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| 01:48.08 | Cyford33 | i should be able to use the AND like this correct? ExecIf($[${strcredit} < 3 AND {strtariff} != 13 ]?agi(googletts.agi,"You should refill your Card.",en)) |
| 02:04.08 | [TK]D-Fender | not with that syntax |
| 02:04.11 | [TK]D-Fender | read the WIKI |
| 02:04.15 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Operators |
| 02:04.25 | [TK]D-Fender | ALWAYS go read the wiki for this stuff. |
| 02:08.43 | MaliutaLap | I don't think any language would parse that the way intended |
| 02:09.11 | MaliutaLap | general mathematic order of precedence says no |
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| 02:49.44 | ctpdump | Hi. I was wondering if the user/password are being sent in cleartext if my provider isn't using TLS? (eg: normal 5060/udp)? or are they encrypted in anyway |
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| 04:21.05 | ChannelZ | ctpdump, they are hashedencrypted |
| 04:21.22 | ChannelZ | oh look I've invented a new word. |
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| 04:43.13 | drmessano | ChannelZ, better than hashedpasta |
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| 06:24.10 | seik0 | hi again! Is it possible to run two asterisk on one machine? What should I take into consideration (google doesn't give much at first glance) ? |
| 06:24.30 | seik0 | I understand that network configurations may mix |
| 06:25.52 | seik0 | But I want two instances for that: one instance is minimal, processing (as proxy mainly) 2*e1 and another instance is fully-capable of any other functions |
| 06:26.53 | seik0 | And that I need because main asterisk is hardly connected with DB and built-in res_odbc is not very good on high loads |
| 06:27.02 | seik0 | even on not very high loads |
| 06:28.23 | seik0 | especially on cases when DB connection is lost. In that case asterisk hangs (maybe that is ODBC issue and, moreover, ODBC oracle driver issue, but for a moment we cannot afford another DB-connection solutions) |
| 06:33.11 | seik0 | As I see, this DB issue happens not only with oracle |
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| 06:44.25 | Chainsaw | seik0: Speak to jkroon, he runs multiple Asterisk instances on a single machine. |
| 06:46.58 | seik0 | jkroon, hi, I want to speak with you |
| 06:47.00 | seik0 | =) |
| 06:47.08 | seik0 | Chainsaw, thanks |
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| 07:06.27 | ldc | seik0: you might want to query him since he's not in channel |
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| 07:11.43 | seik0 | ldc, looks like I found his blog |
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| 08:10.57 | mastahh | What hardware specifications do you guys set for your Asterisk servers when deploying them on customer sites? I am currently considering deploying with 2 processors, 4GB of RAM and 80GB of HDD space |
| 08:11.33 | mastahh | Bearing in mind I will probably be deploying seven or eight servers in a single cluster totally up to 24-28GB of RAM and up to 1TB of HDD with all the servers. |
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| 08:20.10 | Chainsaw | mastahh: My servers have 4GB of RAM and two physical CPUs as well. Seems quite reasonable. |
| 08:20.33 | Chainsaw | mastahh: But jkroon does Asterisk on customer sites. Probably a better bet. |
| 08:20.38 | Chainsaw | handles barely any calls by comparison :) |
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| 08:27.09 | jkroon | Chainsaw, mastahh ?? |
| 08:27.30 | mastahh | jkroon |
| 08:27.37 | Chainsaw | jkroon: mastahh asked about hardware specifications of Asterisk servers. |
| 08:27.40 | mastahh | Thank you for your clarification Chainsaw |
| 08:27.57 | Chainsaw | jkroon: Particularly for customer sites, which I don't really do. |
| 08:28.30 | mastahh | jkroon, I cannot see any reason why we would need to exceed 4GB of RAM and 80GB of HDD on our central call processing server |
| 08:28.45 | jkroon | ah ok - anything i can assist with? I didn't get the question, but experience states that a single CPU and 200MB of RAM is MORE than adequate for just the asterisk portion, typically we use an i3 with 2GB of RAM with 2 x 1TB constellation drives in software RAID1 though. |
| 08:28.51 | mastahh | And voicemails won't really take THAT much space up if mailboxes are limited to a certain size |
| 08:29.23 | mastahh | It is most likely that we will deploy Cisco UCS C220 servers with all the servers virtualised on. |
| 08:29.43 | mastahh | Therefore, for all the servers in the UC solution, they will need to squeeze in to 32GB of RAM and 1TB of HDD space |
| 08:30.30 | mastahh | I just hoped that 4GB of RAM, two processors and a 80GB HDD would be sufficient for the entire life of the Asterisk solution instead of having to add additional HDDs to each server |
| 08:30.43 | jkroon | ok, so I just logged in on our bussiest switch, granted we've configured asterisk to only load what we need there, according to top it's using 231MB of RAM and <50% of *one* of the CPU cores. |
| 08:31.22 | jkroon | if you load a minimal OS on each of those go down to 2GB RAM. |
| 08:32.40 | jkroon | and as long as you only care about voicemail and not loads and loads of call recordings you'll be fine (we've got setups where we currently store around 4TB worth of recordings just to get to the requirements of storing recordings for 12 months - so storage really depends on your call volumes and whether or not your recording calls) |
| 08:33.21 | mastahh | We will be using a third party vendor to handle call recordings - unless you feel Asterisk is more than capable of doing this. It will save us (and our customers) a little bit of money |
| 08:33.24 | jkroon | for a typical "consumer" asterisk we're seeing memory usage from asterisk as low as 30MB, but typically around 50-80MB. |
| 08:33.53 | jkroon | mastahh, we've been throwing out other solutions in favor of :) |
| 08:34.13 | mastahh | That's perfect. Currently they are in a test bed while I build the servers and are running with 512MB of RAM but this is with no calls being processed while I build everything |
| 08:34.23 | mastahh | Ah, maybe I will look in to Asterisk call recording then! |
| 08:34.42 | jkroon | heads up though: Monitor() is tricky to use in a sensible way, so unless you really know what the risks are, use MixMonitor() rather. |
| 08:35.08 | mastahh | Noted. I am still new to Asterisk so that doesn't really mean much to me at the moment but I am sure it will soon! |
| 08:35.31 | jkroon | mastahh, pvt? |
| 08:35.36 | mastahh | Sure :) |
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| 08:46.01 | pawiecki | !help |
| 08:47.38 | WIMPy | ~ask |
| 08:47.39 | 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. |
| 08:51.11 | pawiecki | Hello, I have two questions: 1. How to get help/list of commands on this channel? 2. How to add timestamp to Asterisk CLI output? |
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| 08:53.17 | WIMPy | Don't know if infobot has such a thing. Maybe if you query it. Timestamps are enabled in asterisk.conf. |
| 08:55.08 | LnxBil | Hi everyone. We have an very old, proprietary PBX which will be replaces by asterisk. One thing that bothers me is that in the old PBX, after entering a two-digit number, the call automatically gets connected. There are only two-digit extensions. Can I simulate this with asterisk? Or is this dependend on the used phone (snom 821 in my case)? |
| 08:55.43 | WIMPy | It's a phone thing. |
| 08:56.05 | WIMPy | Well, with Snom phones you can actually leave it to Asterisk by enabling overlap dialling. |
| 08:57.43 | LnxBil | WIMPy: This is currently off for my phone. So if I enable this, the digits are evaluated while entering? |
| 08:57.58 | WIMPy | Yes. |
| 08:58.09 | LnxBil | WIMPy: Perfect. I'll try. Thanks |
| 08:58.27 | WIMPy | It needs to be enabled on both sides and you have to pay more attention to your dialplan. |
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| 09:01.05 | LnxBil | WIMPy: Works out of the box. I use 'Gemeinschaft 3.2' as asterisk fronend. |
| 09:01.34 | WIMPy | But that has been discontinued years ago. |
| 09:01.34 | LnxBil | WIMPy: I think I need to build extensions such that they do not overlap (e.g. 10 and 1000) |
| 09:02.01 | WIMPy | Yes. That's essential to make it work. |
| 09:02.10 | LnxBil | WIMPy: Yes, but there is still community support and the latest release of GS 3 is for Debian Wheezy, so kind of up-to-date |
| 09:04.05 | WIMPy | The trouble comes when you try to pass on calls via SIP again. That might work with chan_pjsip now. |
| 09:05.53 | LnxBil | WIMPy: Oh yeah, external dialing is broken now |
| 09:06.51 | WIMPy | AFAIK it works in all combinations of channeltypes except for sip to sip. |
| 09:07.41 | WIMPy | Unfortunately I think it's impossible to make it work unless your ITSP also supports it. |
| 09:08.12 | WIMPy | But I always say that SIP is not usable for telephony. |
| 09:08.56 | MaliutaLap | WIMPy: well not on it's own |
| 09:09.06 | MaliutaLap | WIMPy: it doesn't carry the audio data |
| 09:09.17 | MaliutaLap | SIP is like the matchmaker of Voip |
| 09:11.15 | WIMPy | It' the deal breaker. |
| 09:16.27 | WIMPy | The only way to get sip2sip in Asterisk is by using # as terminator from the dialplan. That way you only have to assist the machin in doing its work for external calls. |
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| 09:27.12 | LnxBil | WIMPy: What are current open source asterisk frontends that work quite well? What would you recommend? |
| 09:27.49 | WIMPy | I can't recommend any. But FreePBX seems to be the most maintained one. |
| 09:28.13 | WIMPy | Or as some would say: vi |
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| 09:48.57 | pawiecki | WIMPy: I enabled timestamp, but it shows only on warnings, notices and system informations, but not when there are normal calls running (for example == Spawn Extension... lines). Can I enable it to show on every line? |
| 09:50.13 | WIMPy | That's new to me. I see them on every line. |
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| 10:31.43 | mastahh | ~book |
| 10:31.44 | 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 |
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| 10:55.27 | WIMPy | Hmm. Is it just wishfull thinking or wasn't it possible to do something like ${var::4} instead of ${var:0:4}? |
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| 12:17.02 | [TK]D-Fender | WIMPy Hmm. Is it just wishfull thinking or wasn't it possible to do something like ${var::4} instead of ${var:0:4}? <- these are quite different. |
| 12:17.27 | [TK]D-Fender | WIMPy: remember it's ${var:trimleft:trimright} |
| 12:17.56 | [TK]D-Fender | WIMPy: If you use : at all you have to put a number for the left, then the right becomes optional |
| 12:18.08 | WIMPy | Yes they are. But somehow I thought you could leave out a 0. |
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| 12:38.43 | gavimobile | im expirencing an issue where dahdi isn't realizing that a call was hung up. the only thing so far which seems to solve this issue is running service dahdi restart. I've looked online but can't find anything... can someone shed some light |
| 12:44.44 | file | what kind of connection? |
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| 12:51.13 | MaliutaLap | hung up on what end? |
| 12:51.46 | MaliutaLap | what do the dahdi, chan_dahdi and related Dial() configurations? |
| 13:03.36 | [TK]D-Fender | MaliutaLap: You just completely the verb |
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| 13:07.24 | MaliutaLap | [TK]D-Fender: meh. Right now I have larger issues than proper grammar on IRC :P |
| 13:08.07 | [TK]D-Fender | I was thinking a bit more like "not enough to even read really..." |
| 13:08.10 | [TK]D-Fender | but ok :) |
| 13:09.24 | MaliutaLap | [TK]D-Fender: I guess verbing is better than nouning ;) |
| 13:09.36 | MaliutaLap | at least I did something about it ;) |
| 13:10.08 | Chainsaw | gavimobile: I have the same question as file. What type of telephone line? |
| 13:10.23 | MaliutaLap | [TK]D-Fender: http://pastebin.com/PGhbNty9 is more important right now |
| 13:12.41 | [TK]D-Fender | Seems you've made a new friend... |
| 13:12.45 | WIMPy | A fiver on POTS. |
| 13:13.50 | Chainsaw | WIMPy: Well I don't want to launch into the wrong rant |
| 13:13.54 | Chainsaw | WIMPy: Need to be sure here. |
| 13:14.18 | MaliutaLap | [TK]D-Fender: old. Old man, and neighbour of 20+ years. He's tried this on me, my brother, my parents ... pretty much my entire immediate family. Also has a history of domestic violence against I have no idea how many wives and his 3 kids |
| 13:14.56 | MaliutaLap | WIMPy: POTS is just plain |
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| 13:38.36 | seik0 | asterisk 1.8-last does not check if g++ is present |
| 13:41.02 | seik0 | ehm, strange, btw |
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| 13:42.33 | [TK]D-Fender | seik0: You really should be changing off 1.8 ASAP |
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| 13:45.01 | RadJackson | Hi , we are actually using G711 codec , we would like to migrate to G729 , in digium website it says 10$ per channel? does it mean that for 300 simultenous calls, we need to pay 3000 $ ? |
| 13:45.29 | [TK]D-Fender | for transcoding, yup |
| 13:45.43 | [TK]D-Fender | Don't forget that if you are transcoding... that is a big load collectively |
| 13:45.54 | Kobaz | loaaaad |
| 13:45.57 | WIMPy | At that amount getting a transcoder card might make more sense. |
| 13:46.53 | anonymouz666 | I wonder how many people use this transcoder card |
| 13:46.53 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Digium/Digium-TC400B |
| 13:46.58 | [TK]D-Fender | yup |
| 13:47.21 | Kobaz | anonymouz666: as telephonydepot how many they sold this year |
| 13:47.24 | anonymouz666 | 300 simultaneous calls is A LOT for asterisk to handle even without transcode in my opinion. |
| 13:47.59 | WIMPy | What do you do to/with your calls? |
| 13:48.13 | anonymouz666 | split your calls through asterisk instances |
| 13:48.29 | anonymouz666 | consider VMs, docker, etc. |
| 13:48.40 | Kobaz | i've never hit 300 calls |
| 13:48.42 | RadJackson | anonymouz666 trust me , we are able to run 450 simultenous calls on asterisk 13. and it works. |
| 13:48.44 | Kobaz | but i do okay with 200 calls |
| 13:49.00 | RadJackson | 450 outgoing calls. |
| 13:49.42 | anonymouz666 | I believe |
| 13:49.47 | Kobaz | outgoing into my braaaaiiiin |
| 13:50.54 | seik0 | [TK]D-Fender, I am not very ready =) |
| 13:50.55 | *** join/#asterisk kharwell (kharwell@nat/digium/x-qnfyhwvyuelxobwx) |
| 13:51.12 | [TK]D-Fender | seik0: Get ready |
| 13:51.23 | [TK]D-Fender | seik0: 1.8 goes EOL in 3 months |
| 13:52.05 | seik0 | [TK]D-Fender, o.0 |
| 13:52.19 | *** join/#asterisk theron (~theron@199.201.64.129) |
| 13:52.24 | seik0 | [TK]D-Fender, I still have two 1.4 instances and one 1.2 =) |
| 13:52.28 | anonymouz666 | 1.8 R.I.P did an amazing job. |
| 13:52.51 | RadJackson | [TK]D-Fender without Transcoding , is it for free to use G729 codec ? |
| 13:53.01 | anonymouz666 | seik0: do you like deadlocks? he he |
| 13:53.26 | seik0 | anonymouz666, in which module? =) |
| 13:53.28 | WIMPy | RadJackson: You wouldn't use it then. |
| 13:53.37 | WIMPy | Not on Asterisk, that is. |
| 13:53.40 | [TK]D-Fender | RadJackson: * can PASSTHROUGH, but you may find that hard to survive. Every sound file will have to be in g.729, and forget conferencing, spying, etc |
| 13:54.16 | [TK]D-Fender | RadJackson: All call recording would have to be in g.729 as well. Oh.. and no TONETS generated for things like ringing, because tones are not "looped", they are generated |
| 13:54.48 | seik0 | [TK]D-Fender, a month ago tried to start 13 with nearly configs of 1.8, but failed |
| 13:55.00 | seik0 | [TK]D-Fender, so, became sad and postponed |
| 13:55.06 | RadJackson | Ok so better use g729 with transcoding , if not i would rather stick to g711 |
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| 13:55.16 | RadJackson | [TK]D-Fender thanks a lot! |
| 13:55.33 | RadJackson | Is the license price 10$ per month? or 10$ for good? :) |
| 13:55.36 | [TK]D-Fender | RadJackson: I'd seriously stick to G.711 if you can. Compressed codecs are for the desperate |
| 13:55.56 | [TK]D-Fender | RadJackson: $10 per simulataneous channel per server for its life |
| 13:56.52 | MaliutaLap | RadJackson: the Digium licences are tied to the network devices on the machine |
| 13:57.03 | [TK]D-Fender | MAC is a kep |
| 13:57.05 | [TK]D-Fender | key |
| 13:57.26 | MaliutaLap | RadJackson: for this reason I have always used a PCI Intel NIC and disabled any onboard |
| 13:57.47 | MaliutaLap | I have kept my licences over 3 instances of hardware this way |
| 13:58.54 | MaliutaLap | RadJackson: just keep that in mind, you can also add extra licences at any point |
| 13:59.40 | MaliutaLap | [TK]D-Fender: here in the bandwidth poor .au we need compressed codecs :( |
| 14:00.30 | [TK]D-Fender | MaliutaLap: I did say desperate :) |
| 14:01.26 | RadJackson | [TK]D-Fender actualy we have been using g711 for more than two years , we never had any problems with. But lately due to the huge amount of demands we are receiving from our customers requesting SIP TRUNKS, we are discussing a new contract with a voip provider , we asked for more than 500 cac (call admission control) per asterisk instance, they recommanded G729 as it use less bandwidth than |
| 14:01.26 | RadJackson | g711 (8Kbits compared to 64 Kbits). |
| 14:01.41 | RadJackson | they even told us , 90% of theirs customers are on G729... |
| 14:01.47 | RadJackson | actually it is a big provider here in France. |
| 14:02.51 | MaliutaLap | RadJackson: if bandwidth usage is a concern then G729 is a good bet, if not the G711 is fine |
| 14:07.13 | Gostega_ | Cyford33: good morning |
| 14:08.16 | Gostega_ | ready when you are |
| 14:08.22 | [TK]D-Fender | GSM is also not bad if it's an option. G.729 is smaller, but between cost & CPU maybe GSM is a better fit for you |
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| 14:21.03 | MaliutaLap | bedtime for this little black duck |
| 14:27.33 | marceloamorim | guys, do you know if there is a way to send to display(phone) informations about other phone? Like BFL thing? "my idea" is send callerid that are ringing in another phone, but without ring |
| 14:28.35 | WIMPy | That is possible. Don't ask about details. But I had that some years ago |
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| 14:29.56 | marceloamorim | nice WIMPy, I'm thinking about using sipaddheader, but I'll test =) |
| 14:30.13 | WIMPy | You don't have to. |
| 14:30.57 | [TK]D-Fender | marceloamorim: Not happening. sipaddheader adds a header to a call. |
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| 14:36.25 | marceloamorim | I don't know another way to send sip info =( |
| 14:36.54 | [TK]D-Fender | SIP MESSAGE |
| 14:37.05 | [TK]D-Fender | well documented on the WIKI |
| 14:37.20 | WIMPy | You don't have to do anything manually. |
| 14:38.06 | [TK]D-Fender | Some BLF support passing the CID. Not sure what the convention is to support that exactly though |
| 14:38.34 | [TK]D-Fender | wait |
| 14:38.41 | [TK]D-Fender | "notifycid=yes |
| 14:38.45 | [TK]D-Fender | under [generral] IIRC |
| 14:39.00 | [TK]D-Fender | You could try that and see if your phones support it |
| 14:39.12 | marceloamorim | nice, I'll check too |
| 14:39.18 | WIMPy | IIRC there was something about allowing something for subscriptions as well. |
| 14:39.25 | [TK]D-Fender | First thing to try since it's the easiest |
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| 14:41.13 | WIMPy | As I said: It does work. Or at least it did. I hardly ever have more than on IP phone so I wouldn't notice. |
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| 14:54.02 | mastahh | Quick question; the modules AddQueueMember(), RemoveQueueMember() etc. if I wanted to declare them as an extension for agents to dial, do I put these in the extensions.conf file? |
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| 15:03.41 | [TK]D-Fender | mastahh: extensions = extensions.conf |
| 15:03.47 | [TK]D-Fender | mastahh: Everythign you dial is in there |
| 15:06.27 | mastahh | I feel that my assumption is correct then, thanks [TK]D-Fender - we will find out haha |
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| 17:27.01 | WangDang_ | Hi, just upgraded a Debian box to jessie which upgraded asterisk to 11.13 from 1.8. Now * doesn't pick up the phone when a pots line calls, and doesn't place a call when pots handset dials a call. During the upgrade, I was asked if I wanted to install new versions of configuration files. I chose to keep my existing ones. "dahdi show channels" lists only a pp |
| 17:27.38 | WIMPy | pp? |
| 17:28.02 | WIMPy | Anyway you need ro rebuild dahdi. |
| 17:28.37 | WangDang_ | (oops) pseudo device. It should show two fxs and one fxo channel. I've read https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 and nothing there seems to offer help. Where do I go from here? |
| 17:29.14 | WangDang_ | does a rebuild not happen as part of the upgrade? I'm using debian stable packages rather than compiling myself |
| 17:30.05 | WIMPy | I have no idea. Maybe you need to retrigger it after you restarted the upgraded system. |
| 17:30.38 | WangDang_ | retrigger what, the rebuild of dahdi? |
| 17:31.44 | WIMPy | I still have no idea. It's horrible yo build anythign on Debian. |
| 17:32.07 | WIMPy | And yes there should be some automatism as far as I understand. |
| 17:33.00 | WangDang_ | should I see some dahdi or * related kernel modules when I run lsmod, or does that kind of stuff only show up at the CLI> prompt? |
| 17:33.53 | WIMPy | Threre should be modules. |
| 17:34.02 | WIMPy | And that's probably what's missing. |
| 17:34.36 | WIMPy | ... and needs to be rebuild for the current system. |
| 17:34.56 | WangDang_ | There's the problem then. I'd try to reload them but I don't know what they are called. How does one look up such a thing? |
| 17:35.17 | WIMPy | The main module is just called dahdi. |
| 17:35.36 | WIMPy | Others depend on your hardware. |
| 17:37.09 | WangDang_ | h/w is i386 with a digium card that has two fxs and one fxo module on it |
| 17:38.48 | WangDang_ | just tried to modprobe dahdi. Got the error FATAL: Module dahdi not found |
| 17:39.31 | WIMPy | YEs, you still need to build them for your current system. |
| 17:39.48 | WIMPy | And once they are installed, everything should be back to normal. |
| 17:41.11 | WangDang_ | OK, I'll look around for docs on rebuilding the asterisk related kernel modules in Debian. Thanks WIMPy |
| 17:41.41 | WIMPy | DAHDI |
| 17:41.50 | WIMPy | Not Asterisk, even if it's related. |
| 17:42.27 | WangDang_ | OK, gotcha I'm specifically going to rebuild the dahdi modules. Thanks :) |
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| 17:54.04 | zizo | Hi all, noob here :) In my dialplan there is an extension which, if dialed, asks for a number to dial. Now you can say the number and an agi converts it into digits, but i'd like to allow also a dtmf reading. |
| 17:55.33 | zizo | So i am thinking about two solutions: record a file, try with dtmf, if it fails, try with speech; or better can i exec two commands on the same line and pick up the result of the one which isn't failing? Thanks |
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| 17:57.33 | WIMPy | It's entirely up to you what you do in your AGI. |
| 17:59.33 | zizo | WIMPy: ok, but i'd like to do it in the dialplan, so basically: is there a way for executing two parallel commands? |
| 17:59.43 | WIMPy | No |
| 17:59.55 | WIMPy | The dialplan is 100% sequential. |
| 18:00.07 | zizo | WIMPy: thank you |
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| 18:37.17 | [TK]D-Fender | ~polls |
| 18:37.22 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
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| 19:13.23 | Casper | Hi there, I am unsure of what my voip provider use, but should be close enought... Basically, I'm getting lots of incomming call with no audio. The person call back right after and there is audio... |
| 19:14.39 | Casper | can it be any of the voip phone or router config? trying to close every possible doors on my side since the provider seems to have their generic answers in repeat mode.... |
| 19:16.31 | [TK]D-Fender | Could be any piece of this solution |
| 19:16.48 | [TK]D-Fender | First is to prove that your server is treating the other parties properly |
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| 19:21.36 | Casper | hmmm found a router misconfig... maybe.. |
| 19:22.18 | Casper | tell me, if there is no port forwarding on the router for the phone, that would make a no-ringing right? not just a no incomming audio right? |
| 19:23.39 | [TK]D-Fender | You generally never need to forward clients |
| 19:24.12 | Casper | ok, so this is fine then |
| 19:25.02 | Casper | thing is... I'm sure the issue is on their side... but they keep repeating the same thing: since it's intermittant, it have to be on your side. Check your config. |
| 19:27.00 | Casper | Do you know if openwrt is known to cause issue? just to try to close another door... |
| 19:31.11 | [TK]D-Fender | sothing specific, but it's not great in my books |
| 19:31.29 | [TK]D-Fender | And that doesn't mean you aren't running things like proxy or ALG on top of it which can screw with thuff |
| 19:31.32 | [TK]D-Fender | stuff* |
| 19:34.05 | Casper | basically, at work, 5-10% of the calls or so get no incomming audio |
| 19:34.30 | Casper | and it come and go, personally, I think their server run out of outgoing port or something |
| 19:34.39 | Casper | there is even some line mixing issue at time |
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| 19:35.22 | [TK]D-Fender | I'd check ALG & proxy first |
| 19:35.29 | [TK]D-Fender | then prove the rest is responding right |
| 19:35.34 | Casper | what is ALG? |
| 19:35.57 | Casper | I don't recall having seen that anywhere on the router... and no proxy on the router too... |
| 19:40.36 | [TK]D-Fender | Application Layer Gateway. |
| 19:40.41 | [TK]D-Fender | a SIP "helper" if you will |
| 19:40.48 | [TK]D-Fender | which generally only screws stuff up |
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| 19:45.03 | Casper | ok, I wonder if openwrt being linux based... with a qos and stuff... if that can cause issue with trafic identification... but if so... the issue would be more common no? |
| 19:45.34 | [TK]D-Fender | "stuff" is a bad thing to have in the equation |
| 19:45.54 | [TK]D-Fender | You need to start moving on the usual suspects |
| 19:46.16 | [TK]D-Fender | Go prove what your calls are looking like |
| 19:46.32 | Casper | what do you mean? |
| 19:46.40 | ldc | openwrt has no sip alg unless you're using a specific package |
| 19:46.43 | ldc | kmod-ipt-nathelper |
| 19:47.10 | ldc | (actually -extra for sip) |
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| 19:51.54 | Casper | ldc: so if I have kmod-ipt-nathelper but not kmod-ipt-nathelper-extra I'm fine? |
| 19:52.30 | ldc | according to the docs, yes |
| 19:52.36 | ldc | <PROTECTED> |
| 19:52.40 | ldc | <PROTECTED> |
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| 19:56.13 | Casper | good thanks |
| 19:58.14 | Casper | and since I have no voip proxy, the only issue could be the not required port forwarding, port overloading on the router (wouln't that cause a no-ringing?) and phone misconfig (which I use their sample config) |
| 19:59.27 | [TK]D-Fender | server misconfig, etc... |
| 20:01.51 | Casper | server misconfig sure, this is my guess... but it is not my server, so... can't diagnose that side |
| 20:02.18 | Casper | but if I can be sure that it is not on my side... I can come with the proof somewhat |
| 20:02.27 | Casper | I put the forward, so... should be fine |
| 20:02.33 | Casper | well, finer... |
| 20:02.54 | [TK]D-Fender | Not being able to diagnose the server side is a |
| 20:02.54 | Casper | What is RTP? |
| 20:02.57 | [TK]D-Fender | ~soso |
| 20:02.57 | infobot | [~soso] Shoot-On-Sight Offense |
| 20:03.19 | [TK]D-Fender | "should be" doesn't cut it. |
| 20:03.28 | [TK]D-Fender | RTp is the actual voice |
| 20:04.02 | Casper | tell me... do I have to forward it? or will the phone establish an outgoing connection? |
| 20:05.53 | [TK]D-Fender | [15:23][TK]D-FenderYou generally never need to forward clients |
| 20:06.06 | [TK]D-Fender | You need forwarding to your server, and for it to be properly configured |
| 20:12.24 | Casper | that's the only thing that bug me right now... their exact line is: "In your router please make sure to have the ports 5060-5080 UDP and RTP range 10001-20000 UDP open/forwarded." |
| 20:12.52 | [TK]D-Fender | You need to confirm if that is even the proper range |
| 20:12.55 | [TK]D-Fender | rtp.conf <---- |
| 20:13.56 | Casper | as I said, I do not have access to the server.. only the client |
| 20:14.38 | Casper | so I want to make sure that the client is proprelly configured before I become more agressive with the provider |
| 20:15.10 | [TK]D-Fender | Generally there is nothing to set for clients. |
| 20:15.27 | [TK]D-Fender | Asterisk needs to know what to trust from them and that there isn't anything in the path that will screw with the conversation |
| 20:16.11 | Casper | so the RTP on the client side is outgoing connection? |
| 20:16.25 | [TK]D-Fender | Get aggressive on access. You're in a position equivalent to "My car won't start and I've looked just about everywhere except under the hood which is prohibited". |
| 20:16.54 | [TK]D-Fender | RTP should only be allowed first from client>server, never client>client |
| 20:17.50 | Casper | good |
| 20:18.33 | Casper | so I'll have my boss step in and threaten to move out the 60 lines then |
| 20:29.51 | [TK]D-Fender | heads home |
| 20:30.35 | Casper | thanks guys, missed d-fender... but ow well, thanks |
| 20:38.22 | WangDang_ | I've just upgraded a Debian box which moved me from * 1.8 to 11.13. Dahdi channels aren't showing up at the CLI> dahdi show channels. WIMPy helped me get the dahdi kernel modules recompiled. The dahdi kernel modules now show up at lsmod, but I still don't see my 2 fxs and 1 fxo channels showing up in *. Any suggestions where to go from here? |
| 20:51.34 | newtonr | i'd check logs at Asterisk start to see if chan_dahdi is loading and what it spits out |
| 20:51.54 | newtonr | probably want to check logger.conf first to make sure you have the various log channel types going to the log |
| 20:52.10 | newtonr | and make sure you start Asterisk with a high verbose level |
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| 20:56.07 | rmudgett | You should also check if the DAHDI service is started before Asterisk is started. |
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| 21:17.30 | pjensen00 | mjordan: is there a way to get stereo instead of mono for call recording in ARI? |
| 21:18.37 | DivideBy0 | pjensen00: I think I was able to do that by setting the format, lemme go look |
| 21:19.20 | pjensen00 | awesome! |
| 21:20.46 | WangDang_ | newtonr: once I found the right log, I found something interesting. http://paste.debian.net/hidden/3397eadf/ is from /var/log/asterisk/messages. "# /etc/init.d/dahdi restart" wrote nothing to the logs, but "# /etc/init.d/asterisk restart" dumped the above paste. Interesting bit is lines 29 through 40 inclusively. Has the chan_dahdi.conf syntax changed between 1.8 and 11.13? Otherwise, why would I all of-a-sudden have and invalid a |
| 21:22.37 | WIMPy | No. That means the channels don't exist. i.e. dahdi was not correctly loaded or configured. |
| 21:22.43 | *** part/#asterisk Casper (~ThePhanto@unaffiliated/thephantom) |
| 21:23.06 | WIMPy | try 'dahdi_cfg -s;dahdi_cfg -v'. |
| 21:25.09 | pjensen00 | DivideBy0: is there a list of formats that are available by default? I haven't found a definitive source in the wiki |
| 21:25.42 | DivideBy0 | I'm looking for that right now |
| 21:25.55 | DivideBy0 | I saw it somewhere |
| 21:26.04 | DivideBy0 | somewhere very hidden |
| 21:27.29 | pjensen00 | :( sadness and misery |
| 21:27.44 | pjensen00 | but that would explain why i couldn't seem to find it. |
| 21:27.45 | DivideBy0 | I have an old list of this: "The supported sound formats are: sln, g723, g729, gsm, h263, ulaw, alaw, vox, wav, WAV (WAV is the GSM format of wav files)." |
| 21:27.53 | DivideBy0 | but I think I found a new one. still looking... |
| 21:28.01 | WangDang_ | WIMPy: here's the output of those two dahdi_cfg commands : http://paste.debian.net/hidden/883c2b17/ |
| 21:28.18 | WangDang_ | the signalling before the upgrade worked just fine |
| 21:28.54 | WIMPy | Looks like the correct module for your card isn't loaded. |
| 21:29.00 | DivideBy0 | A "core show translation" from the cli will show you what formats it can use. I'm still looking for better info |
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| 21:29.29 | DivideBy0 | and core show codecs |
| 21:29.30 | WIMPy | Was is dahdi_scan or dahdi_hardware? One of them tells you what it find and what module it thinks is needed. |
| 21:29.35 | pjensen00 | Aight. The wiki should get updated |
| 21:29.46 | pjensen00 | I thought the "core show codecs" was just a list of possibly configured ones? |
| 21:29.54 | WIMPy | 'core show file formats'? |
| 21:31.10 | pjensen00 | oh that's a neat list. |
| 21:31.56 | WangDang_ | dahdi_scan silently returns to the command prompt. dahdi_hardware yields the following : "pci:0000:00:09.0 wctdm+ e159:0001 Wildcard TDM400P REV I" |
| 21:32.25 | WIMPy | Hmm. That would mean that the correct module is loaded. |
| 21:32.47 | WIMPy | Has your /etc/dahdi/system.conf been reset? |
| 21:33.09 | WangDang_ | Ya, I just did an lsmod and it's there |
| 21:33.17 | WangDang_ | what do you mean by reset? |
| 21:33.32 | WIMPy | To default/empty. |
| 21:33.33 | DivideBy0 | I'm honestly not sure about codecs |
| 21:33.51 | pjensen00 | Mm, that's ok |
| 21:33.55 | DivideBy0 | but that should at least make it a multiple choice question instead of a fill in the blank :) |
| 21:33.56 | WangDang_ | WIMPy: no, the file is the same |
| 21:33.58 | pjensen00 | I'm mostly worried about seeing if I can get the stereo |
| 21:34.22 | pjensen00 | Currently in my agi script, I ust MixMonitor and then mux the audio files in post |
| 21:34.52 | pjensen00 | but in ARI world, we're looking to do more than just 2 channel calls and our support team uses Stereo for troubleshooting or QA |
| 21:35.01 | WIMPy | WangDang_: I'm ot sure what could go wrong then. |
| 21:36.08 | newtonr | WangDang_, pastebin the system.conf file |
| 21:36.24 | newtonr | WangDang_, and your chan_dahdi.conf |
| 21:36.59 | DivideBy0 | stack overflow said to send it there, then back to ari |
| 21:37.57 | pjensen00 | Hrm, I don't know if that will work so well in our system. |
| 21:38.16 | pjensen00 | I guess I'll just have to record each channel individually |
| 21:40.37 | pjensen00 | So your search for your example of stereo didn't pan out? |
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| 21:45.39 | DivideBy0 | I thought it one of the formats was stereo |
| 21:45.51 | DivideBy0 | I can lookup each one if you want. I have a few mins of downtmie |
| 21:46.16 | WIMPy | Would be news to me. But I'm only on 11. |
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| 21:47.28 | WangDang_ | newtonr: see http://paste.debian.net/hidden/83b7d536/ for the contents of system.conf and chan_dahdi.conf |
| 21:47.42 | pjensen00 | That would be amazing |
| 21:47.58 | pjensen00 | I'm plinking away at this as I stare at the clock waiting for vacation to start |
| 21:50.24 | DivideBy0 | this is the codec page I was looking for. It has no mention of stereo: https://wiki.asterisk.org/wiki/display/AST/Codec+Modules |
| 21:51.57 | DivideBy0 | speex support "intensity stereo encoding" but I dunno what that is (yet) |
| 21:52.17 | pjensen00 | ah, I found that one yesterday. I was hoping there would be a secret page somewhere where the magical christmasland was |
| 21:54.26 | DivideBy0 | it looks like you really want Opus - which obsoleted speex |
| 21:54.31 | DivideBy0 | lots of stereo talk on that one |
| 21:54.59 | pjensen00 | hrm. I'll give it a shot and see if there's hope on that one. Thanks! |
| 21:55.12 | DivideBy0 | opus does stereo, but I have no idea how to tell it to encode that way |
| 21:55.32 | WIMPy | AMR-WB+ also does stereo. |
| 21:56.20 | WIMPy | But non-free |
| 21:56.48 | newtonr | WangDang_, Odd. I don't know. You probably want to gather up all your logs and debug and make a post on the forums. Someone there may be able to spend a bit longer looking at it. |
| 21:57.59 | pjensen00 | Hrm. Will it default to mono every time? |
| 21:58.20 | DivideBy0 | I cant tell yet |
| 21:58.22 | WangDang_ | OK, will do. Thanks for taking a swing or two at it :) |
| 22:01.28 | DivideBy0 | codec_opus.c doesn't mention the word "stereo" |
| 22:01.46 | pjensen00 | ;_; |
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| 22:02.45 | DivideBy0 | yeah, stereo can somehow be set. I see it in in res_format_attr_opus.c |
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| 22:06.05 | DivideBy0 | but I can't see how to set any attributes via ARI |
| 22:06.25 | pjensen00 | Yeah that's why I got worried when playing around with it. |
| 22:11.54 | DivideBy0 | I'm not sure without trying it. Lemme know what you find out when you get back. |
| 22:31.20 | pjensen00 | hrm, I appear to have libopus0 and it shows up in "core show codecs" but ARI doesn't seem to recognize it |
| 22:32.27 | Micc | Is there an AMI action I can use to get the ExtensionState of all hints in the system? |
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| 22:36.40 | rmudgett | misc: "manager show command ExtensionStateList" |
| 22:37.21 | rmudgett | Micc: ^^^ |
| 22:37.41 | pjensen00 | AH, it's because it's only for passthrough, not for transcoding. Sadness and misery. |
| 22:39.27 | mastahh | Has anyone ever deployed ejabberd with Asterisk before? I was going to use Spark but I think it is extremely out of date now isn' it? Source code for the Asterisk-IM plugin was last updated in 2006 and is no longer available anyway. Any suggestions? |
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| 22:42.25 | Micc | rmudgett: thanks. I guess I don't have that on 1.8, but ExtensionState should work fine, just have to do it for each extension I care about. |
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