| 18:00.07 | qakhan | i got you 3 is BAD but i am not putting it in dialplan |
| 18:00.23 | qakhan | why dont you see my config |
| 18:00.30 | qakhan | please see |
| 18:00.36 | [TK]D-Fender | YOU are becase that 3 is THE VALUE OF THAT counter. |
| 18:01.03 | igcewieling | qakhan: ${COUNTER} is 3. Therefore 3 is the name of the variable you are setting |
| 18:01.25 | igcewieling | perhaps you might try something like Set(COUNTER=${ODBC_FETCH(${ODBC_ID})}) |
| 18:01.38 | [TK]D-Fender | igno.... |
| 18:01.43 | [TK]D-Fender | igcewieling: no... |
| 18:01.59 | qakhan | ok please help me how i can stop this 3 |
| 18:02.18 | igcewieling | qakhan: have you read ANY sample and example dialplans written by anyone else? Have you read the Asteirsk book. |
| 18:02.32 | [TK]D-Fender | qakhan: why are you using that variable there? thi is YOUR fault. Stop using that variable that way |
| 18:03.03 | qakhan | i copied this example from asterisk book |
| 18:03.13 | igcewieling | qakhan: Do you understand that ${COUNTER} means "obtain the value of the variable counter and use that" |
| 18:03.21 | igcewieling | qakhan: no, you did not. What page of the Asterisk book? |
| 18:04.00 | slav3_kitten | the asterisk book a a tough read igcewieling, if they added like a secret plot by an off world mega corporation to dominate the phone market and rule it with an iron fist. then the asterisk group eventually beats them during a hard fought war over 3 planets... it'd be easier to read... |
| 18:04.01 | [TK]D-Fender | qakhan: you deleted OTHER stuff you had there before and don't even remember |
| 18:04.06 | qakhan | http://pastebin.com/MYmMGJrL |
| 18:04.08 | qakhan | here |
| 18:04.35 | [TK]D-Fender | qakhan: you deleted OTHER stuff you had there before and don't even remember <------------ |
| 18:04.40 | iGZo | igcewieling: looks like a nat issue/setting in FreePBX, we've had it set as "no" but after the update we had to change it to "yes". The phones are back online. |
| 18:04.42 | [TK]D-Fender | qakhan: LINE 7 |
| 18:04.47 | igcewieling | qakhan: I see no line with a Set(${COUNTER}= only lines with SET(COUNTER= |
| 18:04.57 | [TK]D-Fender | qakhan: they are NOT THE SAME |
| 18:05.26 | slav3_kitten | i think the evil mega corp should be called ocsic |
| 18:05.35 | qakhan | same => n,Set(AVAIL_EXTEN_${COUNTER}=${ODBC_FETCH(${ODBC_ID})}) |
| 18:05.52 | [TK]D-Fender | qwawhat does YOURS now have? |
| 18:06.03 | jmetro | for Cachertclasses=yes what would I reload for that? [MOH conversation from earlier] |
| 18:06.10 | igcewieling | qakhan: right. so that evaluates to Set(AVAIL_EXTEN_[the value of the variable counter, which might be 3]=whatever |
| 18:06.22 | igcewieling | jmetro: I'd restart asterisk |
| 18:06.27 | jmetro | igcewieling: crap |
| 18:06.33 | qakhan | mine is this exten => 3000,n,Set(${COUNTER}=${ODBC_FETCH(${ODBC_ID})}) |
| 18:06.35 | [TK]D-Fender | qakhan: where are those extra letters in YOUR code? |
| 18:06.35 | igcewieling | but you can try something line "moh reload" |
| 18:06.40 | [TK]D-Fender | qakhan: where are those extra letters in YOUR code?<----- |
| 18:06.47 | jmetro | igcewieling: i did moh and features reload |
| 18:06.54 | [TK]D-Fender | qakhan: AVAIL... <----- |
| 18:06.54 | igcewieling | War_Bear_away: so now you have Set([the value of the variable counter, which might be 3]=whatever |
| 18:07.03 | igcewieling | which is obvously not valid |
| 18:07.17 | qakhan | i didnt know about AVAIL_EXTEN_ |
| 18:07.28 | qakhan | do i need to add this AVAIL_EXTEN_ |
| 18:07.39 | igcewieling | [TK]D-Fender: I believe he is beyond my help and beyond even your help. Give up. I am. |
| 18:07.46 | protocoldoug | I want to play an accouncement to the called party, and I know about the "A" option on Dial() however, I don't want there to be dead air for the calling party while the announcement is played -- can I play a message to both calling and called parties when the line is picked up by the called party? |
| 18:07.57 | [TK]D-Fender | qakhan: "didn't know"? you can't even compare 2 lines and see a huge difference |
| 18:08.14 | protocoldoug | announcement* above, typo. |
| 18:08.50 | igcewieling | qakhan is too much for me. BBIAW |
| 18:08.53 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 18:09.04 | protocoldoug | Errr, different messages to each side, that is. Or anything to prevent the caller from hearing "ring, ring ring.... {silence while announcement is played} 'Hello?'" |
| 18:10.46 | protocoldoug | My current thinking to overcome this hurdle is in this situation, is to originate a call with a call file... And treat each channel seperately, and finally when the announcement is finished for the called party -- then Bridge() the channels together |
| 18:11.08 | protocoldoug | But, it's a lot more work, and... Wanted to sanity check that I'm not missing something "big" |
| 18:12.22 | qakhan | [TK]D-Fender i have to go now |
| 18:12.34 | qakhan | i will talk to you tomorrow |
| 18:12.47 | qakhan | if you could help me |
| 18:13.00 | [TK]D-Fender | qakhan: hire a programmer. You have no idea what you are doing |
| 18:13.18 | [TK]D-Fender | qakhan: and it's been years |
| 18:13.49 | qakhan | its been 2 years :P |
| 18:14.52 | qakhan | i am getting veribale with names now |
| 18:15.08 | qakhan | Set("SIP/3288-00000082", "AVAIL_EXTEN_1=Welcome TEST Your Pickup address is 19290 MONTGOMERY VILLAGE AVE Montgomery Village MD 20886 |
| 18:16.03 | protocoldoug | ...I hope those aren't addresses of your subscribers, because... "that ain't cool bro." |
| 18:18.12 | [TK]D-Fender | qakhan: And from what I've seen .... you're doing nothing with them |
| 18:18.17 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 18:20.01 | *** join/#asterisk The_Phil (~Phizzel@41-133-163-63.dsl.mweb.co.za) |
| 18:20.08 | *** join/#asterisk aruntomar (~Thunderbi@49.248.153.130) |
| 18:24.58 | The_Phil | Hello Gents |
| 18:28.59 | The_Phil | can someone please tell me what this message means? |
| 18:29.08 | The_Phil | WARNING[5834] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 18:30.39 | navaismo | No route to destination) |
| 18:30.44 | The_Phil | lol |
| 18:30.46 | The_Phil | I know that |
| 18:30.50 | navaismo | so? |
| 18:31.04 | navaismo | your actual question is?... |
| 18:31.06 | The_Phil | I more want to know what can be the possible cause of the error |
| 18:31.12 | The_Phil | for example |
| 18:31.34 | The_Phil | We have server b which is registered to server b |
| 18:31.39 | The_Phil | b to c * |
| 18:31.52 | The_Phil | so someone registers to b, which then hands the call to c |
| 18:32.09 | navaismo | no route to destination in other words means asterisk can contact your peer maybe because isn't registered, qualified or something I guess |
| 18:32.19 | The_Phil | ahhhhhhhh |
| 18:32.25 | The_Phil | now, comes my next question |
| 18:32.40 | navaismo | can't* |
| 18:32.51 | The_Phil | is there any way to see which peer it is that the error message is related to? |
| 18:33.40 | The_Phil | I basically want to find out how we can narrow the problem down. Is there an error log somewhere in asterisk that will show which peer/call/ip it is related to? |
| 18:35.56 | The_Phil | that's actually another thing I wan't to know. Is there a way to link those WARNING/ERROR/NOTICE messages to a specific instance without having to switch on verbose? |
| 18:36.06 | [TK]D-Fender | The_Phil: obviously the one you just trie dialing there |
| 18:36.09 | jmetro | run with verbose on all the time. |
| 18:36.14 | [TK]D-Fender | tried |
| 18:36.37 | [TK]D-Fender | "core set verbose 10" |
| 18:36.42 | The_Phil | wahahahahaha |
| 18:36.50 | The_Phil | with 125 concurrent calls? |
| 18:37.00 | jmetro | i prefer 999 on verbose and debug |
| 18:37.04 | The_Phil | awesome |
| 18:37.12 | [TK]D-Fender | The_Phil: yes |
| 18:37.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.41) |
| 18:37.25 | jmetro | it could be worse, i know people [i think on here] that run "sip set debug on" for the same amt of calls |
| 18:38.01 | The_Phil | Luckily all the UA's registering to the server have their own IP addresses and there are not two alike at any time |
| 18:38.07 | The_Phil | oh |
| 18:38.12 | The_Phil | never mind |
| 18:38.15 | The_Phil | lol |
| 18:38.22 | *** join/#asterisk bulkorok (~chatzilla@053d9363.dynamic.tele-ag.de) |
| 18:38.42 | [TK]D-Fender | the peer you are dialing has NOT registered and your server does NOT have an IP to call them at |
| 18:39.03 | The_Phil | thanks, that clears it up :) |
| 18:39.05 | The_Phil | so, is there a way to link the warning/error/notice messages to a peer/IP/call ID? |
| 18:39.21 | The_Phil | without going core set verbose 999 ? |
| 18:39.41 | [TK]D-Fender | no. that line shows what it shows... |
| 18:39.58 | [TK]D-Fender | you need the originating dialplan lines |
| 18:40.10 | [TK]D-Fender | and that requires verbose |
| 18:40.14 | The_Phil | so, what is that 5834 next to the warning? |
| 18:40.30 | jmetro | error code |
| 18:40.54 | The_Phil | is that related to the server or is it asterisk in general |
| 18:41.16 | [TK]D-Fender | it is nothing of use |
| 18:41.34 | *** join/#asterisk outtolunc (~me@50-193-41-213-static.hfc.comcastbusiness.net) |
| 18:41.46 | The_Phil | damn, I thought as much |
| 18:42.42 | jmetro | hm. What exactly are those anyway. |
| 18:45.05 | jmetro | on a related note, i wish i could accept tacos as payment. |
| 18:45.58 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.40) |
| 18:46.10 | The_Phil | I don't know, and no amount of googling can clarify the matter :( |
| 18:46.21 | The_Phil | jmetro, I'll fax you one? |
| 18:46.41 | The_Phil | a taco, that is |
| 18:47.08 | jmetro | Am i going to have to run a couple pages through my cross shredder in order to get the cheese? |
| 18:48.00 | The_Phil | haha, indeed |
| 18:48.26 | The_Phil | the mince I'll send via email |
| 18:48.34 | *** join/#asterisk samy_ (~samy@namb.la) |
| 18:49.18 | jmetro | Ahh tacos, my first love. |
| 18:49.44 | The_Phil | yep. The one that got away... |
| 18:51.08 | jmetro | Oh hell no, i head home early if i know tacos are on the stove. |
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| 20:07.33 | *** join/#asterisk uyulala (~IceChat9@2-231-231-43.ip209.fastwebnet.it) |
| 20:07.41 | uyulala | hi all |
| 20:07.50 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:08.28 | uyulala | could I ask a little question to you, gurus? |
| 20:08.45 | uyulala | :) |
| 20:09.36 | jmetro | ~as |
| 20:09.37 | infobot | i guess as is the tranny. so swapping to a v8 and t5 doesn't add as much weight as you'd expect. |
| 20:09.44 | jmetro | ~ask |
| 20:09.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 20:11.06 | uyulala | There is a way to set in the sip.conf a path with a CRL in order to revoke a TLS certificate? |
| 20:12.07 | uyulala | (as you see is a simple but with no replies on the forums ecc..) |
| 20:12.58 | igcewieling | uyulala: if there is it should be documented in sip.conf.sample, included in your Asterisk source code. |
| 20:13.54 | uyulala | no there isn't any kind of documentation about revoking a certificate, only on how to generate it |
| 20:15.05 | igcewieling | then it is unlikely to be something Asterisk supports |
| 20:16.23 | uyulala | I hope someone that found a way to it, revoke a certificate, even if not in a standard way (with the crl) |
| 20:17.54 | uyulala | however thank you for your answer |
| 20:19.07 | *** join/#asterisk mitchrodrigues (~mitchrodr@38.111.144.81) |
| 20:20.12 | mitchrodrigues | Anyone recommend a pressence server to use |
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| 20:44.15 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 20:44.43 | Micc | anyone know of a soft phone that will open a url on an incoming call and allow passing information to the url like caller id? |
| 20:44.56 | protocoldoug | With Dial()'s G(context^ext^priority) - where it sends the calling party to that extension and the called party to that extension + 1 -- is it possible to then put the two channels back together to talk? (I want to play a different message to each side) |
| 20:45.08 | Micc | I thought they all had something like that, but I can't find any docs on jitsi or zoiper that say how to pass parameters. |
| 20:45.32 | Micc | I know this is a bit off topic, but I doubt there is a better channel for my question. |
| 20:47.18 | paulc | Micc: I know you can do something like that with a Yealink hard phone, but haven't seen it in any softphones I've dealt with.. |
| 20:47.24 | igcewieling | Micc: you can set the callerid in Asterisk before you send the call to the endpoint |
| 20:47.28 | paulc | mitchrodrigues: openfire works well |
| 20:47.50 | mitchrodrigues | sweet ill take a look at it :D |
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| 20:55.56 | Micc | igcewieling, I don't want to set the caller id. I just want to open a web page and pass the caller id to the website when there is an incoming call. This is for my customers to integrate with their CRM. |
| 20:59.35 | navaismo | Micc, zoiper biz can do that |
| 21:00.25 | *** join/#asterisk blehxor (~blehxor@ops-nat-pool.ops.expertcity.com) |
| 21:01.24 | navaismo | or you can try this program --->http://asterisktools.blogspot.mx/2012/08/call-monitor-asterisk.html |
| 21:02.56 | WIMPy | That's how we did it 20 years ago, isn't it? |
| 21:03.55 | *** join/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-qbiwdxqytrdacpxj) |
| 21:04.55 | navaismo | right |
| 21:05.24 | WIMPy | still does it that way. |
| 21:05.31 | *** join/#asterisk abradley (~adam@sentry-buick.cpe.newsouth.net) |
| 21:06.16 | Micc | navaismo, zoiper biz does do it, but I can't figure out the parameter for caller id. |
| 21:06.16 | navaismo | Using openvpn to connect to asterisk, the peer is natted or not? My mind is fuc*** rifght now my peer cant dial anything via openvpn+tls transport |
| 21:06.41 | Micc | I like the call monitor, but I don't want to allow AMI access to my server. |
| 21:06.53 | WIMPy | navaismo: Depends on how you set it up. |
| 21:07.10 | WIMPy | Micc: You don't have to. |
| 21:07.29 | WIMPy | Put a little sender script in to your dialplan via System(). |
| 21:07.51 | WIMPy | Or do your own thing that listens on AMI and sends out the information. |
| 21:07.56 | navaismo | WIMPy, I added the TUN address to localnet and set the peer to nat=no |
| 21:09.10 | blehxor | Hi... newb here. Need to capture a SIP response code with dial - have read about ${HASH(SIP_CAUSE,${CHANNEL})}. When would I structure that to get a meaningful value? |
| 21:09.32 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
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| 21:09.41 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:10.23 | blehxor | *how would I structure the dial and SIP_CAUSE call to get a meaningful value* |
| 21:10.36 | navaismo | WIMPy, my peer registered successfully but when i dial nothing happens... |
| 21:10.50 | *** join/#asterisk petris (~petris@ryanpetris.com) |
| 21:10.59 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 21:13.49 | navaismo | Micc, in the zoiper manual explain how to set the cid: "• $(CALLERNAME) - this tag is replaced with the callername of the call. |
| 21:13.50 | blehxor | calling dial from local channel with: exten => _10XX,n,Dial(SIP/${var4}@${var3},30,gM(test)) and don't know how to get the SIP channel name to use ${HASH(SIP_CAUSE,${CHANNEL})} |
| 21:13.51 | navaismo | • $(CALLERNUMBER) - this tag is replaced with the callernumber of the call. |
| 21:13.51 | navaismo | • $(DNID)- this tag is replaced with DNID number of the call if it is incoming call. For |
| 21:13.51 | navaismo | outgoing calls there is no DNID. DNID is the number that the caller has dialed to call |
| 21:13.51 | navaismo | you. " |
| 21:14.17 | Micc | navaismo, thanks I couldn't find that. |
| 21:14.27 | navaismo | you need glasses |
| 21:14.30 | navaismo | :D |
| 21:16.18 | navaismo | blehxor, try with ${HASH(SIP_CAUSE,${CDR(dstchannel)})} |
| 21:16.40 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:22.27 | blehxor | thanks, navaismo. would I call that on the local channel after my dial (so it would trigger after the macro is finished and the dial call is done)? |
| 21:23.48 | navaismo | ah... hmm... I have that after the dial cmd |
| 21:24.02 | blehxor | ok I'll try that, thanks again |
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| 21:33.07 | jmetro | cats, im a kitty cat. |
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| 22:18.12 | *** join/#asterisk Merlin (merlin@evendata.net) |
| 22:18.35 | blehxor | I'm dialing a conference bridge with the dial cmd, then executing a macro which starts mixmonitor to record the call. Issue is the .wav file ends up missing the first ~5 seconds which is the bridge's automated message. Why would mixmonitor miss the beginning of the call like that? This is all local network - latency shouldn't be a factor... |
| 22:19.05 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 22:19.54 | Merlin | can anyone recommend a soho type router that has a built-in ipsec VPN client, plus collects stats on VoIP traffic, including mos/avg jitter/etc? |
| 22:20.48 | jmetro | <PROTECTED> |
| 22:20.52 | jmetro | but junipers are <3 |
| 22:21.08 | Merlin | aruba has the voip traffic stats, but you have to buy a $5k concentrator to get any vpn capability |
| 22:21.36 | [TK]D-Fender | blehxor: just initiate Mixmonitor before your dial. |
| 22:22.00 | [TK]D-Fender | you can set it to record only after it is answerew |
| 22:22.18 | asteriskmonkey | Merlin: use pfsense its free |
| 22:22.31 | asteriskmonkey | does a good job of vpn stuff |
| 22:23.06 | asteriskmonkey | you can probably also use some flavour of openwrt for what you need too |
| 22:23.17 | asteriskmonkey | that fits onalot of home class routers |
| 22:23.24 | Merlin | i know pfsense and openwrt, but i wasn't aware of a built-in voip traffic analysis module |
| 22:23.26 | blehxor | I'm not getting anything recorded when I try init MixMonitor before dial. |
| 22:23.50 | [TK]D-Fender | then something is wrong... |
| 22:23.53 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 22:24.40 | ageis | how to get the originating caller ID name (or a SIP peer name) as an environment variable in my outgoing context? |
| 22:24.46 | blehxor | Guessing local channel is getting "optimized out" when bridge happens? but don't know how to get around that - I'm a newb (obviously) |
| 22:24.51 | ageis | i.e., the person placing the call, not the exten dialed |
| 22:25.35 | Merlin | blehxor: mixmonitor is designed to deal with that circumstance |
| 22:27.21 | blehxor | hm |
| 22:28.21 | [TK]D-Fender | ageis: "core show function CALLERID" |
| 22:28.41 | ageis | [TL]D-Fender: no, that's a callerID of the called party |
| 22:28.54 | [TK]D-Fender | ageis: no, it isn't. |
| 22:29.18 | ageis | ok |
| 22:29.23 | asteriskmonkey | you can always set variables using the __ underscore to make them global too :) |
| 22:29.45 | asteriskmonkey | so if your playing context tag, you wont loose locally set ones |
| 22:29.52 | [TK]D-Fender | ageis: device calls *. that starts your dialplan. that is ONLY the caller so far ... LOOK AT THE CALLERID. Then Dial() |
| 22:29.53 | ageis | I had a problem where I was setting the CID on outgoing before I wanted to capture the person who was dialing |
| 22:30.00 | ageis | so didnt see that as useful or as my impediment |
| 22:30.01 | blehxor | would one of you mind taking a look at a few lines from my dial plan around that mixmonitor->dial? |
| 22:30.05 | ageis | ill jsut transfer it into another variable before I do that :) |
| 22:30.26 | [TK]D-Fender | Then you are setting before looking ... your mistake |
| 22:30.32 | [TK]D-Fender | indeed |
| 22:31.35 | ageis | ya, jsut realized |
| 22:33.40 | [TK]D-Fender | well you've hit your "silly" quota and realized what you needed almost on your own... I'd let this one slide :) |
| 22:35.41 | *** part/#asterisk samy_ (~samy@namb.la) |
| 22:42.08 | blehxor | my dial plan which isn't recording anything from the dial cmd: http://pastebin.com/Pbesq0Z8 thanks |
| 22:42.31 | ageis | heh.. I built this whole call recording system with mixmonitor and some scripts that feed calls into a n sqlite database and encode as mp3. |
| 22:42.58 | ageis | ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 1 |
| 22:43.05 | ageis | How can I trace this warning to a line in the dialplan? |
| 22:43.13 | *** join/#asterisk cmendes0101 (~chris@216.104.166.90) |
| 22:50.01 | [TK]D-Fender | actually have verbose enabled |
| 22:56.49 | blehxor | looks like mixmonitor seems to record fine if I remove the macro call within my dial cmd, is there something I'm missing there? |
| 22:57.20 | [TK]D-Fender | no... it works both ways... |
| 22:57.30 | [TK]D-Fender | hopefully a little faster outside |
| 23:00.38 | blehxor | actually its when I add an s(30) |
| 23:02.02 | blehxor | if I call a macro from dial, then give it a MACRO_RESULT=CONTINUE it should hang up the called party when the macro finishes right? |
| 23:05.23 | [TK]D-Fender | don't recall the specific values |
| 23:05.41 | [TK]D-Fender | read the instructions |
| 23:06.38 | blehxor | thats what the instructions say, but not what I'm seeing... |
| 23:07.05 | blehxor | or at least what I'm interpreting the instructions to say |
| 23:07.42 | CeBe | ageis: can you show whats in the line where error occurs? |
| 23:07.59 | CeBe | ageis: are you upgrading to a new version? |
| 23:09.08 | [TK]D-Fender | show the actual call |
| 23:09.29 | CeBe | ageis: had a similar problem today, see here http://www.voip-info.org/wiki/view/Asterisk+Expressions#Asterisk16Arithmetic since 1.6 you do not wrap dialplan functions in ${} anmore |
| 23:15.13 | *** join/#asterisk italorossi (~italoross@md85036d0.tmodns.net) |
| 23:16.19 | navaismo | WIMPy, if you have time can you take a look on this debug?-->http://pastebin.com/hsiA079d Asterisk receive the invite but then nothing happens not sure where lost the transaction if in phone or in server |
| 23:21.21 | navaismo | or anyone available to help |
| 23:24.32 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 23:27.02 | *** join/#asterisk djgerm (~Adium@207.111.228.20) |
| 23:27.24 | djgerm | hello! Is there a way to see current Calls Per Second in asterisk console? |
| 23:27.29 | djgerm | or in the previous second |
| 23:27.45 | djgerm | or over time |
| 23:27.48 | blehxor | [TK]D-Fender: dialplan with console, http://pastebin.com/SNkXr70y |
| 23:28.19 | blehxor | test-${call_start_org}.wav has nothing recorded in it. |
| 23:28.43 | blehxor | *console output |
| 23:29.00 | [TK]D-Fender | blehxor: bad approach overall |
| 23:31.10 | [TK]D-Fender | blehxor: use a local channel on one end and the playback, etc as regular dialplan |
| 23:31.42 | navaismo | djgerm, maybe: core show channels verbose help you |
| 23:32.09 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:33.08 | djgerm | navaismo: hmm that shows concurrent calls |
| 23:33.20 | djgerm | but not cps.... |
| 23:33.44 | [TK]D-Fender | there no cps |
| 23:35.41 | navaismo | so basically im seeing in the debug An invite from phone to asterisk, then UNAUTHORIZED from Asterisk to phone, then the Phone send a NOTIFY, asterisk respond to NOTIFY, Phone respond with the ACK of the previous INVITE. Then the phone try to register again, then a bunch of SUBSCRIBES transaction between phone and asterisk and done the INVITE is lost and phone hangup |
| 23:35.59 | blehxor | [TK]D-Fender: sorry, not exactly sure how I would structure it that way. I'm starting in a local channel then dialing out, do you mean I shouldn't use a macro? |
| 23:36.25 | [TK]D-Fender | correct |
| 23:36.44 | [TK]D-Fender | that is the DIALPLAN part of the call file |
| 23:37.08 | blehxor | right |
| 23:40.01 | *** join/#asterisk Draecos (~Draecos@124.150.57.219) |
| 23:40.59 | djgerm | gosh, so there's no way to see what my historical calls per second have been? |
| 23:43.38 | [TK]D-Fender | check your cdr |
| 23:43.48 | [TK]D-Fender | do math |
| 23:43.54 | [TK]D-Fender | GREAT VICTORY |
| 23:44.43 | djgerm | heh, yeah looks like that's how it's gonna have to be. though, even that's not gonna have the Calls Per Second... |
| 23:45.02 | djgerm | I'll have to grep out some unique line for session turn up out of the full log. |
| 23:45.34 | [TK]D-Fender | cdr use unixtime |
| 23:45.43 | [TK]D-Fender | not good enough? |
| 23:46.17 | Katty | HI KIDS |
| 23:46.31 | [TK]D-Fender | Katty: mew |
| 23:46.41 | djgerm | well I think that refers to when the call is completed, and since every call has a different duration, wouldn't that be not CPS? |
| 23:46.58 | Katty | [TK]D-Fender: how're you dear |
| 23:47.08 | [TK]D-Fender | No, it's start + duration |
| 23:47.20 | [TK]D-Fender | Katty: craptastic as my FB reads... |
| 23:47.26 | Katty | oh? |
| 23:47.29 | Katty | goes to look at fb |
| 23:47.46 | [TK]D-Fender | Katty: broken clavicle. kiss the summer goodbye |
| 23:47.55 | djgerm | so if i could subtract duration from that unixtime then the CDR would be useful... |
| 23:48.00 | Katty | [TK]D-Fender: WHY WOULD YOU DO THAT?! |
| 23:48.05 | Katty | [TK]D-Fender: Bad fender!!! |
| 23:48.13 | Katty | [TK]D-Fender: stop breaking things!!! |
| 23:48.43 | [TK]D-Fender | djgerm: no. It holds the start AND the duration |
| 23:49.00 | [TK]D-Fender | it is not the end time |
| 23:53.09 | djgerm | hmm. I don't understand. =D The unixtime I am seeing in my CDRs look like so: 1375833008.132653 |
| 23:54.27 | [TK]D-Fender | holds the START. and the DURATION |
| 23:58.05 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 23:58.12 | djgerm | The fleas are jumping of my head as it warms up with thought… I only have one unix time stamp in my cdr log =( |
| 23:59.07 | djgerm | and /etc/asterisk/cdr.conf is basically empty, so I am assuming default logging. |