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| 00:25.03 | Katty | PEANUT BUTTER SAMMICHES |
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| 00:45.28 | carrar | YUM |
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| 01:42.32 | lkthomas | guys, I have AVAYA deskphone 9641G, and I am unsure how to connect with asterisk |
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| 02:15.54 | gusto | lol, i just discovered that there is SIMPLE in pidgin, can i register it to asterisk, like every other SIP soft-phone? |
| 02:16.01 | JamKo | msg nickserv identify jamko fuzzywuzzy |
| 02:16.08 | gusto | ??? |
| 02:16.10 | gusto | lol |
| 02:16.41 | gusto | fuzzywuzzy is not a really strong password |
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| 02:24.38 | Penguin | (2124.32) [freenode] -NickServ(NickServ@services.)- Invalid password for jamko. |
| 02:26.01 | gusto | where do you have that one? |
| 02:26.08 | gusto | ah, you tried it? lol |
| 02:26.34 | Penguin | He either changed it before I tried it or that wasn't the real passwd. |
| 02:36.20 | gusto | however |
| 02:36.27 | gusto | any idea what this SIMPLE protocol is? |
| 02:36.34 | gusto | somehow it does not work with asterisk |
| 02:36.49 | Penguin | ~simple |
| 02:36.49 | infobot | simple is probably Session Initiation Protocol (SIP) for Instant Messaging and Presence Leveraging Extensions |
| 02:37.07 | gusto | yes |
| 02:37.36 | gusto | i created an account on my asterisk and tried to connect to it with pidgin |
| 02:37.51 | gusto | however, he either says that he can not resolve the hostname, or he complains about other things |
| 02:38.04 | gusto | but it can also have something to do with IPv6 |
| 02:38.16 | gusto | when he is trying to connect through IPv6 that would explain the problem |
| 02:39.59 | gusto | ehm, no, tcpdump shows that he does not even try |
| 02:40.00 | gusto | lol |
| 02:41.11 | *** part/#asterisk mmourani (~Adium@64.229.213.168) |
| 02:43.54 | gusto | btw. this ssh is freaking efficient |
| 02:44.27 | gusto | it only transfers packets when there is something going on |
| 02:44.33 | gusto | and that is not always a good idea |
| 02:44.41 | Penguin | Are you new to ssh? |
| 02:44.44 | gusto | no |
| 02:44.58 | gusto | i just saw it first time filtered alone on tcpdump |
| 02:45.30 | *** join/#asterisk JamKo (~JamKo@mail.jaguardata.com) |
| 02:45.39 | gusto | maybe there is somewhere an option to ssh to transfer shit, like to asterisk you have the qualify option to hold the connection in case you are connected over NAT or such |
| 02:45.48 | gusto | btw. Penguin where are you from? |
| 02:45.55 | Penguin | USA |
| 02:46.06 | gusto | ok |
| 02:46.14 | gusto | why are you using a swedish irc server? |
| 02:46.25 | gusto | however ... |
| 02:46.30 | Penguin | I couldn't tell you. |
| 02:46.50 | gusto | the point i wanted to make is that ISP's here are switching to DS Lite on DSL |
| 02:46.57 | gusto | is the same also true for USA? |
| 02:47.07 | gusto | in Slovakia they are still IPv4 only |
| 02:47.15 | gusto | so i suppose to USA it will be the same |
| 02:47.53 | gusto | even my provider does DS Lite now, but i still have real IPv4 only connection, because i made my contract a year ago, they do DS Lite only for new customers |
| 02:49.52 | Penguin | gusto: I'm not familiar with DS Lite. We have IPv4 only for now. |
| 02:50.32 | Penguin | I'm not sure if there are any ISPs in the US with native IPv6 right now. I heard something about it a while ago, but I never saw any proof of it being enabled yet. |
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| 03:00.21 | gusto | i had native ipv6 until may 2013 |
| 03:00.26 | gusto | on my DSL |
| 03:00.48 | gusto | maybe you can remember me connecting with a prefix diffrent to what i have now henet |
| 03:01.10 | gusto | then those idiots started DS Lite for new customers and turned off the testing accounts |
| 03:01.18 | Penguin | I don't pay much attention to that sort of thing. |
| 03:01.23 | gusto | well |
| 03:01.54 | gusto | basically DS Lite is that you have carrier nat on IPv4, so you do not have your own IPv4 address any more, but you get native IPv6 in exchange |
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| 03:08.13 | Penguin | With native IPv6 and no IPv4, how do you access the IPv4 internet? |
| 03:08.39 | TriJetScud | ipv4 in ipv6 |
| 03:08.44 | TriJetScud | aka nat64 |
| 03:09.05 | TriJetScud | oops, nat64 != ipv4 in ipv6 |
| 03:09.33 | TriJetScud | but yeah nat64 is the other option with a box having an ipv4 address connected to the outside world |
| 03:10.19 | Penguin | If I want to access the IPv6 internet, I have to set up a tunnel so that I have IPv6 on my devices (or at least to my edge router). |
| 03:11.02 | TriJetScud | you can do the same thing in reverse Penguin |
| 03:11.03 | Penguin | If I don't have IPv6 on my equipment, I can't access the IPv6 internet. So if you don't have IPv4 on your equipment, how do you access the IPv4 internet? |
| 03:11.19 | TriJetScud | you can have an IPv4 in IPv6 tunnel on an equipment that understands IPv4 |
| 03:11.20 | Penguin | Does the ISP provide the gateway between the two? |
| 03:11.27 | TriJetScud | the IPv6 routers don't give a crap |
| 03:11.37 | TriJetScud | it's just IPv6 headers and data is all it cares |
| 03:12.08 | TriJetScud | but if you were to ask me, having an IPv4 tunnel inside IPv6 is silly anyways |
| 03:12.26 | TriJetScud | it's doable but any sane ISP's won't be doing it due to the overhead that an IPv6 header incurs |
| 03:12.47 | TriJetScud | you might as well as perfrom dual stack instead at that point |
| 03:14.23 | gusto | well, NAT64 is not necessairly the case |
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| 03:15.00 | gusto | you can of course do NAT64 on your router then, of course, that would be a good idea, but otherwise you can just double NAT it and use it like you use UTMS 3G |
| 03:15.51 | TriJetScud | gusto, the smart idea at this point would be deploying dual stack |
| 03:16.09 | gusto | however ... NAT64 on your router is double NAT too, so there is no real difference between this and i havent met some ISP's that do NAT64 |
| 03:16.26 | gusto | look, smart guy, i had real dualstack until may '13 until they cut me off |
| 03:17.05 | TriJetScud | are they really that short on IPv4 addresses where you live? |
| 03:17.11 | gusto | no |
| 03:17.31 | gusto | but at least they are doing something, right? |
| 03:17.39 | TriJetScud | hehe |
| 03:17.39 | gusto | better than nothing |
| 03:17.45 | gusto | they should have started 10 years ago |
| 03:17.57 | TriJetScud | still, on all networks I've been deploying, they're going to be dual stacked |
| 03:18.00 | gusto | now its too late for real dualstack |
| 03:18.10 | TriJetScud | why is that? |
| 03:18.16 | gusto | yes? dream on |
| 03:18.43 | gusto | i talked to a guy from juniper last week, of course he wasnt a fan of NAT as well, but ISP's are buying it |
| 03:19.00 | gusto | they used it for 3G networks, now they are going to use it also on DSL |
| 03:19.49 | TriJetScud | 3G networks is understandable |
| 03:19.52 | gusto | like i said, my ISP - MNet - provided me real dualstack until may '13, because then they shut it down and started dualstacking new customers, leaving the old ones with ipv4 only |
| 03:19.53 | TriJetScud | but on DSL networks? |
| 03:20.05 | gusto | well, not only DSL |
| 03:20.11 | gusto | with DOCSIS its even worse |
| 03:20.37 | gusto | Kabeldeutschland, the biggest cable net provider here in germany gives DS Lite only for 2 years now |
| 03:20.55 | gusto | however ... there you have an option, so you can choose a real IPv4 connection if you need it |
| 03:21.44 | gusto | so there they have at least an option, according to the new terms of service, from '13 to my ISP, you do not even have an option |
| 03:21.53 | gusto | so i put my asterisk out |
| 03:22.15 | TriJetScud | they aren't offering native ipv6? |
| 03:22.37 | gusto | its on the internet now and i am going through my NAT router, not noticing any difference, when they port me to DS Lite |
| 03:22.48 | gusto | DS Lite is native ipv6 |
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| 03:49.54 | Penguin | Here's another technology topic I've been trying to figure out... |
| 03:50.47 | Penguin | Bluetooth devices usually have a range of about 30-35 feet (10 meters), right? |
| 04:00.08 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 04:00.25 | Penguin | How are bluetooth motorcycle helmet intercoms reaching 1km or more? |
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| 04:11.03 | gusto | because they use the new bluetooth protocol |
| 04:11.11 | gusto | this bt2.0 or how it is called |
| 04:11.29 | gusto | thats up to 100m indoors ... and that would mean 1km on free sight |
| 04:12.00 | Penguin | Both old and new bt are 2.4 GHz? |
| 04:12.03 | gusto | 20dB shit, like wlan on 2,4 GHz |
| 04:12.10 | gusto | yes |
| 04:12.48 | gusto | a good idea is not to use these devices at all |
| 04:12.55 | gusto | i for example do not have any of this |
| 04:13.07 | gusto | my bluetooth is this old bluetooth and i connect only mices to it |
| 04:13.30 | gusto | even though i am not using 2,4 GHz wlan |
| 04:14.17 | gusto | i have wndr3800 and that is simultan dualband, but i almost never touch that 2,4 GHz it only goes up when my father turns on his TV set, because that connects to 2,4 GHz, but thats it |
| 04:14.27 | gusto | i am using 5 GHz |
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| 04:27.41 | gartral | hey all, I'm trying to set up asterisk 11.5.1 for gtalk, and I keep getting http://paste.ubuntu.com/6243841/ over and over, very rapidly |
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| 04:54.30 | mnathani | whats a good hybrid 2-line physical telephone that can do voip and regular PSTN ? |
| 04:55.28 | jploh | mnathani: 2 units of SPA-3100? |
| 04:55.55 | WIMPy | I've only ever seen one such thing, the IP202. |
| 04:56.08 | WIMPy | Or DECT Bases. |
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| 05:31.00 | gartral | hey all, I'm trying to set up asterisk 11.5.1 for gtalk, and I keep getting http://paste.ubuntu.com/6243841/ over and over, very rapidly |
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| 05:52.35 | nunne | Anyone have experience with Grandstream phones? And why they Pickup sip-channels directly instead of using Dialplan like expected? (configured *9 as pickup) and using dialplan + PICKUPMARK to do pickups, not anything in features.conf. I also have configures notifycid=ignore-context in sip.conf |
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| 06:33.27 | ChannelZ | you're saying features.conf pickup doesn't work? |
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| 07:12.47 | nunne | ChannelZ: no, it probably will. But i don't want to use it. Since it's a multi customer server. I want to use the dialplan to pickup things. |
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| 07:13.08 | nunne | All i'm wondering is if someone knows why my grandstream phone doesn't seem to want to follow the dialplan |
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| 07:15.40 | ChannelZ | well your phone has a dialplan, is it actually passing the desired extension to asterisk? I don't know about grandstreams specifically, but most SIP phones also have a lot of internal 'service codes', perhaps you are bumping into one of those |
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| 07:35.36 | nunne | ChannelZ: I will try updating asterisk. It seems like it will be at fault. Because it's behaving buggy as hell :/ |
| 07:36.15 | ChannelZ | well you didn't really answer my question |
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| 07:36.32 | nunne | ChannelZ: I have only one Pickup cmd in the dialplan. and it's for Pickup(${CUST}-${EXTEN:2}@PICKUPMARK) and it will Pickup a random incomming/outgoing sip-channel o.O |
| 07:36.50 | nunne | ChannelZ: Yeah. I can see in the verbose log.. But it's behaving very-very weird |
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| 07:38.30 | ipalmer | hi all, is it possible to change the uniqueid variable, I want to put a 4 digit number at the front of it |
| 07:39.51 | nunne | ChannelZ: http://pastebin.com/UD14dUx4 |
| 07:40.12 | nunne | the first pickup is as expected. But that call just drops and it pickups a random call for some other customer |
| 07:40.23 | nunne | (dialing out/in.. seems like it picks a random sip-channel |
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| 07:41.31 | nunne | ChannelZ: I'm using 11.2.0, so will try updating and see if it resolves itself |
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| 07:50.37 | ChannelZ | How are you setting PICKUPMARK? |
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| 08:06.19 | nunne | ChannelZ: I have tried alot of ways (one in my ext-context) by using __PICUPMARK. The ext-context dials Local channels with /n sub-sip, sub-mex, sub-cell etc. And I have also tried setting it in there. (Since it shouldn't matter which channel it picks up, as long as it's one of "them". But it actually picks up something randon |
| 08:06.39 | nunne | ChannelZ: But it works flawlessly if I dial *9202 for example, all the time.. Just when I'm using the BLF |
| 08:07.10 | nunne | so I don't think it's something with my PICKUPMARK.. something seems to be weird in the Grandstream phone.. Since it seems to pickup the channel directly for some reason |
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| 08:09.24 | ChannelZ | Well I mean what are you setting PICKUPMARK _to_? it does matter. |
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| 08:11.01 | ChannelZ | Without really seeing the whole dialplain or knowing why you're using it at all it's hard to say. |
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| 08:30.00 | miha- | hi has anyone having asterisk for media in combination with opensips? I need a little help with scenario. tnx |
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| 09:39.13 | nunne | ChannelZ: I set PICKUPMARK to CUSTOMER202 for example |
| 09:39.29 | nunne | but there is nothing wrong with the PICKUPMARK setup.. since it works dialing the pickup-line manually |
| 09:40.14 | nunne | but i have solved it.. grandstream phones doesn't work well with notifycid=ignore-context ... setting it to no will make it use the dialplan for pickup |
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| 09:41.30 | martinfletcher | hey all |
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| 10:28.38 | martinfletcher | i have an avaya ip office 400 connected to an asterisk box via pri |
| 10:29.13 | martinfletcher | when calls are forwarded from the ip office pbx to the asterisk box, only the 1st 4 numbers are given |
| 10:29.30 | martinfletcher | any ideas on what could cause this, as i have spent 3 days looking into it with no luck |
| 10:30.42 | *** join/#asterisk martinfletcher_ (57ed466d@gateway/web/freenode/ip.87.237.70.109) |
| 10:32.44 | martinfletcher | anyone? |
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| 11:02.15 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 11:02.31 | PLMg | hello, anyone know the asterisk fax log location? |
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| 12:11.59 | wdoekes | PLMg: logger.conf, the "fax" level |
| 12:12.17 | wdoekes | you may need to set fax debugging to 1 though |
| 12:13.38 | wdoekes | (either through the *FAX 'd' option, or 'fax set debug on') |
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| 12:20.53 | PLMg | wdoekes what do I need to enter in logger.conf to be able to see fax related messages to a log file? |
| 12:21.16 | PLMg | in logger conf I have onlt 2 lines enable, [genera] and full=> |
| 12:21.29 | PLMg | I didn't see anything related to fax there |
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| 12:37.03 | wdoekes | check your asterisk source dir: configs/logger.conf.sample |
| 12:37.23 | wdoekes | read that to find out how to configure the logging system |
| 12:37.59 | PLMg | k, thx |
| 12:38.44 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 12:40.09 | PLMg | I do not have that file |
| 12:40.40 | PLMg | instead I did read http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Monitoring_id264504.html but I still do not know how to enable fax logging |
| 12:42.00 | kaldemar | PLMg: fax => fax |
| 12:42.30 | PLMg | so I just type that line into logger.conf? |
| 12:42.45 | kaldemar | and reload logger with "logger reload" in CLI. |
| 12:43.03 | PLMg | ok... ty |
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| 13:01.11 | mase76 | hey! i want to use kde-telepathy do do sip calls via my asterisk server. i can register, and the softphone rings on incoming calls. when answering or dialing, asterisk shows no reaction. is there a known problem? the combination is kde-telepathy, telepathy-rakia and sofiasip. |
| 13:03.24 | *** join/#asterisk damage (~damage@2001:470:51d4:500:5954:fcaa:a251:ef71) |
| 13:03.27 | damage | hi |
| 13:05.49 | Katty | CORN DOGS WITH MUSTARD. |
| 13:08.29 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 13:08.33 | damage | well... thats insane :) |
| 13:08.48 | Chainsaw | damage: No, that's Katty's breakfast. |
| 13:09.39 | Katty | or ramblings of a crazy lady. |
| 13:09.45 | Katty | either would be an acceptable response. |
| 13:09.59 | Chainsaw | What?! Katty is not crazy. How dare you suggest such a thing. |
| 13:10.18 | damage | An acceptable response to hi? |
| 13:10.37 | Katty | yes. |
| 13:10.40 | damage | Well, yes, it is acceptable but still insane :) |
| 13:10.47 | Katty | yes. |
| 13:14.43 | PLMg | kaldemar ty, wokred like a charm after defining what should be logged |
| 13:14.44 | *** join/#asterisk serafie (~erin@nat/digium/x-vmzmcqjlrnepvdxz) |
| 13:22.30 | damage | Alice (Subnet B) is starting a call via a Asterisk (11.5.0, Subnet A) to Bob (also Subnet A). When Bob is hanging up, Alice does not receive a BYE message. |
| 13:22.43 | damage | But if I put Alice into Subnet A everything works fine |
| 13:22.53 | damage | Any ideas about further debugging? |
| 13:24.30 | kaldemar | PLMg: np. |
| 13:24.37 | kaldemar | damage: sip debug |
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| 13:46.27 | damage | kaldemar: I can't see any specials: http://pastebin.com/Xhr4iisJ (this is a debug of the no BYE behavoiur) |
| 13:47.06 | damage | cip7965 is calling dect (via *3328). dect is answering and hanging up after 5 seconds |
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| 14:07.19 | mdalius | Hi |
| 14:07.27 | mdalius | I have registered issue |
| 14:07.28 | mdalius | https://issues.asterisk.org/jira/browse/ASTERISK-22686 |
| 14:07.37 | mdalius | could this be configuration issue? |
| 14:32.22 | *** join/#asterisk bchia (~Adium@nat/digium/x-gayecwgufcibuchj) |
| 14:36.25 | gartral | hey all, I'm trying to set up asterisk 11.5.1 for gtalk, and I keep getting http://paste.ubuntu.com/6243841/ over and over, very rapidly, any ideas why? |
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| 14:56.58 | lnb_ | What does an ITSP have to do in code to provide its customers with support for t.38 ? |
| 14:57.06 | lnb_ | for asterisk 10 |
| 14:57.13 | asteriskmonkey1 | turn it on |
| 14:57.16 | asteriskmonkey1 | in there switch |
| 14:57.17 | asteriskmonkey1 | :/ |
| 14:57.37 | lnb_ | do you happen to know the dialplan for it? |
| 14:57.52 | asteriskmonkey1 | its not dialplan, its a sip setting |
| 14:58.39 | lnb_ | would it be the same as |
| 14:58.40 | lnb_ | [general] |
| 14:58.40 | lnb_ |
|
| 14:58.40 | lnb_ | t38pt_udptl=yes,redundancy |
| 14:59.10 | lnb_ | in sip.conf |
| 14:59.59 | asteriskmonkey1 | yes thats what you need on |
| 15:00.05 | asteriskmonkey1 | yes is sufficient |
| 15:00.42 | asteriskmonkey1 | there are other switches and modes |
| 15:00.57 | asteriskmonkey1 | ie faxdetect can be set for t38 only etc.. |
| 15:01.22 | lnb_ | do you happen to know an URL i can get the details/configuration at? |
| 15:01.39 | asteriskmonkey1 | its in the config files when you do make samples |
| 15:03.10 | lnb_ | my problem is i am a client of an itsp that does not yet support t.38. I am trying to help them out so I can fax out. My other ITSP is in the USA and when we fax in Canada, a lot of toll-free fax DIDs do not accept USA origin calls |
| 15:03.23 | moy | saxa: old discussion, but what are you trying to do with R2? and where? |
| 15:03.29 | asteriskmonkey1 | that is nothing to do with t38 |
| 15:03.40 | lnb_ | i realize that |
| 15:03.57 | lnb_ | but t.38 makes faxing reliable over g711 |
| 15:04.14 | asteriskmonkey1 | only between t38 points |
| 15:05.02 | asteriskmonkey1 | you need your carrier to get proper gear |
| 15:05.05 | lnb_ | yesterday I happen to send a fax through this itsp, and had udptl debug on and saw udptl packets going out. Since then it doesn't happen so I figured its some configuration setting(s) to make it work all the time |
| 15:05.11 | lnb_ | they have good gear |
| 15:05.18 | lnb_ | just a smaller operation |
| 15:05.25 | asteriskmonkey1 | then they need no aid turning on t38 :/ |
| 15:05.26 | lnb_ | but a clec nonetheless |
| 15:05.41 | lnb_ | they do, since its not working all the time |
| 15:05.49 | asteriskmonkey1 | its probably there upstreams then |
| 15:06.15 | asteriskmonkey1 | get dumps of there gear and read it out |
| 15:06.18 | lnb_ | not sure. they told me they did not configure anything for t.38 gateway |
| 15:06.38 | asteriskmonkey1 | well im sure the manufacture of there hardware can support this, this is not an asterisk thing |
| 15:07.06 | lnb_ | now you're saying its not asterisk? |
| 15:07.26 | lnb_ | i thought from above you said it has to be configured in asterisk |
| 15:07.31 | asteriskmonkey1 | yes it does |
| 15:07.44 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:07.44 | asteriskmonkey1 | if you have it configured and its in the right mode should work 100% of time |
| 15:07.51 | lnb_ | ok, so you mean for the connection between us and them |
| 15:07.57 | lnb_ | you still need from them to pstn |
| 15:08.17 | asteriskmonkey1 | pstn is pstn |
| 15:08.23 | lnb_ | right mode? |
| 15:08.35 | asteriskmonkey1 | passthru gateway etc.. |
| 15:08.38 | gartral | hey all, I'm trying to set up asterisk 11.5.1 for gtalk, and I keep getting http://paste.ubuntu.com/6243841/ over and over, very rapidly, any ideas why? |
| 15:08.39 | asteriskmonkey1 | nm you need to go read |
| 15:08.56 | *** join/#asterisk jpcansa (~jpcansa@201.199.100.178) |
| 15:09.34 | file | gartral, Google has probably locked out your account - you need to go find where to tell them that it was you attempting to use it |
| 15:09.44 | file | or add an app level password |
| 15:12.14 | lnb_ | asteriskmonkey1: thats what i asked you.. a good URL so i can read :) |
| 15:13.26 | gartral | file: so those errors aren't asterisk's configs, it's googles security? |
| 15:13.41 | file | so far 100% of the time it has been |
| 15:13.49 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:13.50 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 15:13.56 | asteriskmonkey1 | lnb_: build samples read samples, of voip info |
| 15:14.15 | lnb_ | ok |
| 15:14.18 | lnb_ | thanks. |
| 15:14.41 | lnb_ | i think i will setup another vm and install asterisk and make the samples |
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| 15:21.35 | gartral | file: by application level, you mean the seperate password that is used when 2-factor authentication is on? |
| 15:21.56 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 15:23.59 | file | I haven't use it, so I don't know |
| 15:25.47 | gartral | file: yea, neither do i, and I can't find anywhere to allow a specific request |
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| 15:33.52 | pmantis | Hi everyone. Is there an Asterisk version that directly supports beep tones during call recording? |
| 15:33.58 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:35.00 | asteriskmonkey1 | <PROTECTED> |
| 15:35.04 | asteriskmonkey1 | youll hear key presses |
| 15:35.06 | asteriskmonkey1 | :/ |
| 15:35.14 | asteriskmonkey1 | of you mean insert beeps/messages into your stream? |
| 15:35.23 | pmantis | LOL ? |
| 15:35.47 | WIMPy | assumes he wants to meet legal requirements. |
| 15:36.06 | asteriskmonkey1 | legal requirements is stating the call is being recorded prior to recording |
| 15:36.07 | pmantis | Yes, I mean adding a goverment mandated beep every 15 seconds during a call when recording is in progress. |
| 15:36.19 | asteriskmonkey1 | oh thats a wierd one |
| 15:36.23 | asteriskmonkey1 | what government is that |
| 15:36.27 | pmantis | California *requires* the beep. |
| 15:37.07 | asteriskmonkey1 | oh ok |
| 15:37.10 | pmantis | It's pretty common, and I *thought* I found a version of mixmonitor that did this... but now I can't find it. |
| 15:37.15 | asteriskmonkey1 | well you can do that with agi |
| 15:37.20 | asteriskmonkey1 | $agim->send_request('PlayDTMF',array('Channel'=>$chan,'Digit'=>"1")); |
| 15:38.17 | pmantis | Interesting idea. What about playing a wav file of a beep, so it sounds more like the recording tone people expect? |
| 15:38.28 | asteriskmonkey1 | sure |
| 15:38.31 | asteriskmonkey1 | you should be able to do that |
| 15:38.47 | asteriskmonkey1 | using Playback instead |
| 15:38.48 | asteriskmonkey1 | :) |
| 15:39.15 | pmantis | I could chanbarge, dump to a context that plays a beep every 15 seconds. |
| 15:39.30 | asteriskmonkey1 | Sure probably lots of interesting ways of doing it |
| 15:39.30 | pmantis | Playback will work in an already bridged call? |
| 15:39.45 | WIMPy | IIRC that didn't work. |
| 15:39.46 | asteriskmonkey1 | youd have to try it |
| 15:39.58 | asteriskmonkey1 | I know most people use freeswitch for things like this |
| 15:40.09 | asteriskmonkey1 | as it supports stuff like that more easily |
| 15:40.52 | asteriskmonkey1 | hang on looking at switches |
| 15:40.54 | pmantis | Isnt't that a fork? |
| 15:41.05 | asteriskmonkey1 | no freeswitch isnt a fork |
| 15:41.22 | pmantis | Must be another project I was thinking ofo. |
| 15:41.24 | asteriskmonkey1 | its a differnte engine entirley.. |
| 15:41.59 | asteriskmonkey1 | bunch of devs with different views the 1.0.x days buggered of and start fs |
| 15:42.17 | asteriskmonkey1 | asterisk used to be one large core process where fs wasnt |
| 15:42.29 | asteriskmonkey1 | now asterisk in a sense is like fs :/ hhaha lol |
| 15:42.34 | asteriskmonkey1 | i like asterisk better |
| 15:42.51 | pmantis | Yeah, I started using asterisk around v 0.9x |
| 15:44.43 | asteriskmonkey1 | well gl with your recording voodoo magic |
| 15:44.45 | asteriskmonkey1 | :) |
| 15:44.48 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 15:45.30 | [TK]D-Fender | asteriskmonkey1: was around 1.2/1.4 to my memory... |
| 15:45.38 | *** join/#asterisk AL13N_work (~alien@91.183.52.232) |
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| 15:45.39 | [TK]D-Fender | (for FS start) |
| 15:45.46 | pmantis | Heh, thanks.. just wish it was a little easier. Any other ideas? |
| 15:45.55 | asteriskmonkey1 | [TK]D-Fender: no it was no later than 1.2 |
| 15:46.16 | asteriskmonkey1 | it was starting in the 1.1x era i remember i was there lol |
| 15:46.40 | AL13N_work | is it possible to have automatic direct dial to extension working with a number like +3216834756#401 and sort of pick up that the number was sent like that and directly forward to ext 401? |
| 15:46.41 | asteriskmonkey1 | The big complaint was all the modules where tied to core and massive thread locking issues |
| 15:47.01 | *** part/#asterisk keycruncher (~Dennis_Li@c-174-55-112-94.hsd1.pa.comcast.net) |
| 15:47.02 | asteriskmonkey1 | AL13N_work: core show application dial |
| 15:47.11 | asteriskmonkey1 | there is a switch for sending dtmf after connect |
| 15:47.18 | AL13N_work | ah |
| 15:47.28 | AL13N_work | how do i pick that up in the code? |
| 15:48.09 | pmantis | Are you talking about an inbound call or outbound? |
| 15:48.45 | WIMPy | didn't understand the question at all. |
| 15:49.07 | [TK]D-Fender | AL13N_work: It's your dialplan. parse out the # in there to get the 401 out of it to pass. |
| 15:49.19 | asteriskmonkey1 | AL13N_work: if your using it for inbound call direction, just use sip headers man make your life easy |
| 15:49.28 | asteriskmonkey1 | or use CUT etc.. |
| 15:51.18 | AL13N_work | [TK]D-Fender: i don't really understand out of what variable i need to parse the # part |
| 15:51.37 | WIMPy | EXTEN |
| 15:52.05 | pmantis | Again, it helps to know where the call is coming FROM, and where TO? |
| 15:53.03 | AL13N_work | from outside to a local extension |
| 15:53.33 | [TK]D-Fender | AL13N_work: you haven't clarified when you are getting this number. |
| 15:53.50 | [TK]D-Fender | Alis this a DIALED number directly from a device to your server? |
| 15:53.59 | pmantis | Most telcos can't pass a #, but it it does, you use ${EXTEN} and split it on #. |
| 15:54.06 | [TK]D-Fender | AL13N_work: Or is this further on in some dialplan processing in some sort of IVR / Read? |
| 15:54.40 | AL13N_work | my dial plan is like Answer();Dial( all extensions); Hangup(); someone from outside rings the number with #401 appended |
| 15:54.46 | AL13N_work | ah, ${EXTEN} |
| 15:54.47 | AL13N_work | i'll try |
| 15:56.11 | AL13N_work | crap, the provider doesn't pass # suffixes |
| 15:56.23 | AL13N_work | pmantis: is there another way to accomplish this? |
| 15:56.36 | AL13N_work | or do i have to ask sip provider to turn this stuff on? |
| 15:56.42 | WIMPy | How are you getting the call? |
| 15:56.52 | AL13N_work | sip provider passes it to us |
| 15:57.22 | pmantis | If it's SIP end-to-end, you can use a sip header, or on the sending PBX, waitf or answer and auto-dial DTMF. Not any other way, afaik. |
| 15:58.00 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:58.00 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:58.28 | WIMPy | You'd need sn ITSP that gives you DID. But I dont know if they exist. |
| 15:59.33 | AL13N_work | pmantis: i obviously don't have control over the sip provider, what is the chance that they are already parsing it out and putting it into some header i will need to fetch? |
| 15:59.59 | WIMPy | 0 |
| 16:00.12 | pmantis | It's possible. Use sip set debug on, then watch the traffic as a call comes in. |
| 16:00.56 | asteriskmonkey1 | AL13N_work: if the stuff before the # which dosnt exsist is all same length you can do length based chopping and routing |
| 16:01.16 | WIMPy | Usually you can't get anything other than numeric digits across anyway. |
| 16:02.41 | rrittgarn | you could always have your callers call +3216834756,,401 and have your PBX grab the 401 as an extension... its replacing the # with pauses but it works on most phones. |
| 16:03.01 | WIMPy | And unless you have DID, appended digits will either be discarded or might even make the call fail. |
| 16:04.59 | AL13N_work | ok, nothing in EXTEN and nothing in the headers :-( |
| 16:05.14 | AL13N_work | just appending digits makes the call fail |
| 16:06.01 | AL13N_work | rrittgarn: is that a comma? |
| 16:06.06 | AL13N_work | how do you press this? |
| 16:06.43 | rrittgarn | yes just a comma |
| 16:06.49 | rrittgarn | depends on the phone |
| 16:06.53 | rrittgarn | usually its for saved numbers |
| 16:06.57 | vandyk | [TK]D-Fender: do you remember last week we talked about a lot of canceled calls in my server? You told me to change the indications.conf file and see how it goes. I did this and I'm still having a lot of calls being canceled |
| 16:06.57 | rrittgarn | its insert a pause |
| 16:08.20 | [TK]D-Fender | AL13N_work: Your description is still vauge. SHOW US the call. |
| 16:08.22 | [TK]D-Fender | ~pb |
| 16:08.22 | 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. |
| 16:08.24 | [TK]D-Fender | ^^^ |
| 16:08.42 | [TK]D-Fender | vandyk: Yeah, I was just about out of inspiration at that point... |
| 16:11.30 | gartral | ugh getting this working is giving me a headache x.x |
| 16:12.34 | Greenlight | Say I wanted to connect 32 ISDN30 channels into Asterisk, how would I go about it ? |
| 16:12.53 | Greenlight | s/channels/bearers |
| 16:13.06 | WIMPy | 32 what? |
| 16:13.09 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 16:13.21 | Greenlight | 32 ISDN30 trunks |
| 16:13.23 | [TK]D-Fender | Greenlight: Get a nice huger interface for them. |
| 16:13.28 | WIMPy | 32 PRIs (i.e. 960 channels)? |
| 16:13.32 | Greenlight | Yea |
| 16:13.54 | WIMPy | With 4 octo-PRI cards. |
| 16:14.01 | [TK]D-Fender | Greenlight: Something like an AudioCodes Mediant, or Cisco DS3 interface that can push them out as SIP, etc |
| 16:14.04 | Katty | CHICKEN BISCUITS. |
| 16:14.11 | WIMPy | Unless the Sangoma E3 card can be used for that now. |
| 16:14.20 | Katty | hugs Qwell |
| 16:14.26 | [TK]D-Fender | WIMPy: No channelized E3 yet |
| 16:14.43 | Greenlight | So, I'd not hit issues with 4x cards in one box ? |
| 16:14.47 | WIMPy | Bad luck. |
| 16:15.52 | WIMPy | Are you sure you want them all on one box? |
| 16:16.16 | Greenlight | Well, separate boxes would work too. Depending on cost. |
| 16:16.33 | Greenlight | Basically I want to sit as the middleman to offer call recording of all calls made over the ISDNs |
| 16:16.45 | WIMPy | It's a lot of outage you produce if that box goes down. |
| 16:17.17 | Greenlight | Yea, I see your point. I suppose it makse more sense to have separate boxes |
| 16:17.34 | pmantis | 32 boxes. lol |
| 16:17.52 | WIMPy | 16 max |
| 16:17.54 | Greenlight | Max would be 16, since I have a network side and a user side |
| 16:18.16 | Greenlight | BUt I'd say 4 or 8 might work well |
| 16:18.17 | pmantis | OH, so 16 PRIs, yu're just getting in the middle. |
| 16:18.27 | Greenlight | Yea, so I'd have 16 IN and 16 OUT |
| 16:18.56 | Greenlight | 4 boxes with OctoPRI cards perhaps |
| 16:18.59 | WIMPy | Is it only recording? |
| 16:19.10 | Greenlight | Yea, that's the request at the moment. |
| 16:19.35 | WIMPy | Then you don;t have to be in the middle, but could do passive sniffing. |
| 16:19.53 | Greenlight | Oh, I didn't realise that could be done on ISDN |
| 16:19.58 | pmantis | On PRIs? Interesting! |
| 16:20.06 | WIMPy | That way wouldn't produce anything more than missing recordings if the box goes down. |
| 16:20.30 | WIMPy | IIRC there was a patch for dahdi to do that. |
| 16:20.39 | Greenlight | And how would Asterisk see the calls ? |
| 16:20.57 | WIMPy | It still gets all the signalling. |
| 16:21.30 | Greenlight | Hmm interesting |
| 16:21.36 | Greenlight | goes to Google it |
| 16:22.49 | WIMPy | There are both specialised hadware and software for that purpose out there, |
| 16:22.54 | WIMPy | . |
| 16:23.56 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 16:24.02 | Greenlight | Yea, was trying to stick with what I know |
| 16:24.08 | wasanzy | hello |
| 16:24.14 | Greenlight | And thus what I can charge for ^^ |
| 16:24.37 | wasanzy | pls is there a bug in asterisk version 11.5.1? |
| 16:24.52 | Greenlight | wasanzy: Yes, I suspect there are. |
| 16:25.19 | wasanzy | when I start asterisk, is dieing quickly and I cant do asterisk -r as well |
| 16:25.29 | saxa | moy: hi, I want to connect the E1 line in Brasil to my asterisk box |
| 16:25.35 | Greenlight | I hightly doubt that's a bug. |
| 16:25.39 | *** part/#asterisk pmantis (~sswitzer@cpe-74-65-7-79.rochester.res.rr.com) |
| 16:25.52 | Greenlight | Start asterisk with -cvvvv and watch for the error |
| 16:26.00 | wasanzy | ok |
| 16:29.05 | moy | saxa: working already or still having issues? |
| 16:29.18 | WIMPy | saxa: Then you should find out how you want to do it |
| 16:29.41 | wasanzy | I saw this but no file specified: Failed to load configuration file |
| 16:29.42 | drmessano | There's lots of bugs in Asterisk 11.5.1 |
| 16:30.04 | drmessano | There were in previous versions, and there will be in future versions |
| 16:30.19 | wasanzy | pls I want to revert to version 11.5.0, how do I uninstall the 11.5.1? |
| 16:33.25 | drmessano | [12:29:41] <wasanzy> I saw this but no file specified: Failed to load configuration file <-- Look at the lines before it. Usually thats the "I give up" message |
| 16:34.36 | wasanzy | It actually says at the end " Asterisk Ready." and I didn't see any I give up message too |
| 16:35.17 | asteriskmonkey1 | wasanzy: you install by source or pkg? |
| 16:35.39 | wasanzy | asteriskmonkey1: source |
| 16:35.44 | asteriskmonkey1 | also if you do asterisk -vvvvvvc |
| 16:35.49 | asteriskmonkey1 | you should see where its dieing |
| 16:36.01 | wasanzy | ok |
| 16:36.31 | asteriskmonkey1 | if you want to go back a version you can just do a make deinstall in the 11.5.1 src folder and hop back into your 11.5.0 folder and do a make install |
| 16:37.03 | wasanzy | ah great |
| 16:37.05 | asteriskmonkey1 | you may find though you have some lingering module its breaking on, which is easy enough to comment out as a no load |
| 16:37.06 | asteriskmonkey1 | :) |
| 16:37.26 | karl-s | since when is deinstall a make option??? |
| 16:37.38 | karl-s | i'm gonna go try that out |
| 16:37.50 | wasanzy | am not loading any module, I am doing autoload |
| 16:38.06 | Penguin | If the person who wrote the code included a deinstall target, it is an "option" since the time they wrote it. |
| 16:38.43 | Penguin | If you're talking about FreeBSD, since as far back as I can remember. |
| 16:38.44 | wasanzy | does it mean version 11.5.0 is more stable than 11.5.1? |
| 16:39.14 | Penguin | Stable means what? |
| 16:39.19 | Penguin | (to you) |
| 16:39.20 | karl-s | we'll i mean, without having to look at Makefile, I wish all of the make options were documented in README |
| 16:39.20 | asteriskmonkey1 | start asterisk with -vvvvvvvvvvvvc |
| 16:39.39 | asteriskmonkey1 | there is no such thing as stable, only a version where a bug hasnt been found that horrible kills it lol |
| 16:40.03 | Penguin | To me, stable means it runs for months or years as opposed to only days. |
| 16:40.42 | Penguin | To me, stable has nothing to do with how goofy something may be created, but how reliable it is. |
| 16:40.45 | wasanzy | Penguin: to me, no bugs also counts, at least not much of a bug |
| 16:40.58 | asteriskmonkey1 | well start asterisk like a said |
| 16:41.05 | asteriskmonkey1 | you should see whats causing it grief |
| 16:41.14 | asteriskmonkey1 | it will break / crash where there is module issue |
| 16:41.15 | wasanzy | asteriskmonkey1: ok |
| 16:41.24 | Penguin | 11.5.0 is going to have bugs in it that have been fixed in 11.5.1 |
| 16:41.35 | *** join/#asterisk bchia (~Adium@nat/digium/x-zuqdqcderczuzkuo) |
| 16:41.48 | wasanzy | Penguin: ok |
| 16:42.05 | Penguin | And that is documented in the changelog. |
| 16:43.29 | vandyk | any other idea to discover why calls are being canceled on Asterisk 11.5.1 (FreePBX distro)? |
| 16:43.46 | Penguin | ~freepbx |
| 16:43.47 | 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 |
| 16:44.29 | [TK]D-Fender | Penguin: It isn't a FreePBX issue from what I can see. * is cancelling on a SIP progress message following a dial |
| 16:44.54 | wasanzy | asteriskmonkey1: Some unable to load configuration files (modules not activated) |
| 16:46.41 | asteriskmonkey1 | more specific? |
| 16:47.03 | asteriskmonkey1 | is asterisk running as a user that dosnt have perms to the config file folder? |
| 16:47.17 | asteriskmonkey1 | check that stuff |
| 16:47.23 | asteriskmonkey1 | heads out for lunch |
| 16:49.09 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 16:53.58 | saxa | moy: no I will get a new line, which they call Digitronco, and I find out this is a E1 line. |
| 16:54.17 | saxa | moy: they use R2D protocol over it, if i understand well |
| 16:54.42 | saxa | WIMPy: basically i want to maintain the phones i have already up now, they are sip extensions |
| 16:55.08 | saxa | WIMPy: now I need to find which card i have to put into my box to get * working as it works now |
| 16:55.49 | anonymouz666 | saxa: yeah We do call digitronco |
| 16:56.13 | anonymouz666 | R2D all the way, unfortunately |
| 16:56.53 | coppice | digitronco sounds like something advertised endlessly on TV :-) |
| 16:57.41 | saxa | anonymouz666: you have any suggestion of a card which I should go with ? |
| 16:58.02 | saxa | coppice: i found out its an commercial name used in Brasil for E1 lines. |
| 16:58.25 | anonymouz666 | coppice: two co-workers went to hong kong in hauwei for 2 weeks, they took nice pictures from the country and city |
| 16:58.50 | anonymouz666 | saxa: bingo |
| 16:59.07 | wasanzy | I commented out the runusers and group in the conf, so automatically, it is running as the root user |
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| 16:59.54 | saxa | anonymouz666: yep, i read up one Oi pdf i found online explaining how to sell digitronco |
| 17:00.23 | coppice | anonymouz666: we have some beautiful countryside in HK |
| 17:00.55 | saxa | anyway i would like to know which way would be the best for me , to maintain what I have now, and switch from a TDM410P to another card to connect my sip phones and extensions to this E1 line I will get ? |
| 17:06.56 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 17:07.54 | WIMPy | saxa: Ah, that interesting story. |
| 17:08.27 | saxa | WIMPy: yeah |
| 17:08.44 | saxa | WIMPy: always the same thing :) |
| 17:10.25 | WIMPy | saxa: Greetings, |
| 17:10.52 | WIMPy | Ooop. The best way would be to change that line to a E1 PRI. |
| 17:11.27 | coppice | do you think people would use MFC/R2 if PRI was an option? |
| 17:11.57 | WIMPy | Brazil is not the US. PRI should be an option. |
| 17:12.24 | coppice | really? ask the Brazilians |
| 17:13.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:14.37 | coppice | its amazing how much MFC/R2 there still is in many countries, but in South America it is the *only* choice for a huge number of people |
| 17:15.16 | WIMPy | Still better than FXO. |
| 17:16.04 | coppice | if you only want to carry voice there's nothing really bad about it. its just so antique |
| 17:18.39 | anonymouz666 | coppice: correct, in most cases it's the only option |
| 17:19.36 | tm1000 | test |
| 17:19.54 | anonymouz666 | it is antique and slow... and the speed becomes more evident when you connected through legacy pbx's. |
| 17:20.01 | coppice | you'd be amazed where iits still used, even places like Australia have numerous Asterisk users with MFC/R2 |
| 17:21.01 | coppice | the only real speed issue is if your E1 is for something like an IVR, with a huge number of short calls. Otherwise the overhead for the slow signalling is really not that great |
| 17:22.12 | coppice | anonymouz66: is everyone still using locally produced cards, because of the crazy taxation? |
| 17:22.14 | anonymouz666 | in our case we have the delay from legacy pbx to asterisk (first leg of call) in mfc/r2, and then asterisk to telco using mfc/r2 (second leg). |
| 17:22.45 | anonymouz666 | we try to integrate with ISDN PRI when legacy supports and talk mfc/r2 with the telco |
| 17:23.04 | WIMPy | DO YOU DO STORE-AND-FORWARD? |
| 17:23.08 | WIMPy | oops |
| 17:23.15 | *** join/#asterisk danbell77 (~danbell77@host90-152-0-35.ipv4.regusnet.com) |
| 17:24.48 | anonymouz666 | coppice: people are using locally produced cards for a number reasons. I can mention support, partnership... speaking about technical stuff, some companies makes your own channel driver and bypass some chan_dahdi/dahdi limitations |
| 17:25.34 | anonymouz666 | another important thing: they deliver much faster |
| 17:26.18 | WIMPy | Do you know any channels I don;t know, yet? |
| 17:26.50 | coppice | anonymouz666: do huawei have ways around the crazy import duty? there are a lot of Chinese companies buying into all sorts of industries in Brazil |
| 17:28.06 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:30.47 | anonymouz666 | I really don't know what they do, but they are present and very strong in public projects bidding |
| 17:31.09 | *** join/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |
| 17:31.33 | anonymouz666 | They are fighting with CISCO in Wireless stuff, but now the focus is eLTE. |
| 17:32.26 | anonymouz666 | WIMPy: channel driver you mean? |
| 17:32.37 | WIMPy | yes |
| 17:34.09 | anonymouz666 | WIMPy: I don't know what you know, but "chan_khomp" is a channel driver made by a local company here. Used by KHOMP boards. |
| 17:34.23 | coppice | anonymouz666: huawei are doing really well in the LTE market, although the network I use doesn't use huawei |
| 17:35.16 | coppice | anonymouz666: there is another brazilian company with its own cards and drivers, isn't there? |
| 17:35.30 | anonymouz666 | digivoice |
| 17:35.46 | anonymouz666 | and another called "aligera" that uses chan_dahdi. |
| 17:36.46 | *** join/#asterisk ttyp123 (~ttyp1@59.90.163.87) |
| 17:37.09 | WIMPy | Yes, they are new to me. |
| 17:37.18 | anonymouz666 | KHOMP claim to have a better FXO call progress than DAHDI, AMD in pre-connect stuff for callcenters, and things like that. may ou may not make a difference in your setup. |
| 17:37.20 | WIMPy | And I already found quite a few. |
| 17:38.13 | coppice | pretty much anything would have better call progress handling than dahdi. its very crude |
| 17:39.25 | anonymouz666 | coppice: we are very close to people from hauwei. they're getting better in wireless stuff. Working hard to improve their firmware to make the radios more robust with a bigger capacity |
| 17:40.29 | coppice | a lot of other LTE equipment is just huawei rebadged. |
| 17:40.39 | coppice | a lot of 3G was too |
| 17:43.01 | saxa | WIMPy: i have now FXO |
| 17:43.14 | saxa | WIMPy: yes I asked for ISDN, but its not available |
| 17:48.33 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:48.55 | anonymouz666 | saxa: if you explain what you are trying to do from the beggining maybe I can help you |
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| 17:56.38 | WIMPy | He just wanted to know what card to get for R2. |
| 17:58.36 | anonymouz666 | Digium, Sangoma... should be fine. |
| 17:58.51 | anonymouz666 | the R2 signalling support is native beggining from version 1.8. |
| 17:59.40 | WIMPy | I think even LCR can do it. |
| 17:59.57 | anonymouz666 | 1.2 was UNICALL times. 1.4 was moy's patches for that version. Ohh that support starts from version 1.6.2, I think |
| 18:00.06 | anonymouz666 | started |
| 18:01.57 | WIMPy | khomp do only fxs/fxo and gsm, no wired digital? |
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| 18:04.17 | boom^time | Hey guys, regarding cdrs. If I call a number with asterisk which bridges them to an IVR where hitting 0 dials a different number and then bridges them I should assume I'm going to incur charges per minute for both calls right? Yet asterisk only generates one cdr including the total duration of the original call |
| 18:04.36 | danbell77 | Has anyone here done much with Cisco 7970's on Asterisk? |
| 18:05.37 | danbell77 | In particular I'm looking to use the buttons like they run on a CallManager Box (i.e. the callforward button says the number it is forwarding to and then becomes a cancel button). That type of thing. |
| 18:05.45 | WIMPy | boom^time: CDRs are known to be pretty useless. You could try to use CEL or something in your dialplan. |
| 18:05.55 | anonymouz666 | WIMPy: no, there are cards and the ebs-e1-spx series. |
| 18:06.55 | WIMPy | Seems hard to find any information in english at all. |
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| 18:11.22 | smkelly | file: hi |
| 18:11.42 | boom^time | WIMPy, thanks, is CEL the standard or are there good alternatives? |
| 18:12.26 | WIMPy | It's one way to gather the data. |
| 18:13.30 | file | smkelly, hi |
| 18:13.47 | smkelly | file: nice hat. |
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| 18:18.41 | moy | anonymouz666: yes native R2 support starts on 1.6.2 ... 1.4 works with patches or with chan_unicall |
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| 18:38.22 | mjordan | boom^time: your other option is ForkCDR, ending the original CDR to create a new one. |
| 18:38.57 | mjordan | that's actually why it exists: stop the original "billing record" - which may or may not be something you want to bill for - and start a new record |
| 18:39.19 | mjordan | or, yes, use CEL and roll your own billing system |
| 18:41.45 | boom^time | mjordan, Thanks! I was just in the middle of reading about ForkCDR when I looked over and saw your comment. I'm trying to learn what options are appropriate. |
| 18:41.49 | Penguin | danbell77: Look at chan-sccp-b. |
| 18:42.09 | mjordan | the options are confusing for it :-) |
| 18:42.14 | mjordan | looks it up |
| 18:42.40 | vandyk | WIMPy: regarding wired digital, Khomp does not have that. I've used Digium and Openvox cards for years, now I moved to E1 gateways |
| 18:42.45 | mjordan | what version of Asterisk are you using? |
| 18:42.50 | boom^time | 11.5 |
| 18:42.52 | mjordan | kk |
| 18:43.01 | mjordan | Definitely use the 'e' option |
| 18:43.05 | vandyk | you can look for Suncomm equipment |
| 18:43.17 | mjordan | If what you want is to end the current CDR and create a new one, then you need that |
| 18:43.29 | boom^time | right otherwise I'll get overlap |
| 18:43.47 | mjordan | You may also want the 'v' option to copy over variables |
| 18:43.50 | WIMPy | strongly dislikes gateways. Just much less functionality. |
| 18:44.07 | vandyk | the advantage is that you can have your servers virtualized |
| 18:44.08 | mjordan | If you want the answer time set - so that billsec starts incrementing - you'll need the 'a' option |
| 18:44.18 | mjordan | The rest are going to depend heavily on weird conditions that are hard to predict |
| 18:44.35 | boom^time | like A? |
| 18:44.41 | mjordan | correct. |
| 18:44.58 | vandyk | what kind of functionality that you can't find in gateway and you have on cards? |
| 18:45.16 | WIMPy | A lot. |
| 18:45.22 | mjordan | The options on ForkCDR expose a lot of weird internal implementation details that you won't have to bother with unless things don't look "right" |
| 18:45.27 | mjordan | so unless you need em, I'd leave them alone |
| 18:45.37 | boom^time | The application is an outbound call to a number, they answer, if they press 0 it runs a Dial on a different number. I'd need to fork before the Dial to know that I'm billing for two call legs after |
| 18:45.41 | mjordan | But if your disposition doesn't look right, or you aren't getting variables, or other funky things happen, then just ignore them |
| 18:45.47 | boom^time | gotcha |
| 18:45.52 | mjordan | yup |
| 18:46.31 | boom^time | Thanks for your help I really appreciate. I'm off to do some testing |
| 18:46.35 | boom^time | appreciate it* |
| 18:47.14 | vandyk | I'm curious now, because in my perspective I have much more on gateway than on cards. I know that is your opinion. |
| 18:47.24 | WIMPy | vandyk: Although thare are some functions I miss when using cards as well. |
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| 18:57.14 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:57.53 | n3rdyguy | Hi |
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| 19:25.47 | danbell77 | penguin: Do you have experience with the 7970? Do you know which version of skinny works best with chan-sccp? |
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| 19:27.02 | Penguin | danbell77: I don't personally use any 7970s, but I do use 7960s. I would suggest the latest version you can get for the phones and use chan-sccp-b 4.1-STABLE on asterisk. |
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| 19:36.41 | danbell77 | penguin: I'm not to crazy about even going down the skinny route as I have an entire platform built around SIP... |
| 19:38.34 | carrar | 7970 will do sip just fine |
| 19:38.47 | carrar | as 7975 also |
| 19:39.38 | danbell77 | I can make the phones work fine in SIP...my issue is getting the buttons to behave properly. |
| 19:39.48 | carrar | ah |
| 19:39.49 | danbell77 | And conference calling seems to be an issue |
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| 19:56.05 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 19:56.14 | gmalsack | yo all... |
| 19:57.12 | gmalsack | so here's the deal. sales manager wants round robin on call queue.... ok easy. however he wants people that are skipped because they are already on the phone to be next in line..... any ideas? |
| 19:57.28 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 19:58.16 | [TK]D-Fender | No solution. |
| 19:58.33 | [TK]D-Fender | The strategies you see listed in the sameple config are what you've got |
| 19:59.05 | gmalsack | fuck..... that's what I was afraid was going to be the answer..... another custom devel here we come... :-( ugh... |
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| 19:59.58 | [TK]D-Fender | just because they were skipped for being busy in a previous pass does not mean they are answering any fewer calls |
| 20:00.41 | gmalsack | I know. it's a mental game the management wants to play to give an incentive to those working.... |
| 20:00.50 | gmalsack | *actually working... ;-) |
| 20:00.57 | boom^time | How about random eavesdropping? |
| 20:01.06 | gmalsack | oh they do that too... |
| 20:01.25 | boom^time | Manager gets an even bigger power trip, employees work harder, win win. |
| 20:01.58 | [TK]D-Fender | gmalsack: Doesn't eman they'll answer their phones. |
| 20:02.09 | gmalsack | this call center is like a boiler room. constant pressure. |
| 20:02.17 | [TK]D-Fender | gmalsack: You know.. that's why you have QUEUE LOGS... to evaulate performace... |
| 20:02.52 | *** join/#asterisk danjenkins_ (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 20:03.14 | gmalsack | tkd-fender: I know. you're preaching to the choir here.... but I'm just a techy.... what do I know. lol |
| 20:03.36 | boom^time | Tell him you took sweat shop 101 back in college. |
| 20:05.26 | *** join/#asterisk ttyp123 (~ttyp1@59.90.163.87) |
| 20:10.53 | paulc | What's wrong with "longest waiting call to longest idle agent"? Seems.. "fairer"? in terms of call distribution.. maybe? or am I missing something blindingly obvious? |
| 20:12.01 | [TK]D-Fender | paulc: His manager coming up with some random idea out of their head and expecting it to become reality just because they thought of it |
| 20:12.29 | gmalsack | [TK]D-Fender: lol !!! nice! :-) |
| 20:13.02 | paulc | I'm always a fan of "I'm sorry, it just doesn't work that way. Here's what I CAN do for you - I think it covers most of what you want, right? RIGHT?! <jedi mind control stare> " |
| 20:13.39 | [TK]D-Fender | "You want to make people more productive? This is NOT a productive use of MY time" |
| 20:13.41 | gmalsack | paulc: awesome!!! :-) unfortunately he's an owner as well, sooooo..... |
| 20:14.54 | paulc | soooooo? who cares? You're a professional, he's paying you to do a job, and with the tools available, here's how it works.. if he wants you to start hacking internals of Asterisk to make his pipe dream a reality.. wellllllll... it becomes way less maintainable in the long run! |
| 20:14.55 | danbell77 | Has anyone used a 7970's conference button successfully using SIP? |
| 20:14.57 | gmalsack | basically he doesn't want sales reps that spent 20 minutes on the phone with a client to feel like they are missing out on sales leads because when their done their phone doesn't ring for another 20 minutes until the round robin works it's way back around.... |
| 20:15.16 | paulc | And I'm sure he'd rather spend less on maintenance in the future, right? RIGHT?! <jedi mind control stare x2> |
| 20:15.47 | gmalsack | paulc: I like your way of thinking! |
| 20:15.49 | paulc | So skip round robin and most most idle (or least recent, or whatever its called) |
| 20:16.05 | paulc | It's way fairer because long calls vs short calls don't impact the distribution |
| 20:16.30 | paulc | meaning the reps can deliver good service, rather than trying to make calls as short as possible.. so they're more likely to get hit with the robin.. |
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| 20:18.32 | gmalsack | paulc: thanks for the thoughts! I think I'll just slip that one in there and see what he reports without really telling him what I'm doing. I'll just say it sounded like the closest option to what he wanted.... ;-) |
| 20:19.02 | paulc | "It's what all the big expensive PBX/ACD systems do" :-) |
| 20:19.58 | paulc | He'll love that.. he's getting big box $$$$$$$ functionality for a way lower price. Isn't he clever? Doesn't his business rock? (no, not the guy doing the work, no no no, it's not about you) |
| 20:20.10 | paulc | hehe gotta love the PHBs of this world eh? :-) |
| 20:20.33 | [TK]D-Fender | [16:14]gmalsackbasically he doesn't want sales reps that spent 20 minutes on the phone with a client to feel like they are missing out on sales leads because when their done their phone doesn't ring for another 20 minutes until the round robin works it's way back around.... <- leastrecent |
| 20:26.46 | [TK]D-Fender | checkout time, heading home... |
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| 20:38.41 | datarecal | hey guys, quick question is it possible to sync my microsoft exchange contacts to my aastra ip phone ? |
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| 20:45.05 | pabelanger | datarecal, might want to ask #microsoft, since that's not asterisk related |
| 20:48.05 | *** join/#asterisk mmourani (~Adium@64.229.213.168) |
| 20:48.16 | mmourani | hi |
| 20:48.26 | mmourani | I have a problem with codec g729 |
| 20:49.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:49.53 | mmourani | can someone help with asterisk 11 ? |
| 20:50.08 | pabelanger | ~ask |
| 20:50.08 | 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. |
| 20:50.27 | vandyk | mmourain: tell us what is your problem |
| 20:50.55 | mmourani | i get an error message : no compatible codecs, not accepting this offer ! |
| 20:51.10 | anonymouz666 | [TK]D-Fender is at home now! |
| 20:51.16 | mmourani | Notice 5003 |
| 20:51.29 | [TK]D-Fender | mnathani: So go set it so both ends have something they can agree on |
| 20:51.39 | gmalsack | anonymouz666: don't you ever rest.... lol |
| 20:51.57 | anonymouz666 | me? sure I do |
| 21:13.00 | newtonr | mmourani, you need to double-check what codecs you have enabled for all phones involved, and what codecs you have allowed for their configuration within Asterisk. |
| 21:13.27 | newtonr | mmourani, phones/sip accounts/sip devices/whatever |
| 21:14.39 | newtonr | You can also search the mailing list archives and google, there is about a million conversations on that topic that will help you out. |
| 21:17.44 | [TK]D-Fender | No need to randomly search mailing lists while sitting in IRC |
| 21:23.18 | _Corey_ | OK, I may have smoked too much crack today but can anyone tell me under what scenarios Asterisk will send a local IP in its SDP OK to an INVITE from a public IP? I've got RTP going into oblivion... |
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| 21:24.46 | jeev | if i asked that question, fender's reply: not enough information. |
| 21:24.58 | *** join/#asterisk Tekz (~Tekz08@unaffiliated/tekz) |
| 21:25.49 | newtonr | _Corey_, Possibly when you don't have localnet, externaddr and what not configured appropriately, or you don't have directmedia turned off for those accounts. |
| 21:26.40 | _Corey_ | newtonr: Yeah, I'm suspecting externaddr at the moment. I've ruled everything else out. This WAS working at one point, so I'm baffled as to what had changed. |
| 21:27.44 | [TK]D-Fender | Look at the call, look at the configs |
| 21:27.57 | WIMPy | finds one-way-audio on transcoded call much more interesting. |
| 21:28.04 | _Corey_ | It's a weird environment with one trunk going out to the Internet and a lot of other stuff getting NATed in from an MPLS source who would definitely not respond favorably to the real external IP showing up |
| 21:29.05 | mmourani | ok i am just doing a passthru on my asterisk |
| 21:29.12 | mmourani | I am not transcoding g729 |
| 21:29.20 | mmourani | the two phones are talking g729 |
| 21:29.31 | mmourani | so i should not get this error on the codec |
| 21:30.09 | _Corey_ | newtonr: Yeah, that makes it work (again). I still need to investigate what happened. Anyhow, thanks for the sanity check. |
| 21:31.28 | [TK]D-Fender | mmourani: Show us the actual call. |
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| 21:33.14 | [TK]D-Fender | ~pb |
| 21:33.15 | 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. |
| 21:33.16 | [TK]D-Fender | ^^^ |
| 21:35.05 | newtonr | _Corey_, nooo problemo |
| 21:35.17 | mmourani | well i set my debug to 10 on the CLI but i get only the message : No compatible codecs, blah blah - That's it |
| 21:37.25 | [TK]D-Fender | "sip set debug on" <- |
| 21:37.44 | [TK]D-Fender | verbose will not prove what peer is hit and what each side is offering |
| 21:37.48 | mmourani | ok |
| 21:38.07 | newtonr | mmourani, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:40.56 | mmourani | ok thanks |
| 21:40.59 | mmourani | let me do this |
| 21:45.07 | Tekz | so just so i have this right, can anyone verify that i'm correct on this? If I have 4 pots lines, I need either a PCI/PCI-E card with 4 fxo ports OR some sort of fxo gateway device, yeah? |
| 21:45.32 | Tekz | and if i get an fxo gateway device, i will basically need to set it up so it points to the IP address of my asterisk box once I get it set up? |
| 21:45.35 | [TK]D-Fender | Tekz: Well you need something to plug them into |
| 21:45.46 | *** join/#asterisk vomit (~Unknown@unaffiliated/vomit) |
| 21:46.12 | [TK]D-Fender | Tekz: Fora SIP gateway... it speaks SIP ... it's little different than anything else. |
| 21:47.46 | Tekz | so i'm correct then? |
| 21:47.55 | Tekz | just trying to wrap my head around this. |
| 21:48.17 | vomit | Hi. I wanted to ask about the experience with ReceiveFAX function to receive faxes. Faxes sent to me are getting fax at first attempt |
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| 21:49.14 | vomit | Hi. I wanted to ask about the experience with ReceiveFAX function to receive faxes. Faxes sent to me are getting fax session timed-out error at first attempt but then they succed. Anybody experiences similiar issues. Kinda ancient asterisk at that setup (1.4). |
| 21:50.02 | WIMPy | Tekz: Or as a third option you can use an ITSP. |
| 21:50.59 | Tekz | That's not an option on the table for us at this point. |
| 21:51.24 | WIMPy | Might work better. |
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| 21:54.49 | mmourani | thanks guys |
| 21:54.54 | mmourani | i solved the problem |
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| 23:56.31 | vandyk | someone is having issues with Asterisk 11.5.1 with a lot of SIP 487 ? |