| 00:00.25 | linuxgeek | and they wouldnt route a 100 % of the calls - 70 / 75 % would hit them |
| 00:02.12 | ChannelZ-Wk | That's still pretty high... a noticable drop in call volume (assuming regular volume was consistent enough) |
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| 00:04.16 | linuxgeek | ChannelZ-Wk : from a certain destination , its pretty difficult to notice |
| 00:04.45 | linuxgeek | another thing these guys do , they call series of numbers and find out the ones not issued / in use and target them |
| 00:05.43 | WIMPy | Does that mean I invite them if I have numbers allocated but tell callers they're not? |
| 00:06.09 | WIMPy | ... which Asterisk does by default anyway. |
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| 01:33.10 | Penguin | Asterisk doesn't do that by default. |
| 01:34.35 | Penguin | In order for your unallocated number to have a message saying it's not in service, you have to have a matching extension actually exist and execute a playback of such audio files. |
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| 03:09.11 | hariom | I am planning to use asterisk for SIP communication only. No Hardward cards are used. Is it required to install Dahdi and LibPRI? |
| 03:09.52 | Penguin | That depends on your definition of "required." |
| 03:10.00 | Penguin | Oh, and your actual needs. |
| 03:10.21 | ChannelZ | But generally no. |
| 03:10.27 | Penguin | And maybe your asterisk version. I don't know if the same needs for dahdi still exist in 10+. |
| 03:10.39 | ChannelZ | DAHDI would be needed if you were living in the past on an old asterisk and needed MeetMe |
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| 03:12.44 | hariom | ChannelZ: I don't have immediate use of Meetme but may in the future. But it is for sure that I am not using telephony cards as I am runing * in AWS |
| 03:13.18 | hariom | ChannelZ: so looks like I should install Dahdi due to timing module dependency but I can skip libpri |
| 03:13.24 | Penguin | I'm not using any cards either, and I need dahdi. |
| 03:15.22 | ChannelZ | Well, maybe if you're installing as packages yes, because they're forcing some module (like meetme) you don't need |
| 03:15.52 | ChannelZ | If you're running Asterisk 10 or above, you can use ConfBridge (MeetMe is pretty much dead) which doesn't require DAHDI. |
| 03:16.20 | ChannelZ | (Yes ConfBridge was in earlier than that but I believe it was in 10 that it pretty much got feature-par with MeetMe, the first implementation of ConfBridge was basic) |
| 03:17.19 | hariom | ChannelZ: I think dahdi required for res_timing_dahdi but not sure what if res_timing_dahdi not present |
| 03:17.43 | hariom | res_timing_dhadi is required for SIP timing? |
| 03:17.49 | Penguin | No. |
| 03:17.51 | ChannelZ | no |
| 03:18.21 | ChannelZ | It was pretty much only needed for MeetMe and like one other thing that I'm not remembering this second |
| 03:18.22 | hariom | ok. So for now I can safely go ahead without dahdi and libpri |
| 03:18.53 | Penguin | You never did state your asterisk version (or even which branch) that you're installing. |
| 03:19.00 | hariom | ChannelZ: I am just using AGI at this moment |
| 03:19.12 | hariom | 11.6 Cert LTS |
| 03:19.20 | ChannelZ | Well AGI is unrelated |
| 03:19.25 | hariom | yea |
| 03:19.31 | hariom | Thought to share some info |
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| 03:39.17 | Penguin | So I've been thinking about tc -s class ls... |
| 03:39.38 | Penguin | ls isn't even a valid tc command according to the manual. |
| 03:40.18 | Penguin | add, delete, change, replace, link |
| 03:42.52 | Penguin | And what I mean by this is that 'tc -s class ls' does not generate any output for me whatsoever. Straight back to the prompt. |
| 03:43.16 | ChannelZ | you have to give it an interface with 'dev xxx' |
| 03:43.51 | ChannelZ | I agree it's not in the docs but it's doing something. tc is a black art |
| 03:44.02 | Penguin | If I use tc -s class show dev eth1, that gives me info. |
| 03:44.05 | *** join/#asterisk jzaw (~jzaw@community-wifi.dzki.co.uk) |
| 03:44.32 | ChannelZ | OK yeah that looks the same. ls must be an alias for show. |
| 03:44.59 | ChannelZ | An undocumented one, but there you go |
| 03:45.09 | Penguin | And now that I have run that, I see why you didn't approve my answer regarding rate. |
| 03:45.48 | Penguin | When I replied about rate, I was talking about rate in this line: |
| 03:45.51 | Penguin | class htb 1:a parent 1:1 leaf 8014: prio 0 quantum 24750 rate 1980Kbit ceil 4400Kbit burst 15Kb/8 mpu 0b overhead 0b cburst 1599b/8 mpu 0b overhead 0b level 0 |
| 03:45.53 | ChannelZ | Yeah I was talking of the statistics rate it shows for the class, not the rate you are configuring. |
| 03:45.59 | Penguin | Where rate is the bandwidth I have allocated. |
| 03:46.24 | Penguin | But now I see that you were talking about: |
| 03:46.27 | Penguin | <PROTECTED> |
| 03:46.29 | ChannelZ | I'm talking about the subclasses |
| 03:46.31 | ChannelZ | yes |
| 03:47.00 | ChannelZ | It's basically the current rate through that class. But it's averaged over a fairly long period of time. |
| 03:47.40 | Penguin | I can tell you that it only changes once every 5th second. |
| 03:48.02 | Penguin | For four seconds, it remains unchanged. |
| 03:48.05 | ChannelZ | Like earilier I had 1 call going, so my voip priority class had about 85kbit/sec going through it. When the call ended it took like 15+ seconds for it to work its way down to near-0 |
| 03:49.40 | Penguin | I doubt this information would be in a typical document about tc. You might find it in the source, though. |
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| 03:53.25 | Penguin | I will say that it is not apparent. |
| 03:53.49 | Penguin | I did a test from 0bit, make a single call... |
| 03:54.10 | Penguin | It jumped to near 20000bit as soon as RTP was engaged. |
| 03:54.39 | Penguin | I ended the call as soon as I saw any rate >0. And then it took about 2 minutes for it to drop back to 0 again. |
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| 03:57.02 | ChannelZ | Yeah. I guess it's not entirely important, I just wondered if someone happened to know |
| 03:57.39 | Penguin | I thought I did, but I was telling you about a different rate from the one you were asking about. |
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| 04:31.53 | hariom | I have successfully installed asterisk. I have now created a system user "asterisk" without home directory and password. Ownership of asterisk files, log, run directories assigned to "asterisk" user and also changed runuser and rungroup in asterisk.conf |
| 04:32.11 | hariom | But when I connect asterisk as "asterisk -cvvvvvvvv" I get this msg: http://pastebin.com/31yJsCc5 |
| 04:32.57 | hariom | I am logged in as a normal user other than "asterisk". |
| 04:34.07 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 04:37.03 | Penguin | hariom: What does 'getent passwd asterisk' print out? |
| 04:37.36 | Milos | [Nov 1 14:47:35] WARNING[11852]: chan_sip.c:3755 __sip_xmit: sip_xmit of 0x7f17780683c0 (len 544) to 192.168.2.10:54660 returned -1: Operation not permitted |
| 04:37.38 | Milos | dafuq |
| 04:37.40 | Milos | just saw this in my logs |
| 04:37.41 | hariom | Penguin: asterisk:x:108:115::/var/lib/asterisk:/bin/false |
| 04:38.03 | Penguin | Okay, so the home dir and shell are set to something sensible. |
| 04:38.04 | Milos | there's about 4 of them in the space of 30 seconds and then went away |
| 04:38.08 | Milos | should I be investigating something? |
| 04:40.21 | hariom | Penguin: |
| 04:41.45 | ChannelZ | If asterisk is trying to chown but you're running it as a different user, it can't. |
| 04:43.20 | *** join/#asterisk theron (~theron@173.252.71.189) |
| 04:44.15 | ChannelZ | IE if you run it as root it will be able to change its identity to asterisk. But when run as some other non-priviledged user, it can't. |
| 04:44.53 | Penguin | Use the init script or safe_asterisk to start it. |
| 04:47.36 | ChannelZ | Milos: Yes and no. Seeing as the IP was a LAN one, I assume those popped up when someone hung up in the middle of a playback, like voicemail or something. |
| 04:47.55 | ChannelZ | Or the peer unexpectedly died |
| 04:56.58 | Milos | ah ok |
| 04:57.01 | Milos | must have died |
| 05:28.05 | ChannelZ | Dangit. I still haven't figure out why my traffic shaping kills Twitch streams when some other app is downloading something.. like Steam downloading game updates, or Adobe DLing updates. |
| 05:29.05 | Penguin | What's twitch, and does it have a service type on its packets? |
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| 05:29.27 | Penguin | or a specific port? |
| 05:30.01 | ChannelZ | twitch.tv, live game streaming. Via Flash (uggggh) |
| 05:32.06 | ChannelZ | I guess I need to actually look and see if the stream is UDP or TCP, I don't actually know. |
| 05:32.53 | Penguin | If it has a specific port or a type, that would make it a lot easier to set a guaranteed bandwidth for it. |
| 05:32.57 | ChannelZ | every damn time I launch Steam though it seems there's a new Team Fortress update. Ugh. |
| 05:32.59 | Penguin | Or if they only have one IP address. |
| 05:33.20 | Penguin | Any unique bit of info would be useful. |
| 05:34.20 | ChannelZ | Well this is all incoming of course. But I would figure I should be getting a decent balance on the incoming traffic I'm using SFQ on my outgoing, so the ACKs would be fairly distributed and not allowing one download to drown out another. |
| 05:37.11 | Penguin | It would also depend on which interface you've applied your qdiscs to. |
| 05:38.34 | Penguin | I don't try to control any traffic other than outbound of my WAN-connected interface. |
| 05:38.42 | Penguin | Nothing else really matters. |
| 05:38.47 | ChannelZ | Yeah that's where mine's all at. |
| 05:42.56 | Penguin | I do have a slight disadvantage with tc. I don't ever have to manually run any of the tc commands since I am using VyOS on my router. I just have to configure the stuff, and, when applied, it gets translated to commands behind the scenes. |
| 05:44.19 | ChannelZ | I need to ponder my config a bit and see if I can figure out if there's something I'm doing badly, or anything I can do about it. |
| 05:46.25 | ChannelZ | I guess I can try to debug this somehow by using tcpdump or something to look at the ACKs and see if they're clumping for some reason, or something. Hmm. |
| 05:46.41 | Penguin | It can't hurt. |
| 05:47.16 | Penguin | But with sfq, that really shouldn't happen at all. |
| 05:47.34 | ChannelZ | Yeah you'd think. |
| 05:47.56 | Penguin | That's what that qdisc is supposed to prevent. |
| 05:47.59 | ChannelZ | I need something to eat first. BBL |
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| 10:37.02 | hariom | I am trying to load chan_sip.so in modules.conf file as: required => chan_sip.so but it is not loading and asterisk is failing to start. What could be the reason why chan_sip.so is not loading. Could you help in debug? |
| 10:38.12 | WIMPy | A broken config file. |
| 10:38.48 | WIMPy | Try to load it manually with verbose and debug enabled. |
| 10:39.01 | hariom | WIMPy: http://pastebin.com/AbhuP1fh |
| 10:40.14 | WIMPy | Looks like you need more verbose/debug. |
| 10:41.15 | hariom | WIMPy: How do I enable more verbose/debug? |
| 10:41.50 | WIMPy | -vvvvvddd e.g. |
| 10:42.42 | hariom | WIMPy: Same result even after adding 7 times d |
| 10:44.24 | WIMPy | Then do it really manually. Remove it from your modules.conf and try to manually 'module load chan_sip' after setting verbose and debug to some high values (like 9 or 99). |
| 10:45.02 | hariom | WIMPy: http://pastebin.com/swvVxYjk |
| 10:46.32 | hariom | WIMPy: Surprising, "module load chan_sip" worked fine |
| 10:46.57 | WIMPy | You're not loading the websocket module. |
| 10:49.11 | hariom | WIMPy: ah! so that is a dependency for chan_sip.so |
| 10:49.38 | WIMPy | That's unfortunately not optional. |
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| 10:51.23 | hariom | WIMPy: ok, I have added "load => res_http_websocket.so.so" before chan_sip.so but still it is failing to load |
| 10:52.26 | WIMPy | Let's se... you don't have any rtp modules, either. |
| 10:53.33 | WIMPy | And without any bridging modules, I don't think you will get very far, even after it loads. |
| 10:53.59 | hariom | WIMPy: you mean res_rtp_asterisk.so |
| 10:54.36 | hariom | WIMPy: I am using this installation only for AGI based IVR |
| 10:54.50 | hariom | and only via SIP. |
| 10:55.04 | WIMPy | Ok, you might get away without bridging then. |
| 10:55.35 | WIMPy | So what's chan_oss about? |
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| 10:56.36 | hariom | WIMPy: it is loaded by default I think. Either chan_oss or chan_alsa |
| 10:57.06 | WIMPy | looks at that modules.conf and wonders "by default"??? |
| 10:59.15 | hariom | WIMPy: It was mentioned in the default modules.conf that chan_oss is loaded by default |
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| 11:02.51 | hariom | WIMPy: If I remove "required" and use "load" then asterisk starts fine and I can see the chan_sip loaded in the list generated from "modules show" |
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| 12:49.05 | pa | is it possible to interface asterisk with a google voice account for calling landlines and mobiles? |
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| 13:31.50 | pa | i found an article about it, but it's dated 2013, and as far as i know google somehow locked down the voice protocol afterward |
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| 14:11.44 | pa | nobody has any idea? |
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| 15:38.07 | ChannelZ | Well they privatized it integrating it into Hangouts, but I'm not sure if they killed the old servers/protocols yet or not. I thought it was slated for earlier this year |
| 15:48.28 | pa | it still works from gmail though |
| 15:48.32 | pa | but it requires the plugin |
| 15:50.32 | ChannelZ | It was the external XMPP interface they were supposedly killing (how Asterisk talked to it) |
| 15:50.42 | pa | ah.. |
| 15:54.57 | Penguin | It still works. I use it every day. |
| 15:55.10 | Penguin | gtalk, jabber, asterisk 1.8 |
| 15:56.56 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:57.16 | ChannelZ | So they just said they were going to kill it to try and get people to stop abusing it, but didn't? |
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| 15:57.38 | TandyUK | hi, is there a way i can get a list of the current active channels, and the codec in use on each? |
| 15:57.48 | Penguin | It was more about support of the service as opposed to the service. |
| 15:58.05 | pa | i found this: http://hobbiesbytwinclouds.wordpress.com/2014/02/07/how-to-make-and-receive-call-using-google-voice-without-xmpp/ |
| 15:58.28 | pa | it seems one can use a DID channel, but dont know what it is. im gonna read now |
| 15:58.31 | Penguin | Oh, you left for a minute... |
| 15:58.37 | Penguin | (1054.53) <Penguin> It still works. I use it every day. |
| 15:58.37 | Penguin | (1055.11) <Penguin> gtalk, jabber, asterisk 1.8 |
| 15:58.44 | pa | ah |
| 15:58.56 | pa | that's great then! i simply need a xmpp channel? |
| 15:59.17 | Penguin | I use res_jabber for that. |
| 15:59.31 | pa | ah i see. and i can call landlines right? |
| 15:59.42 | pa | do i need a specific codec? |
| 16:00.14 | Penguin | ulaw |
| 16:00.50 | pa | thanks! |
| 16:01.04 | pa | i really need it because it seems that now the phone operator changed the prices |
| 16:01.12 | pa | and all calls cost 10cent per minute |
| 16:01.17 | pa | that is crazy |
| 16:01.26 | Penguin | Where are you located? |
| 16:01.32 | pa | in italy |
| 16:01.47 | Penguin | And you're able to use the google voice service from there? |
| 16:02.23 | pa | hm.. to be honest i haven't tried, but i can try now |
| 16:02.34 | pa | is thre some common knowledge it is blocked from italy? |
| 16:02.35 | file | you can only sign up for Google Voice in the US |
| 16:02.37 | WIMPy | AFAIK you can use it from everywhere, yu just can't sign up from outside the US. |
| 16:02.40 | pa | i can use to call italian numbers |
| 16:02.50 | pa | hm |
| 16:02.56 | pa | i am using it from norway tho |
| 16:03.22 | pa | but maybe it's not the same thing you mean, no? |
| 16:03.24 | Penguin | You're in Italy and using the service from Norway? How does that work? |
| 16:03.24 | pa | now |
| 16:03.27 | pa | i start to doubt |
| 16:03.55 | pa | what i mean for google voice is what you get from gmail -> call phone |
| 16:04.16 | pa | well i'min norway but my asterisk is in italy |
| 16:05.37 | Penguin | Okay, then you're using it from Italy. |
| 16:06.09 | pa | well at the moment i just tried it to call from norway to italy, but i would like to use it to call from italy to italy |
| 16:06.12 | pa | iirc it should work |
| 16:06.21 | pa | i used it from there to call the US some time ago |
| 16:06.42 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 16:07.25 | Penguin | I guess the limitation was sign up. I was kind of under the impression that access was limited, but I don't have any evidence to support this thought. |
| 16:09.09 | pa | i'm installing the plugin now, so to try :) |
| 16:10.33 | Penguin | What asterisk branch are you using? |
| 16:12.37 | pa | hm.. ubuntu 14.04 one, i think is 11.7 |
| 16:13.09 | Penguin | I'm using asterisk 1.8, chan_gtalk, and res_jabber. |
| 16:14.12 | pa | is that older than asterisk 1.11? i guess, right.. chan_gtalk is for? |
| 16:14.36 | Penguin | 1.11 isn't a thing, but 1.8 is an older branch than 11. |
| 16:14.37 | pa | btw google voice seems to work from italy too, from the browser |
| 16:15.21 | pa | ok, let's see then |
| 16:15.41 | Penguin | I wouldn't know if chan_gtalk or res_jabber are available in 11. |
| 16:16.44 | pa | Penguin, it seems i have both |
| 16:17.19 | ChannelZ | it's xmpp and motif now |
| 16:17.51 | ChannelZ | res_xmpp and chan_motif that is |
| 16:18.02 | pa | ah |
| 16:18.15 | pa | so chan_gtalk became motif |
| 16:18.21 | Penguin | I also have chan_jingle. I forgot about that. |
| 16:18.38 | *** join/#asterisk blee (~blee@fl-66-86-182-134.dhcp.embarqhsd.net) |
| 16:19.11 | Penguin | It seems that chan_jingle is used for getting incoming calls from google voice into asterisk. |
| 16:19.43 | pa | ah i see.. well this in fact i don't really need |
| 16:19.55 | pa | here one doesn't have to pay to receive calls |
| 16:19.59 | file | chan_gtalk did not become motif, chan_motif was a complete rewrite |
| 16:20.03 | pa | (here= europe) |
| 16:20.17 | Penguin | file: How does chan_jingle relate to those? |
| 16:20.43 | file | chan_jingle implements Jingle, chan_gtalk implements the Google Voice/original Google Talk version |
| 16:20.58 | file | chan_motif implements all 3 current derivatives in one |
| 16:20.59 | *** join/#asterisk zpotoloom (~quassel@tom.data.ee) |
| 16:21.15 | pa | what's jingle? the newest declination of the xmpp google mod? |
| 16:21.19 | Penguin | Oh, interesting. |
| 16:21.23 | file | Jingle is a standard |
| 16:21.39 | file | what Google Voice uses is NOT Jingle |
| 16:22.00 | Penguin | used before? |
| 16:22.16 | Penguin | Why would we need chan_jingle for google voice in the past? |
| 16:22.24 | file | you wouldn't |
| 16:22.44 | file | but it allowed voice calls over XMPP with stuff implementing the standard |
| 16:23.50 | Penguin | It seems like you're making contradictions there. |
| 16:24.03 | file | how so? |
| 16:24.12 | file | there are 3 ways to do voice calls over XMPP |
| 16:24.25 | pa | in motif.conf it mentions jingle in transport=google |
| 16:24.34 | file | Google has 2 versions unique to them - one is the oldest and is used on Google Voice and one is newer, and then there is an actual XEP which is a standard called Jingle |
| 16:25.11 | file | chan_gtalk implemented the older Google one, chan_jingle implemented the XEP - and then I wrote chan_motif to clean up the mess and it supports all 3 |
| 16:26.19 | Penguin | So two to three years ago, I could still have full calling capability without chan_jingle? |
| 16:26.35 | file | it may or may not have worked |
| 16:26.42 | file | the Google Voice protocol still changed/evolved some |
| 16:27.58 | Penguin | So the instruction set which included configuring chan_jingle for google voice calls coming inbound must have been to cover that avenue just in case? |
| 16:28.19 | pa | it's also confusing this nomenclature, google-talk and google-voice. it seems they do the same thing |
| 16:28.28 | pa | (from google side i mean) |
| 16:28.29 | file | I doubt it - there was a time, A LONG TIME AGO, gtalk functionality was in chan_jingle |
| 16:29.20 | pa | file: in res_xmpp is it valid if i set status=invisible? |
| 16:29.21 | Penguin | Nov 18 2011 /etc/asterisk/jingle.conf |
| 16:29.29 | Penguin | The last change to the file was a while ago. |
| 16:29.52 | pa | also, do i have to open 5222 on the firewall? |
| 16:30.01 | pa | in principle i shouldnt have to, right? |
| 16:30.46 | Penguin | If you want a new unrelated connection to be allowed to reach a listening daemon on 5222, you'd need to allow it through the firewall. |
| 16:31.08 | file | invisible is valid, but I have no idea what it'll do within the context of Google |
| 16:31.23 | Penguin | Unless you are using jabberd or some other xmpp server, you don't need to. |
| 16:33.02 | pa | file, also, in motif.conf i at the moment have a gtalk-endpoint and a gvoice. I assume that the gtalk-endpoint is the one i have to use (because gvoice uses google-v1 transport). The question is: the name in connection= has to be the same as the section in xmpp.conf? the default is "gtalk-account" |
| 16:33.17 | Penguin | Did you read the wiki at all? |
| 16:33.27 | pa | i am on that now |
| 16:33.30 | Penguin | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 16:33.36 | pa | but it seems that it's alittle different |
| 16:33.50 | pa | i mean the sections described in the wiki are a little different from what comes with ubuntu |
| 16:34.06 | *** part/#asterisk blee (~blee@fl-66-86-182-134.dhcp.embarqhsd.net) |
| 16:34.19 | pa | for example i dont have a context |
| 16:34.35 | pa | and i have a transport |
| 16:34.39 | Penguin | You need to define your context. |
| 16:34.58 | file | context all the things! |
| 16:35.47 | *** join/#asterisk areski (~areski@91.177.234.207) |
| 16:36.36 | pa | but what is the connection then? |
| 16:36.53 | pa | i mean can i leave it to its default "gtalk-account"? or do i have to change it? |
| 16:36.58 | Penguin | I think if you use connection=google in motif.conf, you need to create section [google] in xmpp.conf. |
| 16:37.10 | pa | ok now i get it :) |
| 16:37.12 | pa | thanks |
| 16:37.25 | Penguin | s/google/gtalk-account/ |
| 16:37.35 | Penguin | Ah, no! |
| 16:37.46 | Penguin | I forgot the g! |
| 16:37.58 | Penguin | Let's try that again... |
| 16:38.00 | Penguin | I think if you use connection=google in motif.conf, you need to create section [google] in xmpp.conf. |
| 16:38.03 | Penguin | s/google/gtalk-account/g |
| 16:38.10 | Penguin | There. Much better. |
| 16:38.52 | pa | can i somehow disable incoming calls? |
| 16:42.21 | Penguin | I can think of a couple possibilities. |
| 16:42.52 | Penguin | In the call routing in the web page, you could select something that isn't google chat or whatever they've renamed it now. |
| 16:43.13 | Penguin | In asterisk, you could create the appropriate extension with a no service message or a simple hangup. |
| 16:44.08 | pa | well but i would like to be able to pick the call up from a browser if im logged in |
| 16:44.16 | pa | i simply want asterisk to ignore incoming calls |
| 16:44.37 | Penguin | Don't answer the call in asterisk, then. |
| 16:46.58 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:221:6aff:feb8:e0b2) |
| 16:48.43 | pa | hm.. no channel type registered for 'Motif' |
| 16:49.07 | Penguin | Did you load chan_motif? |
| 16:49.28 | pa | shouldnt it do by default? |
| 16:50.00 | pa | i can see that rex_xmpp works |
| 16:50.07 | Penguin | Why would it? |
| 16:50.08 | pa | i need to check motif |
| 16:50.39 | pa | Penguin, well, is there a way to make asterisk load the channel automatically then? |
| 16:50.48 | Penguin | After you configure motif.conf, load chan_motif. |
| 16:51.46 | pa | ok i think there's an error |
| 16:51.48 | pa | thats why |
| 16:51.53 | pa | iin motif.conf i mean |
| 16:51.58 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:221:6aff:feb8:e0b2) |
| 16:55.36 | Penguin | When the conf is present, and when you have autoload enabled, it will load automatically. |
| 16:57.49 | pa | great now it works i think :) but i have to write the country code all the time i think |
| 16:58.20 | Penguin | Learn how to create better asterisk extensions to reduce what you have to key in. |
| 16:58.32 | pa | maybe i could detect when there's no 00 or + in front, and automatically add one country code |
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| 17:05.49 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 17:05.53 | [sr] | hi |
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| 17:21.26 | pa | shouldnt this work to mangle numbers without area code? http://pastebin.ubuntu.com/8777089/ |
| 17:22.40 | ChannelZ | ${0039EXTEN} doesn't mean anything |
| 17:23.01 | ChannelZ | I think you want 0039${EXTEN} if you're intending to add 0039 to the front of the dialed exten |
| 17:23.15 | pa | oh i see |
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| 17:24.10 | pa | thanks |
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| 17:31.29 | pa | WIMPy, i think this genious move of the operator makes also disappear my need for outgoing ISDN calls :) |
| 17:31.46 | pa | but it's good that i can still use it for callerid logging |
| 17:32.07 | racho | i'm having a terrible audio delay (~10-15s) when entering a queue. neither the asterisk box is under heavy load nor is the network connection under stress. can anyone provide a clue where to look for the problem? |
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| 17:38.09 | Penguin | racho: You already increased asterisk's CLI verbose level to watch call processing? |
| 17:42.21 | racho | Penguin, yeah i did. everything is great..it goes through the ivr and up to after the Queue command when the the audio just degrades horribly |
| 17:42.47 | Penguin | Are you using the same codec end to end? |
| 17:43.19 | racho | i'm pretty sure I do |
| 17:44.53 | racho | the call originates from another trunk that acts as a 'router' to this one...both of the are restricted only to alaw/ulaw |
| 17:46.41 | racho | as are the the softphones of the agents too |
| 17:47.07 | Penguin | My only good ideas were network or CPU load creating latency, and you said there isn't any load. |
| 17:47.52 | racho | 0.16, 0.08, 0.09 <- current load levels |
| 17:48.52 | racho | i also suspected the network intially..but after tracerouting/pinging it really didn't show anything unusuall |
| 17:49.15 | Penguin | Did you use mtr in udp mode? |
| 17:49.43 | racho | yes i did |
| 17:50.32 | ChannelZ | So when a call goes into a queue, once the call is actually connected to someone there's a 10-15 sec delay in the audio? |
| 17:51.18 | ChannelZ | 10-15 *seconds* is a LOT |
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| 17:53.41 | racho | ChannelZ, yes it is. I can live with 1-2sec but this...I tried calling a static extension on the same server and the audio is basically excellent |
| 17:54.22 | racho | however once am agent takes a call off the queue both parties suffer severe lag/degradetion |
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| 18:05.56 | racho | any idea what else could lead to this? |
| 18:13.37 | ChannelZ | Not offhand |
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