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| 01:25.35 | volga629 | Hello Everyone, I am looking for confirmation if video should work through IAX2 trunk |
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| 02:08.25 | pensmit | core reload does that kill connections |
| 02:08.43 | pensmit | what can i do if i've changed manager.conf and just need it to reload that file |
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| 02:22.50 | tzanger | manager reload? |
| 02:30.24 | pensmit | thanks |
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| 05:13.58 | linocisco | softswitch vs asterisk ? |
| 05:19.50 | Penguin | yes |
| 05:21.24 | ChannelZ | burger vs hotdog? |
| 05:21.34 | Penguin | No contest. |
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| 06:12.13 | ChannelZ | Hmm. The You Tube is broken. |
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| 07:40.25 | pietro | hello, |
| 07:41.03 | pietro | can asterisk subscribe to external event dialog ? |
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| 07:59.14 | ChannelZ | What does that mean? |
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| 09:39.36 | kemmler | Is there any way to limit a call attempt to 10 seconds. Meaning if it takes longer than 10 seconds to initiate the call regardless of the reason, just scrap it? |
| 09:42.37 | asghar144 | kemmler: see wiki for dial options |
| 09:43.37 | asghar144 | Dial(${EXTEN},,10) |
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| 09:44.45 | kemmler | ahh ok thanks asghar144, i got confused about call timeout and that |
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| 10:34.32 | BorjaGVO | Hi, how does the "include" work when using it in the dialplan? http://pastebin.com/C9LYKXfi. Here I want to give priority to [ext-queues-custom]. Why is it that Asterisk is reaching extension 78 (the one that is not in the "include")? |
| 10:35.52 | WIMPy | Includes will only be searched if there's no match in the context itself. |
| 10:37.08 | BorjaGVO | WIMPy: I see |
| 10:37.36 | BorjaGVO | WIMPy: so, how could I achieve this behaviour without removing first appearence of extension 78? |
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| 10:41.43 | WIMPy | I'm not sure what you're trying to do, but having the same extension twice doesn't make much sense. |
| 10:45.52 | BorjaGVO | WIMPy: I work with FreePBX. It has autogenerated dialplan. I know I should ask there, but they won't answer. I'm trying to be as much general as possible |
| 10:45.57 | kaldemar | BorjaGVO: use more contexts. the matching order in a context is extensions first, then contexts in the order they are included. |
| 10:46.21 | BorjaGVO | I'll think how to deal with this. I have the answer I wanted related to Asterisk :-) |
| 10:47.44 | BorjaGVO | kaldemar: I think that wouldn't solve the issue, as it would match the existing extension before any contexts are taken into accountt |
| 10:49.08 | kaldemar | BorjaGVO: yes it would. |
| 10:49.46 | kaldemar | BorjaGVO: you would put the 78 extension in another context and include that in ext-queues after ext-queues-custom. |
| 10:51.15 | BorjaGVO | kaldemar: you mean the "first" 78 extension (putting it in another context and including it instead of leaving it inside exte-queues) |
| 10:51.17 | BorjaGVO | ? |
| 10:53.16 | kaldemar | s/78 extension in/78 extension in ext-queues into/ |
| 10:54.33 | BorjaGVO | kaldemar: right. I cannot do that, as ext-queues context is autogenerated |
| 10:56.41 | kaldemar | well, that is how you would do it in asterisk. #freepbx will tell you if your goal is doable with it. |
| 10:57.25 | BorjaGVO | kaldemar, WIMPy I think I found a solution. Before in the call-flow, there is a context where there is just includes. I'll start from there :P |
| 10:57.27 | BorjaGVO | Thanks! |
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| 14:49.52 | Katty | randomly throws parts around the room |
| 14:51.07 | Katty | aww. a tdm400 Rev F |
| 14:51.13 | Katty | it's so cute |
| 14:52.25 | Katty | a factory sealed pack of floppy disks! I"M RICH! |
| 15:01.03 | *** join/#asterisk Geek-Linux (~mubbashir@static-host210-2-165-210.link.net.pk) |
| 15:02.17 | Geek-Linux | Hello Every one. Can any one guide me . can i use separte sip trunks for voice and signalling |
| 15:02.17 | Geek-Linux | Or Simply is there any concept of signalling in Sip. ? |
| 15:02.38 | WIMPy | SIP is inly signalling. |
| 15:02.40 | WIMPy | only |
| 15:02.52 | WIMPy | Apart from that I don't get the question. |
| 15:03.16 | [TK]D-Fender | Geek-Linux: SIP *is* the signalling, not the voice. |
| 15:03.27 | Geek-Linux | WIMPy: if it is signalling then what about the voice ? |
| 15:03.36 | [TK]D-Fender | Geek-Linux: RTP <- |
| 15:03.46 | Geek-Linux | how is the media handled. |
| 15:04.21 | [TK]D-Fender | Geek-Linux: SIP (signalling) negotiates RTP (media) |
| 15:05.09 | Geek-Linux | [TK]D-fender: Means on the Sip trunk only signalling travels ? |
| 15:05.15 | *** join/#asterisk felipealmeida (~user@139.82.86.17) |
| 15:05.36 | WIMPy | ~siptrunk |
| 15:05.47 | [TK]D-Fender | Geek-Linux: Next, never use the term "SIP trunk". You are mixning some implied meaning of media in it as you go. |
| 15:05.57 | [TK]D-Fender | ~trunk |
| 15:05.57 | infobot | somebody said trunk was a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 15:06.27 | Katty | i'm going to start calling them snickerdoodle sessions. |
| 15:06.46 | WIMPy | A trunk is a single voice channel??? |
| 15:07.22 | Katty | it's what you put your clothes in when you're travelign 'cross the pond |
| 15:07.33 | [TK]D-Fender | WIMPy: It could be, but it's a question of encapsulation |
| 15:07.55 | [TK]D-Fender | WIMPy: Much like I had with my frame-relay tie-line a decade ago |
| 15:08.07 | WIMPy | I would expect a trunk to be a number of lines. |
| 15:08.21 | Geek-Linux | infoot: if it is a stream of UDP packets. Then i can say it is taking the Voice packets also |
| 15:08.44 | WIMPy | Geek-Linux: Yes, but that's not SIP. |
| 15:09.11 | *** join/#asterisk rawrio (~Adium@12.28.240.162) |
| 15:09.42 | Geek-Linux | WIMPy: if it is not SIP. then you mean it is RTP who is doing the JOB. |
| 15:09.53 | WIMPy | yes |
| 15:09.57 | Katty | file: scrumdiddlyumptious! |
| 15:10.04 | Katty | file: pls ship cookie. kthx. |
| 15:10.07 | file | gives Katty a cookie |
| 15:10.11 | Katty | yay |
| 15:10.12 | Katty | noms |
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| 15:10.32 | rawrio | we are trying to upgrade our installation of asterisk from 11.4 to 11.5.1 and are having problems. When we look at the debug we are seeing "SIP/2.0 403 Forbidden". Any suggestions on what we might be missing/need to tweak? |
| 15:11.03 | rawrio | everything else the same on 11.4 works fine, but when we try to dial any extension once on 11.5.1 it gives a busy signal |
| 15:11.22 | Geek-Linux | WIMPy: Can we configure SIP or RTP to recieve signals on one and send voice traffic on another way. means two different paths |
| 15:11.39 | Katty | rawrio: insecure=invite comes to mind |
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| 15:11.54 | Katty | rawrio: a change of public IPs comes to mind |
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| 15:12.09 | Katty | rawrio: chocolate also comes to mind... |
| 15:12.16 | Katty | rawrio: but i think about chocolate a lot |
| 15:12.48 | WIMPy | Geek-Linux: That's a quite common thing to happen with ITSPs. |
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| 15:13.30 | Geek-Linux | WIMPy: but i have never come accross this. can you please help me ? |
| 15:13.40 | Katty | grins |
| 15:13.52 | Katty | WIMPy: i think you need a coffee break |
| 15:13.56 | WIMPy | With what EXACTELY? |
| 15:14.07 | Katty | WIMPy: with coffee! |
| 15:14.19 | Katty | WIMPy: surely there is coffee somewhere. with creamer. that's calling you |
| 15:14.20 | WIMPy | Drugs are bad! |
| 15:14.32 | Katty | it comes in decaf >.< |
| 15:15.01 | [TK]D-Fender | [11:11]Geek-LinuxWIMPy: Can we configure SIP or RTP to recieve signals on one and send voice traffic on another way. means two different paths <- NO. |
| 15:15.13 | WIMPy | Wat's the point in drinking stuff that tastes horrible if it doesn't even have an effect? |
| 15:15.14 | [TK]D-Fender | Geek-Linux: * does not negotiate a 3rd party RTp server. |
| 15:15.44 | WIMPy | But that's exactely what you do when configuring externhost, isn't it? |
| 15:15.47 | [TK]D-Fender | Geek-Linux: * is a B2BUA. Not a set of separate signalling and media servers |
| 15:16.09 | Katty | WIMPy: what about iced coffe? |
| 15:16.41 | WIMPy | That's ok if there's enough milk. |
| 15:16.54 | Katty | nods |
| 15:16.57 | Geek-Linux | [TK]D-Fender: Means there is only way to create a single trunk to recieve calls ? |
| 15:16.59 | Katty | i prefer mine iced too, actually |
| 15:17.17 | Katty | also, i miss danny. |
| 15:17.18 | Katty | drmessano: ping |
| 15:17.25 | Katty | drmessano: where are you?! i've nto seen you all week |
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| 15:17.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:17.29 | WIMPy | Stok that "trunk" thing. |
| 15:17.34 | Katty | sruffell: ohai |
| 15:17.39 | WIMPy | Stop... |
| 15:17.40 | sruffell | bows graciously |
| 15:17.53 | [TK]D-Fender | [11:16]Geek-Linux[TK]D-Fender: Means there is only way to create a single trunk to recieve calls ?What is this i"single trunk" you're speaking of? |
| 15:18.04 | [TK]D-Fender | Geek-Linux: Your terminology is dangerously vague |
| 15:18.19 | [TK]D-Fender | Geek-Linux: and nver use that term "SSIP trunk" again. |
| 15:18.22 | Katty | that's because Geek-Linux is confused, and has yet to wrap his brain around it. |
| 15:18.56 | [TK]D-Fender | Geek-Linux: Asterisk has PEER ENTRIES that define auth. * is a B2BUA, not a SIP SERVER, or proxy, or media server. It is effectively a dumb end-point no different than any SIP phone |
| 15:19.54 | Geek-Linux | [TK]D-Fender: Actually a scenario. besides TDM lines i am going setup my machines with SIP trunks with telco to recieve calls. and increase my call capacity. |
| 15:20.18 | Katty | Geek-Linux: lots of people do that |
| 15:20.42 | WIMPy | Geek-Linux: Maybe we should ignore you for 10 minutes, each time you say "sip trunk". |
| 15:20.49 | Katty | oh be nice guys. |
| 15:21.08 | Katty | sales folk use "sip trunk" all the time |
| 15:21.16 | Katty | there is a reason we have these cutesy little phrases that don't actually mean anything |
| 15:21.21 | [TK]D-Fender | Geek-Linux: Again, stop using the word "trunk". Permanently. And * can speak SIP and use TISPs just fine |
| 15:21.22 | WIMPy | We are talking about sip. There's nothing nice in there. |
| 15:21.26 | [TK]D-Fender | ITSPs* |
| 15:21.30 | Katty | well fine. you two be cranky. |
| 15:21.35 | Katty | Geek-Linux: please continue. |
| 15:21.43 | Geek-Linux | [TK]D-Fender: but at telco side requested to use separate IP Addresses for signalling and media. and i am confused in that. as for as i know there is no such thing in asterisk. |
| 15:21.43 | rawrio | katty: we didn't change ips and we tried the insecure setting change, still getting forbidden errors |
| 15:22.01 | rawrio | katty: we are running FreePBX along with it |
| 15:22.09 | Katty | rawrio: meep |
| 15:22.15 | Katty | rawrio: go visit #freepbx then |
| 15:22.28 | tm1000 | ~freepbx |
| 15:22.28 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:22.37 | [TK]D-Fender | [11:21]Geek-Linux[TK]D-Fender: but at telco side requested to use separate IP Addresses for signalling and media. and i am confused in that. as for as i know there is no such thing in asterisk. <- perhaps you misunderstood what they were asking for. |
| 15:22.47 | WIMPy | Geek-Linux: It's pretty normal that they would use different IPS for SIP and RTP. Requesting that from you seems very unusual. |
| 15:22.59 | [TK]D-Fender | Geek-Linux: THEY might have separate servers for this and their SIP negotiation would TELL you where to go for the media |
| 15:23.01 | Geek-Linux | [TK]D-Fender: than what they means ? |
| 15:23.19 | [TK]D-Fender | Geek-Linux: But I have *NEVER* heard of any service that forces YOU to have separate signalling and mdeia IP's |
| 15:24.26 | Geek-Linux | WIMPy: why unusual ? :( |
| 15:24.54 | WIMPy | I haven't heard of such requests before, either. |
| 15:25.15 | [TK]D-Fender | Geek-Linux: Why do *I* need multiple server IP's just to do voip? |
| 15:25.21 | Katty | oh do i want to tinker with blacklists or cdr stuffs today |
| 15:27.58 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:28.47 | Geek-Linux | <[TK]D-Fender> i actually dont know but they have asked just for the possiblity. |
| 15:29.04 | [TK]D-Fender | Geek-Linux: Show us this request. |
| 15:29.55 | Geek-Linux | [TK]D-Fender: it was just a verbal discussion with the telco, not in written. |
| 15:30.11 | navaismo | we need a wall of shame for providers |
| 15:30.21 | [TK]D-Fender | Geek-Linux: and the likelyhood of something misunderstood given your lack of background knowledge of SIP is high |
| 15:33.41 | Geek-Linux | [TK]D-Fender: May be. but one other question why you are using terminology of TISPs instead of trunk. |
| 15:33.55 | [TK]D-Fender | ~itsp |
| 15:33.56 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 15:34.22 | navaismo | ~itsplist-mx |
| 15:34.28 | navaismo | ¬¬ |
| 15:34.32 | [TK]D-Fender | Geek-Linux: ITSP describes their role as a terminator. "Trunk" has a more specific meaning |
| 15:34.33 | [TK]D-Fender | ~trunk |
| 15:34.33 | infobot | well, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 15:34.41 | [TK]D-Fender | it's about ENCAPSULATION]\ |
| 15:35.06 | [TK]D-Fender | IAX can bond several voice channels into a single stream. |
| 15:35.14 | [TK]D-Fender | SIP is stand-alone |
| 15:35.27 | Geek-Linux | infobot: if i am in a room of asterisk then i am talking about asterisk trunks not the others :) |
| 15:35.28 | WIMPy | And even stateless. |
| 15:35.45 | [TK]D-Fender | Geek-Linux: Stop talking to the bot BTW, it's too early here for that sort of humour |
| 15:35.58 | Katty | no it's not. |
| 15:36.07 | Katty | infobot: tell fender it's not too early |
| 15:36.20 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:36.20 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:36.20 | Katty | infobot: tell [TK]D-Fender it's not too early |
| 15:36.25 | newtonr | Arguing against the term SIP trunk is sort of pointless. In 4 years in tech support, every single customer calling in referred to their SIP service from an ITSP as a SIP trunk. It has already moved into common usage. |
| 15:36.58 | Geek-Linux | [TK]D-Fender: hmm ok. |
| 15:37.32 | [TK]D-Fender | newtonr: And then every idot starts calling their sip.conf entries "trunks". And then the word trunk becomes the next "smurf" or "marklar". I don't empower ignorance. |
| 15:38.00 | [TK]D-Fender | HOW I CAN TRUNK MY TRUNK. i ALREADY DID MY SIP?!?! |
| 15:38.34 | Katty | thinks [TK]D-Fender should go have a coffee and come back in 10 minutes. |
| 15:38.35 | [TK]D-Fender | Get your terminlogy right or everyone will be guessing what you're trying to communicate |
| 15:40.32 | Geek-Linux | [TK]D-Fender: Still have questions about you attitude, just ask some thing and you are getting angry why ? |
| 15:40.50 | Katty | Geek-Linux: let's just leave everyone's attitude out of the question |
| 15:41.08 | Katty | Geek-Linux: i'm sure you didn't come here to ask about fender's attitude. |
| 15:41.39 | sruffell | smirks |
| 15:41.51 | Katty | sruffell: you wipe that smirk off your face. |
| 15:41.59 | Katty | sruffell: or i'll have file stuff a cookie in it! |
| 15:42.01 | sruffell | looks out the window |
| 15:42.21 | [TK]D-Fender | Geek-Linux: You asked why one shouldn't use the term "SIP trunk". This was why |
| 15:42.56 | [TK]D-Fender | Geek-Linux: I am clarifying proper terminology for you that will save you a lot of grief down the road configuring all of this |
| 15:43.00 | Geek-Linux | [TK]D-Fender: ok |
| 15:43.30 | [TK]D-Fender | Geek-Linux: Vague terms will lead to you getting inappropriate advice for what you actually need to accomplish |
| 15:43.51 | Geek-Linux | [TK]D-Fender: thanks for the advise |
| 15:44.14 | [TK]D-Fender | Geek-Linux: Them having separate media servers, vs you, etc. |
| 15:44.24 | [TK]D-Fender | Geek-Linux: Details make or break your deployment |
| 15:44.27 | Geek-Linux | [TK]D-Fender: thats why i am here. to ask somethink from you people. |
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| 15:56.08 | Katty | chirp. chirp. chirp. chirp. |
| 15:57.22 | navaismo | chips chips chips |
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| 16:13.41 | *** join/#asterisk joesuffceren (~chatzilla@ip72-223-76-226.ph.ph.cox.net) |
| 16:13.57 | joesuffceren | is there a significant qualitative difference between hardware echo cancellation on a digium T1 card versus software echo cancellation (using a digium T1 card without the echo cancellation module)? I know that offloading the echo cancellation will save CPU, but if CPU isn't an issue (low call volume), will the quality of the echo cancellation being performed be much different? |
| 16:15.02 | WIMPy | Well, when running it on the host CPU you have different EC algorithms to choose from. |
| 16:16.29 | joesuffceren | I've used some of them before when using a TDM800 card. I just wondered if the general consensus was "software echo cancellation is a joke; use as a last resort" or if the community considered it a legitimate option. The echo cancellation module is around 40% of the cost of the card, which I'd rather not spend if I don't have to |
| 16:16.58 | joesuffceren | that said, $500 is a relatively small price to pay if there's a big qualitative difference. |
| 16:17.16 | joesuffceren | WIMPy: have you had good success with software echo cancellation? |
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| 16:18.41 | WIMPy | I haven't tried it in situations whe echo would be expected. |
| 16:18.57 | WIMPy | But I'd ecpect it to be just as good. |
| 16:19.23 | WIMPy | But it does eat CPU time and it cannot make use of multiple cores. |
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| 16:22.15 | joesuffceren | gotcha. thanks for the tip about SMP. I didn't realize that. I'll probably just bite the bullet on the modules to be safe and future proof. Thanks for your help! |
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| 16:49.28 | niplo | What are the supported charsets for cdr_mysql.conf |
| 16:50.49 | niplo | Should i use utf8 || UTF8 || utf8_general_ci? |
| 16:50.58 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bktwkqtaguwjwcpu) |
| 16:53.13 | pensmit | Hey Guys...how do I get exten *97 to go to a users voicemail and ask for their password immediately without keying the extension |
| 16:53.58 | pensmit | same => n,VoiceMailMain(${CHANNEL}@default) |
| 16:54.46 | navaismo | ${CALLERID(num)} |
| 16:56.23 | pensmit | ok thanks...I thought that was deprecated |
| 16:56.48 | pensmit | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 16:57.03 | pensmit | ${CALLERID(num)}: The current Caller ID number - ${CALLERIDNUM} was used in versions of Asterisk prior to 1.2.0, it was DEPRECATED in 1.2.0 and removed in 1.4. |
| 16:57.27 | pensmit | oh derp |
| 16:57.27 | pensmit | i see |
| 16:57.35 | pensmit | use to be one command |
| 16:57.44 | pensmit | er variable |
| 16:58.00 | pensmit | thanks a lot |
| 16:58.08 | [TK]D-Fender | No, used to be multiple variables, now it's a function with multiple sub-values |
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| 17:00.28 | newtonr | niplo, I don't think there are any possible values for the charset option other than "koi8r". https://issues.asterisk.org/jira/browse/ASTERISK-12958 is the issue where it was added. |
| 17:01.04 | newtonr | "This option allows you to work with non latin characters in DB when DB's character set differs from asterisk's locale." |
| 17:03.08 | niplo | I ve already read that but koi8r is for Russian-Bulgarian characters |
| 17:03.29 | niplo | Am truing to display Greek one |
| 17:04.11 | newtonr | niplo, I've never messed with character set differences. Have you tried it out to see if there is any problems? |
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| 17:05.24 | niplo | Am goin to do it right now |
| 17:06.30 | niplo | No hope |
| 17:06.54 | niplo | I'll open a thread in the forum. |
| 17:06.57 | niplo | Thanks anyway |
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| 17:26.09 | pabelanger | ifconfig |
| 17:26.12 | pabelanger | grr |
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| 18:12.20 | *** join/#asterisk dorphalsig (b532ffa2@gateway/web/cgi-irc/kiwiirc.com/ip.181.50.255.162) |
| 18:14.10 | dorphalsig | Hi. I want to do a stress test on my asterisk system before I move it up to production. Essentially I would like to run a simulation of the behaviour of the queues. I know I can originate calls with SIPp, but how can I answer them? |
| 18:15.18 | navaismo | via dialplan--Answer |
| 18:15.31 | leifmadsen | right |
| 18:16.02 | leifmadsen | dorphalsig: https://github.com/mojolingo/sippy_cup |
| 18:18.52 | navaismo | +10000 ^ |
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| 18:23.54 | roderickm | dorphalsig: If Asterisk answering for itself is too cold, and a sipp/sippy_cup test scenario is too hot, pjsua with auto-answer enabled may be just right. |
| 18:24.16 | *** join/#asterisk serafie (~erin@nat/digium/x-aetribnztlavrppn) |
| 18:39.04 | dorphalsig | roderickm: I did see pjsua, but I didnt quite get how it works. Do I send it raw SIP commands? |
| 18:40.53 | roderickm | dorphalsig: No. Here's a cheatsheet: http://tomeko.net/other/sipp/sipp_cheatsheet.php?lang=pl |
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| 18:42.19 | dorphalsig | But thats for SIPp, or is it for pjsua |
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| 18:44.23 | dorphalsig | roderickm: Cool!! Just saw how it should work. And it actually streams audio out |
| 18:45.03 | roderickm | Yes, not as configurable as sipp/sippy_cup, but rather accessible. |
| 18:46.36 | dorphalsig | roderickm:ohh so you're saying to do both ends with pjsua? make it call and answer/ |
| 18:46.43 | dorphalsig | (of couse different instances) |
| 18:46.48 | roderickm | either way. |
| 18:47.18 | roderickm | I used sipp to generate and pjsua to answer, but either can do either/both. |
| 18:49.48 | dorphalsig | roderickm: How would I make pjsua (or sipp) dial some extension? |
| 18:50.40 | dorphalsig | I mean. I guess the basic test would be: call extension 900, wait for answer , hang up |
| 18:52.25 | roderickm | Yes. There are examples on that cheatsheet with sample scenarios and everything. |
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| 19:40.51 | *** join/#asterisk vittorio88 (~vitto@net-2-34-114-99.cust.dsl.vodafone.it) |
| 19:41.31 | vittorio88 | hello everybody! |
| 19:44.51 | vittorio88 | I am trying to set ip sip/tls on asterisk 11.5 with blink on win7. |
| 19:44.51 | vittorio88 | I successfully authenticate and can dial, but get NO AUDIO. Same config with no tls works just fine. |
| 19:44.51 | vittorio88 | Asterisk prints: |
| 19:44.51 | vittorio88 | <PROTECTED> |
| 19:44.52 | vittorio88 | [Sep 19 14:40:20] WARNING[3965]: tcptls.c:261 handle_tcptls_connection: FILE * open failed! |
| 19:44.52 | vittorio88 | [Sep 19 14:40:52] NOTICE[3803]: chan_sip.c:27543 handle_request_subscribe: Failed to authenticate device "vitto" <sip:vitto@sip.promaq.mx>;tag=gyrBSLiINm5odzuoicYskTta2IE-edB3 for SUBSCRIBE |
| 19:44.52 | vittorio88 | Any ideas as to the cause or debugging steps? |
| 19:53.37 | navaismo | when i saw this message FILE * open failed! i added the ca.crt file to windows by double click in it |
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| 20:11.35 | vittorio88 | are you saying you saw the problem because you added ca.crt to windows, OR you solved the problem by adding ca.crt to windows? |
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| 20:17.40 | navaismo | solved by adding, at least the message FILE *.. dissapear |
| 20:18.59 | vittorio88 | i tried, it persists. |
| 20:20.36 | navaismo | too bad. Are you sure blink is the only client open in the pc |
| 20:20.59 | vittorio88 | in task bar, yes. |
| 20:21.03 | vittorio88 | lemme check services |
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| 20:22.33 | vittorio88 | yeah, it's the only one. |
| 20:23.07 | navaismo | are you following the secure calling tutorial from the asterisk wiki? |
| 20:23.13 | vittorio88 | precisely. |
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| 20:23.40 | navaismo | do you see errors on sip reload? |
| 20:24.58 | vittorio88 | not on console |
| 20:25.32 | vittorio88 | lemme grep logs |
| 20:28.16 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:28.40 | boom^time | So I'm trying to follow AMI events of a call origination by calling a local extension to store the channel name to follow the events (as suggested to me by nextime and WIMPy yesterday) and when I do it this way it creates three channels now, seemingly bridges them a couple of different times, making it even harder to track now. Does anyone have any suggestions for me? |
| 20:29.28 | vittorio88 | Here's the grep. |
| 20:29.29 | vittorio88 | grep -i 'error' messages |
| 20:29.29 | vittorio88 | [Sep 19 15:28:51] ERROR[4181] pbx.c: Function JABBER_STATUS already registered. |
| 20:29.29 | vittorio88 | [Sep 19 15:28:51] ERROR[4181] pbx.c: Function JABBER_RECEIVE already registered. |
| 20:29.29 | vittorio88 | [Sep 19 15:28:51] ERROR[4181] message.c: Message technology already registered for 'xmpp' |
| 20:29.30 | vittorio88 | [Sep 19 15:28:51] ERROR[4181] chan_motif.c: Connection 'local-jabber-account' configured on endpoint 'jingle-endpoint' could not be found |
| 20:29.30 | vittorio88 | [Sep 19 15:28:51] ERROR[4181] config_options.c: Error parsing connection=local-jabber-account at line 81 of |
| 20:29.30 | vittorio88 | [Sep 19 15:28:51] ERROR[4181] config_options.c: In motif.conf: Processing options for jingle-endpoint failed |
| 20:29.31 | vittorio88 | [Sep 19 15:28:51] ERROR[4181] chan_motif.c: Unable to read config file motif.conf. Not loading module. |
| 20:29.55 | *** part/#asterisk sgriepentrog (~sgriepent@nat/digium/x-mzreabwufonzugza) |
| 20:30.45 | navaismo | vittorio88: use pastebin next time |
| 20:30.52 | navaismo | ~pb |
| 20:30.52 | 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. |
| 20:30.58 | vittorio88 | ok. sorry about that |
| 20:31.17 | navaismo | and those errors are for different module not sip |
| 20:31.35 | vittorio88 | yeah, i was trying to show that. |
| 20:32.00 | navaismo | boom^time: hire a guy to do that for you |
| 20:32.28 | boom^time | navaismo, I'd rather write my own AMI wrapper. |
| 20:32.36 | boom^time | It's just the API was poorly designed. |
| 20:32.48 | boom^time | Unless of course I'm missing something obvious. |
| 20:34.42 | navaismo | seems like you are trying to write a dialer, many dialers are in the web, you can take a look on those ami dialers |
| 20:37.00 | boom^time | Most dialers just connect and place calls. That's easy enough to do. I want to follow each channel and know it's state. Which would be very easy to do if the channel creation event would point to the origination in anyway. I need to look at the source and hope it was laziness. |
| 20:37.43 | navaismo | yes that kind of dialers are very common |
| 20:37.54 | navaismo | and based on the logs many guys already tell you hints to do that |
| 20:38.08 | boom^time | Yes, and I exhausted those hints. |
| 20:38.15 | boom^time | Or have follow up questions. |
| 20:38.35 | navaismo | that why i recommend to hire someone |
| 20:38.38 | boom^time | I'd rather learn how to do something than hire someone. |
| 20:38.58 | navaismo | google |
| 20:39.35 | boom^time | Thanks |
| 20:39.53 | navaismo | or try to use the asterisk 12 |
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| 20:50.30 | WIMPy | boom^time: Yes, that's because of the local channel. You need to track those events as well. |
| 20:50.49 | WIMPy | That's unfortunatly just the way it is. |
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| 20:51.35 | boom^time | WIMPy, thank you. I'm digging into manager.c out of curiosity but that's definitely my next step. |
| 20:52.58 | WIMPy | That has nothing to do with the manager. That's just 1. the two legs of the call being bridged together (twice because of the local channel in between) and 2. the local channel being optimized away. |
| 20:53.22 | WIMPy | You can eliminate 2. when using /n on the local channel. |
| 20:53.42 | boom^time | Where do you use /n? |
| 20:53.58 | WIMPy | On the dialsting. |
| 20:54.10 | WIMPy | local/extension@context/options |
| 20:54.21 | boom^time | btw, I'm looking at the manager to see if I can get the Newchannel event to display an ActionID |
| 20:54.28 | boom^time | so I won't have to take the local route |
| 20:55.42 | WIMPy | I'm sure a lot of people would find that extremely usefull. |
| 20:56.39 | boom^time | Right, which is probably why it won't be a bandaid style fix. But I might as well check before writing all the code to trace all of this |
| 20:56.58 | boom^time | If I have any luck I'll let you know. |
| 20:57.34 | WIMPy | I haven't looked at that place, but I'd imagine it shouldn't be too hard to do. |
| 20:57.59 | boom^time | My thoughts are: if it was easy to do, then why in the hell didn't anyone do it? |
| 20:58.06 | boom^time | but logic doesn't always match reality. |
| 20:58.22 | boom^time | Maybe it's just commented out somewhere :) |
| 20:58.27 | WIMPy | There are just too many things to do. |
| 20:58.32 | WIMPy | Unlikely. |
| 20:58.50 | boom^time | Yeah it was just a bad joke. |
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| 22:06.26 | bobbyz | If I wanted to prioritize sip voice quality above bandwidth, would a codec order of speex, ilbc, ulaw, alaw, g729 be appropriate? |
| 22:07.03 | WIMPy | no |
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| 22:07.22 | bobbyz | what would you recommend? |
| 22:07.34 | WIMPy | G.722>G.711>most of the common stuff |
| 22:07.46 | jrose_atDigium | second G722 |
| 22:08.07 | WIMPy | Siren and opus would also be above G.711. |
| 22:08.39 | bobbyz | interesting, so speex, ilbc are behind 722, 711, siren, and opus |
| 22:11.10 | bobbyz | so more appropriate might be: g722, ulaw, alaw, g729? I don't even think my DID provider supports speex, siren, or opus anyway |
| 22:12.17 | WIMPy | Might make sense to find out first. |
| 22:13.23 | bobbyz | will do, just trying to get an idea for quality. I've been trying to google it, but the information I have found has been kind of mixed |
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| 22:21.31 | [TK]D-Fender | bobbyz: the PSTN is almost invariably G>711 anyway so any conversion up adds precisely NO value, and then when that gets transocded back you actually LOSE quality |
| 22:21.47 | bobbyz | good point |
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| 22:24.55 | mic__ | hello, just thinking loud |
| 22:25.00 | mic__ | channel names can repeat |
| 22:25.15 | mic__ | they are not 100% unique. Right? |
| 22:27.21 | WIMPy | yes |
| 22:28.50 | mic__ | but it does hold true, that there will never be two channels existing at a given moment with the same name? |
| 22:30.24 | WIMPy | yes again |
| 22:31.25 | mic__ | I was just grepping a log today |
| 22:31.35 | mic__ | and the channel names were pretty close to each other time-wise. |
| 22:34.41 | pensmit | where does comedian mail store passwords once the user makes a change |
| 22:35.00 | WIMPy | voicemail.conf |
| 22:36.47 | mic__ | WIMPy: so in theory sending a Bridge command to AMI with two channels named - taking into account statements above - should always connect the the right parties together |
| 22:37.52 | WIMPy | yes |
| 22:38.40 | WIMPy | Unless you took so long to do ith that one of the channels might have gone and onother one has been created with the same name. |
| 22:39.13 | pensmit | voicemail.conf seems to contain the initial password but if you change it in comedian male, the old one is still in voicemail.conf. |
| 22:39.32 | pensmit | So I'm wondering where comedian mail stores |
| 22:39.36 | pensmit | the changed one |
| 22:39.51 | Penguin | You don't have the right permissions on the file. |
| 22:40.09 | Penguin | Asterisk will change the password in voicemail.conf when you do it over the phone. |
| 22:40.36 | mic__ | WIMPy: yes, of course. But what I think happens -> two active calls, both do attended transfers and then they Bridge the two parties together in each situation. What they get is that sometimes they connect together these people in different fashion, like crossed |
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| 22:41.33 | WIMPy | They will have unique channel names at that time. |
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| 22:58.53 | pensmit | yep permissions |
| 22:58.57 | pensmit | i love you guys |
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| 23:06.04 | monsterco | Hi everyone - I would like to part way with my OpenVPN which is now being used on my clients sites and make a connection to my Asterisk servers. I would like to allow client traffic based on their DynDNS (or similar service) url so that I can limit access to known networks only. Would this give me a stable solution? and also a secure one to some degree? |
| 23:07.13 | Penguin | You'd have to refresh your firewall settings before their dynamic IP address would get updated in your firewall. |
| 23:07.42 | phix | Limiting by hostname / IP address is usually not very secure, addresses can be spoofed |
| 23:08.05 | monsterco | I thought there is no way to spoof IP source address |
| 23:08.31 | phix | Of course there is a way |
| 23:08.41 | monsterco | or at least the reply back to the source IP would yield nothing as my packets will go to proper source even if the IP is spoofed |
| 23:08.50 | phix | just send packets with the source address changed |
| 23:08.53 | Penguin | More importantly, if you are using the dynamic DNS host name in your firewall settings, it will lock in the current IP address. It will not change dynamically, so you have to refresh using the hostname many times. |
| 23:09.23 | monsterco | Penguin - my firewall can allow or block traffic based on url and IP - I am using pfSense |
| 23:09.51 | Penguin | If you set a rule by host name, it translates that to the IP address at the time you set it. |
| 23:10.02 | phix | monsterco: And does it refresh DNS? or does it resolve the hostname to an IP address straight away and not refresh later? |
| 23:10.17 | Penguin | It will not change dynamically with the IP address of the host. |
| 23:10.18 | monsterco | I beleive ir resolves the hostname to IP each and every time |
| 23:10.30 | phix | monsterco: What if the hostname resolves to more than one IP address? DOes it add all of them or the first one? |
| 23:10.41 | monsterco | phix - how would that be possible? |
| 23:10.44 | phix | monsterco: Sounds like a slow firewall :) |
| 23:10.54 | phix | monsterco: How is what possible? |
| 23:10.55 | Penguin | Never seen multiple A records before? |
| 23:11.02 | monsterco | are you guys familiar with pfSense? |
| 23:11.06 | Penguin | Yes. |
| 23:11.13 | phix | nope, I am familar with firewalls though |
| 23:11.58 | phix | ah FreeBSD, well I know FreeBSDs firewall implementation looks up the IP address once when you add the rule |
| 23:12.07 | monsterco | Penguin - you think pfSense resolves DNS to IP each time or does it use the cache? |
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| 23:12.39 | phix | If it looked it up on every packet you would have a very slow transfer speed |
| 23:12.47 | Penguin | I think pf behaves just like iptables in that sense. If you set a rule using a NAME, it translates it to an IP address at the time you set it. |
| 23:12.53 | phix | DNS look ups are expensive |
| 23:13.29 | phix | (in terms of round trip / time to complete) |
| 23:13.38 | monsterco | Penguin - I will ask this in pfSense channel |
| 23:13.54 | Penguin | If host.domain.com has IP address 1.2.3.4 when you set the rule, and then it changes later to 2.3.4.5, the firewall will continue to use 1.2.3.4 until something forces it to resolve again. |
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| 23:14.07 | monsterco | so, what if I install a nice littel box which will ping me every 5 seconds and then I create a script that grabs that IP and updates my firewal? |
| 23:14.13 | Penguin | There's no way it is going to resolve every time. |
| 23:14.48 | Penguin | That's a lot of additional traffic. |
| 23:15.01 | monsterco | I don't mind - or maybe once every 60 seconds |
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| 23:15.09 | monsterco | an optimized ping |
| 23:15.56 | phix | monsterco: ok so then I ping you, your firewall adds my address and then I exploit your server? |
| 23:16.02 | monsterco | or just "wget http://telecom-server/client-id=123" where index is empty and I get to see the source IP in |
| 23:16.16 | monsterco | phix - you would have to ping with a password ID |
| 23:16.26 | Penguin | ICMP doesn't support that. |
| 23:16.27 | monsterco | or else you get added to firewall block list |
| 23:16.33 | phix | pings are unencrypted |
| 23:16.42 | phix | Penguin: you can ping with random data as the packet though |
| 23:16.43 | monsterco | won't use ICMP - curl it maybe |
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| 23:17.15 | phix | Penguin: I know that because I used to do +++athd1 back in the modem days :P |
| 23:17.25 | phix | or was it ath0 |
| 23:17.30 | phix | *shrugs* I dont remember anymore |
| 23:17.32 | Penguin | ath0 |
| 23:17.38 | Penguin | Ping of death. |
| 23:17.58 | phix | well, just hang up modems that don't have a high enough guard timing set :) |
| 23:18.16 | phix | was useful when I was over hearing people chat nonsense, just hang up their modem :) |
| 23:18.40 | phix | Ah that brings back memories, I was 14 I think |
| 23:19.02 | Penguin | 14? I did that stuff last week! |
| 23:19.18 | Penguin | Not really. |
| 23:21.06 | phix | haha |
| 23:21.57 | phix | so any way monsterco lets think of something more practicle ;) |
| 23:22.04 | phix | What are your objectives? |
| 23:23.22 | monsterco | just a secure network - I want to ditch openvpn (has been working greatly but more overhead and an extra box) |
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| 23:34.21 | phix | monsterco: secure network, so vpn? or srtp? what services are you trying to protect (besides asterisk) |
| 23:35.41 | phix | also are you running OpenVPN in tcp or udp mode? |
| 23:35.43 | Penguin | He wants to discontinue use of the VPN. |
| 23:36.15 | phix | but still retain some of its capability? |
| 23:37.13 | Penguin | It sounded like he wanted to try to employ some other mechanisms to regain the security he'll be giving up when he dumps the vpn. |
| 23:37.36 | Penguin | firewall ACLs |
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| 23:55.20 | phix | I supose he could use ipsec if he wants to keep security but use a lower lvl, although the overhead wouldn't be too much different |
| 23:57.24 | monsterco | phix - just Asterisk - and all SIP phones with clients - I just can't deal with SIPVICIOUS type of things - rather keep things closed at the bottle neck |
| 23:57.24 | phix | monsterco: ok so why not use OpenVPN? That gives you a secure network, what other requirements do you have? |
| 23:57.39 | monsterco | phix - openvpn in UDP mode - what difference does it make? |
| 23:57.45 | phix | Speed |
| 23:57.50 | phix | UDP is a hella alot quicker |
| 23:58.19 | phix | running tcp over tcp is slower than tcp over udp |
| 23:58.37 | monsterco | can't use IPSEC - I want to dump pfSense embedded boxes which create the OpenVPN tunnels now - that's extra cost and management overhead - let's stick to Aastra phone and what they are capable of |
| 23:58.38 | Penguin | But SIP and RTP are UDP anyway. |
| 23:59.05 | monsterco | yeah I am runningin UDP - I don't have issues with current setup - just want to keep same level of security if possible |
| 23:59.24 | phix | Penguin: yes but if OpenVPN is set in TCP mode then you are using UDP over TCP, which negates the advantages of UDP (speed) |
| 23:59.45 | Penguin | You said TCP over TCP. I was saying SIP/RTP are UDP. |