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| 00:52.18 | rogers- | So we got a new modem from time warner, a DDW3611. Looks like you cannot put it into bridge mode, and have time warner static ips assigned. They assign the ips to the modem, and do some sort of static translation |
| 00:52.30 | rogers- | so my incoming sip connections do not work |
| 00:52.53 | rogers- | apparently after hours on the support with time warner, its a known issue |
| 00:53.14 | rogers- | so having them put back in the old modem tomorrow |
| 00:53.20 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-dvoajbtrqrvzmyvj) |
| 00:53.40 | rogers- | some sort of translation or double NAT problem going on |
| 01:01.27 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
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| 01:18.50 | gusto | so |
| 01:19.05 | gusto | hey Penguin, how are you? |
| 01:23.51 | AkkerKid | echo... |
| 01:24.14 | AkkerKid | ping... |
| 01:24.18 | AkkerKid | Marco... |
| 01:24.21 | navaismo | pong.. |
| 01:24.25 | AkkerKid | WOO |
| 01:24.44 | AkkerKid | i'm not the last system admin alive... |
| 01:25.07 | navaismo | im not system admin |
| 01:25.28 | AkkerKid | well, i guess I have to do all the repopulating myself then... |
| 01:29.22 | navaismo | mic + speakers alot of echo |
| 01:31.04 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 01:32.22 | AkkerKid | pulling all nighters alone at work sucks |
| 01:32.56 | AkkerKid | the airconditioning turns off too. |
| 01:33.54 | WIMPy | And the security system is on so you can't move? |
| 01:35.27 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:35.33 | AkkerKid | not quite. |
| 01:35.46 | AkkerKid | if there was a security system, I'd have installed it. |
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| 05:11.54 | Rahail | Hi there I have question not realted to asterisk.. i need help if any one can help me |
| 05:12.11 | Rahail | we have no profit organization where we have about 40 grandstream ATA |
| 05:12.15 | Rahail | and we use asterisk |
| 05:12.33 | Rahail | but lack of knowledge i can not figerout how to configure them with asterisk auto |
| 05:25.34 | jpsharp | Asterisk itself cannot configure the devices. |
| 05:26.25 | Rahail | is there any sort config file etc |
| 05:26.30 | Rahail | i can just push to ata's |
| 05:29.17 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
| 05:29.37 | jpsharp | You can't "push" it to the ATAs. You have to configure them to pull it. |
| 05:30.35 | Rahail | how |
| 05:30.37 | jpsharp | Or if you can, configure the network that they are on to tell the ATAs where to pull the config files from. |
| 05:30.48 | Rahail | can they pull from remote location |
| 05:31.40 | jpsharp | You either have to manually configure each ATA with provisioning information *or* you can tell the DHCP server on the network the ATAs are on to hand the ATAs the information they need when they power up and get an IP address. |
| 05:32.00 | jpsharp | Assuming you have a DHCP server and it is smart enough to do that. |
| 05:36.33 | jpsharp | search for "grandstream auto provisioning". That should get you started where you need to go. |
| 05:37.12 | Rahail | thank you jpsharp |
| 05:53.48 | apb1963_ | my softphone won't register... this is recent... not sure what I managed to break. running asterisk 1.8 trimmed logfile here: http://ix.io/3VK extension of current interest is 4200.... |
| 05:55.27 | apb1963_ | any ideas as to what might be wrong? |
| 06:14.22 | ChannelZ | well from what I see it's not even trying to auth when asked but perhaps you just cut the log off |
| 06:15.53 | apb1963_ | channelZ... I don't understand? |
| 06:16.44 | apb1963_ | it says SIP/2.0 401 Unauthorized |
| 06:17.15 | apb1963_ | which I also don't understand. What is it trying to authorize? I have the right password... what else does it want? |
| 06:17.35 | ChannelZ | That's not how it works. |
| 06:17.53 | apb1963_ | Prolly why I don't understand what's wrong |
| 06:19.06 | ChannelZ | First it tries to REGISTER with no credentials really. Asterisk replies Unathorized and via the WWW-Authenticate header basically asks it to try again with its password hashed with the nonce it gave. |
| 06:19.15 | ChannelZ | At which point it should try again.. but I don't see that. |
| 06:19.29 | ChannelZ | I see another peer 301 from that same IP try at the end of the log |
| 06:20.26 | apb1963_ | Yeah I'm not sure where that's coming from... I used to have 301 configured, but I thought I disabled that.... but then, I've gone through so many softphones and configs I'm not sure if I'm coming or going. |
| 06:20.29 | ChannelZ | or rather I see it try to SUBSCRIBE. But then the log ends. So maybe the peer in question did try to re-register but you stopped too soon, I don't know. |
| 06:21.05 | ChannelZ | they're both coming from Zoiper |
| 06:21.38 | apb1963_ | Yes... it tries to re-register continuously... I can provide a longer log.... I just know some people complain when I have too many lines so I limited it to what I thought was reasonable. |
| 06:21.59 | apb1963_ | I just started a different softphone |
| 06:22.22 | apb1963_ | log coming up |
| 06:23.03 | apb1963_ | http://ix.io/3VR |
| 06:24.22 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 06:24.26 | apb1963_ | you want me to disable the extra trunks to make it easier to read? |
| 06:25.04 | apb1963_ | the softphone says that asterisk isn't responding to the register request |
| 06:25.04 | ChannelZ | yeah, now I don't even see that same IP even doing anything |
| 06:25.13 | *** join/#asterisk aruntomar (~crazy@61.17.193.163) |
| 06:25.22 | apb1963_ | kk just a sec, I'll disable |
| 06:31.09 | apb1963_ | http://ix.io/3VU |
| 06:33.25 | ChannelZ | I still don't see it even trying to auth |
| 06:34.32 | ChannelZ | I think you have multiple things running because I'm still seeing 201 |
| 06:35.00 | aruntomar | hi. total digits in my phone no. are 8. in my dialplan i'm using last 4 digits as extension no. using pattern matching _XXXX, but rather than just dialing the extension, asterisk is dialing the entire 8 digits and gives error that the no. is not in the context. can somebody help me with that? |
| 06:35.51 | ChannelZ | So it's only already sending just the last 4 digits to Asterisk? |
| 06:35.54 | apb1963_ | yes the softphone has multiple extensions... |
| 06:36.26 | apb1963_ | I guess I'll disable those too |
| 06:36.35 | ChannelZ | well neither of them seem to be answering the auth request |
| 06:36.52 | apb1963_ | meaning asterisk isn't answering or the softphone isn't answering? |
| 06:37.06 | ChannelZ | they should try again with a Authorization: header with a hash |
| 06:37.20 | kaldemar | aruntomar: asterisk only dials what you tell it to. pastebin a CLI output of a call. |
| 06:37.38 | apb1963_ | so... the softphone should try again? |
| 06:37.54 | ChannelZ | The softphone is being denied for not providing authorization, but then it's not trying again WITH authorization |
| 06:38.27 | apb1963_ | so perhaps I screwed up some config option? I was monkeying... |
| 06:38.42 | ChannelZ | monkeying on the Asterisk side? |
| 06:38.46 | apb1963_ | yes |
| 06:38.50 | apb1963_ | well, both sides really |
| 06:38.53 | apb1963_ | but yes |
| 06:39.32 | apb1963_ | I have an extension overseas that was having problems so I fiddled until it broke completely :) |
| 06:39.37 | kaldemar | åä |
| 06:39.51 | ChannelZ | well I'm not sure what on the Asterisk side would cause this behavior since it's the softphone's lack of action, at least from what you've shown |
| 06:40.06 | apb1963_ | well, softphone says asterisk isn't responding |
| 06:40.21 | ChannelZ | then perhaps you have a networking/routing problem |
| 06:40.52 | apb1963_ | did you notice that both from and to are from the same host? Is that normal? |
| 06:41.09 | aruntomar | kaldemar: http://pastebin.com/asuMDwjs |
| 06:41.38 | ChannelZ | yes |
| 06:42.41 | ChannelZ | aruntomar: well it's telling you there is no extension 12341700 (or anything that matches) |
| 06:43.03 | kaldemar | aruntomar: your asterisk is getting 8 digits. |
| 06:43.08 | ChannelZ | _XXXX is only 4, it won't automatically match the last 4 which is what it sounds like you're after |
| 06:43.32 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:44.27 | ChannelZ | apb1963_: Is your Asterisk behind a firewall? Is the machine at .244? |
| 06:44.49 | apb1963_ | well, the only network mucking I did was to add a vpn address |
| 06:45.03 | aruntomar | kaldemar: so do i need to modify all my extensions to use 8 digits rather than 4? |
| 06:45.04 | ChannelZ | seems significant |
| 06:45.29 | apb1963_ | Yes to firewall (though I disabled it at one point for testing... it's back up now) |
| 06:45.38 | apb1963_ | iptables firewall |
| 06:45.44 | apb1963_ | no mucking there! |
| 06:45.51 | aruntomar | ChannelZ: the exten is 1700, and i was expecting that asterisk will dial 1700 rather than 12341700 |
| 06:46.18 | apb1963_ | softphone is at 144 |
| 06:46.32 | kaldemar | aruntomar: it is not asterisk that dials the 8-digits. |
| 06:46.54 | apb1963_ | oops... 244 |
| 06:47.18 | ChannelZ | aruntomar: Your phone is sending 12341700 to Asterisk. If you are only interested in doing something with the last 4, fine, but you still have to have a matching extension that would match all 8 for it to execute at all. |
| 06:48.59 | ChannelZ | Like _XXXXXXXX and then ${EXTEN:4} would represent just the last 4 (by skipping the first 4 with the :4) |
| 06:49.00 | aruntomar | ChannelZ, kaldemar ok, got it. my understanding was that we decide, how many digits to use in extension, and accordingly dial it. so i guess, i'll modify my dialplan and extensions to match the 8 digit pattern then. |
| 06:49.54 | aruntomar | ChannelZ: owesome, i'll try that ${EXTEN:4} |
| 06:50.53 | kaldemar | aruntomar: you understood right that you can decide what your extensions are and how long they are. but for a 4-digit extension to match you need to dial only 4 digits, not 8. |
| 06:51.10 | apb1963_ | I just deleted the vpn from asterisk... no effect. |
| 06:52.10 | apb1963_ | wait a minute |
| 06:52.13 | ChannelZ | apb1963_: Can you ping the .244 IP from the asterisk machine? It sounds like the return traffic isn't making it off the asterisk machine, or is and isn't making it to the .244 machine, for whatever reason that might be in your config |
| 06:52.38 | ChannelZ | I'm assuming if you monitored the SIP debug longer you'd see a few "Retransmitting..." lines |
| 06:53.53 | apb1963_ | dns appears to think both machines have the same address |
| 06:53.57 | apb1963_ | jeez |
| 06:57.10 | *** join/#asterisk vedic (~V@183.82.26.208) |
| 06:59.39 | vedic | Hey friends, could you suggest which software/module can connect with Asterisk and provide CRM like facility. What I am looking for is when I call to asterisk box, that software should be able to pop up or manual entry facility to get my details associated with my phone number (consider that this associated information is available but should have a way to enter into the software connected to asterisk) |
| 07:00.55 | apb1963_ | da hello... that didn't change anything :-S |
| 07:01.10 | apb1963_ | yes I can ping both directions |
| 07:02.03 | apb1963_ | wait... dns is still screwy for some reason |
| 07:05.28 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:06.11 | vedic | any suggestion for open source crm which is well documented to how to use? |
| 07:06.13 | *** part/#asterisk vedic (~V@183.82.26.208) |
| 07:07.03 | ChannelZ | (no) |
| 07:07.24 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
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| 07:12.38 | apb1963_ | 3VY |
| 07:13.06 | apb1963_ | http://ix.io/3VY |
| 07:13.07 | ChannelZ | 8hc |
| 07:13.14 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 07:13.22 | apb1963_ | I fixed DNS.. no change in problem |
| 07:13.29 | apb1963_ | I have no idea how dns got screwed up |
| 07:13.44 | apb1963_ | but fixing it made no difference and that is all totally bizarre |
| 07:14.47 | ChannelZ | Don't know what to tell you. Either Asterisk's replies aren't making it back to that machine, or it's so upside-down it's not paying attention to them. |
| 07:15.31 | apb1963_ | well, I'll reboot the asterisk server just for giggles... can't hurt. |
| 07:19.44 | *** join/#asterisk vedic (~V@183.82.26.208) |
| 07:19.55 | apb1963_ | that didn't help... time for a little wireshark ay? |
| 07:19.56 | *** join/#asterisk aruntomar (~Thunderbi@117.219.113.75) |
| 07:20.34 | aruntomar | ChannelZ: what u suggested worked ${EXTEN:4} |
| 07:20.45 | aruntomar | ChannelZ: kaldemar: thanks for your help :) |
| 07:24.56 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 07:33.13 | vedic | which one is better openERP or vTiger for connection with Asterisk? Any practical experience? |
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| 07:46.52 | schmidts | good morning |
| 07:58.52 | oej | morning |
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| 08:33.50 | pbxman | morning |
| 08:34.00 | ChannelZ | yawns |
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| 08:52.58 | BlackDex | Hello there.. i'm trying to use AUDIOHOOK_INHERIT but it is not registerd according to rasterisk |
| 08:55.08 | kaldemar | what version of asterisk are you using? |
| 08:55.16 | BlackDex | 1.6.2.20 |
| 08:55.19 | *** join/#asterisk v0lZy (~chatzilla@84-255-194-41.static.t-2.net) |
| 08:56.17 | kaldemar | what does "module load func_audiohookinherit.so" tell you? |
| 08:56.32 | kaldemar | and what is the exact print in CLI you see? |
| 08:56.37 | BlackDex | Loaded func_audiohookinherit.so => (Audiohook inheritance function) |
| 08:56.43 | BlackDex | hmm |
| 08:57.24 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 08:58.57 | BlackDex | kaldemar: thx... :) |
| 08:59.24 | kaldemar | looks like you didn't have the module loaded for some reason. check that you don't have a noload for it in /etc/asterisk/modules.conf. |
| 08:59.41 | BlackDex | yea.. just checked it.. it was missing |
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| 10:51.10 | bulkorok | I use script-generated callfiles. I have one with the errer: Call failed to go through, reason (3) Remote end Ringing |
| 10:51.28 | bulkorok | any ideas what that is telling me!? |
| 10:51.48 | WIMPy | Noone answered? |
| 10:52.02 | bulkorok | I suppose, but have to be sure... |
| 10:52.25 | bulkorok | the end is talling me the fax is fine |
| 10:52.32 | bulkorok | s/talling/telling |
| 10:52.34 | WIMPy | The timing will tell you. |
| 10:52.46 | bulkorok | timing!? |
| 10:52.54 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 10:54.09 | WIMPy | The time the call took. |
| 10:54.27 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 10:54.56 | bulkorok | ah... you mean time from dialing number until other end is taking the call... |
| 10:55.07 | WIMPy | Or not. |
| 10:55.23 | bulkorok | yeah... I have WaitTime: 30 |
| 10:55.56 | bulkorok | I'll check the other end... |
| 10:58.06 | bulkorok | it takes really long until I hear the fax-beeps... |
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| 11:56.55 | hrolf | Hi #asterisk, I need to ask a question, is there any way I can get the SIP CallID in the dialplan and set it as variable? |
| 12:00.38 | kaldemar | hrolf: ${SIP_HEADER(Call-ID)} |
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| 12:29.28 | ratbox | yawns |
| 12:36.18 | hrolf | kaldemar: thanks. |
| 12:37.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 14:05.50 | Katty | hello my asterisk does not work at all how to fix plz??? is urgent thx |
| 14:06.53 | chuckf | dial the right numbers |
| 14:07.11 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ocdnlqllyamdntrw) |
| 14:07.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:07.49 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 14:07.52 | [TK]D-Fender | Katty, Pre-heat the over to 350F then cook for 10 minutes/side. Baste between inversions. |
| 14:08.55 | Katty | that's not nearly enough time to cook anything. |
| 14:08.57 | Katty | not even bacon. |
| 14:09.18 | chuckf | mmm bacon |
| 14:09.47 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.71) |
| 14:16.56 | Katty | hi chuckf |
| 14:17.44 | chuckf | Hiya Katty, how is your morning going so far? |
| 14:18.05 | coppice | I remember when I was a lad and ovens had labels like "350F" |
| 14:18.18 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 14:19.07 | chuckf | coppice: mine still does, of course it was probably made when you were a lad.... |
| 14:19.53 | coppice | chuckf: is it wood burning? |
| 14:21.11 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 14:21.59 | chuckf | nope |
| 14:27.59 | Katty | chuckf: irritating, as usual. |
| 14:28.15 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.83.238) |
| 14:28.21 | Katty | chuckf: but at least people weren't meeting me in the parking lot with their problems. |
| 14:28.34 | chuckf | Katty: that's an improvement |
| 14:28.39 | Katty | mhmm |
| 14:29.32 | chuckf | Just tell them to take a number and you might get to them at some point |
| 14:31.08 | *** join/#asterisk volga629 (~volga629@host7.pythian.com) |
| 14:31.16 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.71) |
| 14:32.48 | volga629 | Hello Everyone, I am looking for information about PRACK asterisk 10 |
| 14:32.58 | *** join/#asterisk sustav (~vpp@nat/digium/x-niuyphnhhcdxefss) |
| 14:33.40 | carrar | bacon? |
| 14:34.33 | kaldemar | volga629: PRACK asterisk? |
| 14:35.14 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 14:35.16 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:5dd3:e94d:6827:bdcd) |
| 14:37.34 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:38.02 | volga629 | yes |
| 14:38.27 | chuckf | what is PRACK asterisk? |
| 14:39.30 | kaldemar | volga629: please elaborate. |
| 14:41.42 | volga629 | SIP/2.0 405 Method Not Allowed |
| 14:41.51 | volga629 | Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, UPDATE |
| 14:43.18 | [TK]D-Fender | volga629, You know Google offers a whole lot of answers with 3 magic words entered into the "search" box. |
| 14:43.33 | [TK]D-Fender | volga629, Including references to code branches you could look into... |
| 14:43.55 | coppice | [TK]D-Fender: 3 words? you mean "WTF"? |
| 14:44.20 | [TK]D-Fender | coppice, No, just like Scrabble, acronyms don't count ;) |
| 14:45.05 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:45.47 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat35.beelinegprs.ru) |
| 14:46.12 | kaldemar | volga629: rfc3262 |
| 14:47.09 | kaldemar | but prack probably has nothing to to with your issue, which is... what? |
| 14:48.55 | [TK]D-Fender | volga629, And quick tip : Asterisk 10 is dead. No more bug fixes. |
| 14:49.10 | [TK]D-Fender | volga629, Certainly no new features. |
| 14:50.30 | volga629 | http://fpaste.org/rmmB/ |
| 14:50.39 | volga629 | so need go to 11 |
| 14:50.45 | Faustov | [TK]D-Fender: really? that's earlier than this states: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 14:51.20 | Faustov | oh, nevermind |
| 14:51.21 | kaldemar | volga629: they did not like your NOTIFY. |
| 14:51.46 | Faustov | i'd swear the dates were further in the future... |
| 14:52.46 | volga629 | Notify might need set to register |
| 14:53.38 | *** join/#asterisk PbxMan (c335d91f@gateway/web/freenode/ip.195.53.217.31) |
| 14:54.05 | volga629 | you mean CSeq: 102 NOTIFY ? |
| 14:54.07 | kaldemar | volga629: no, a notify is not needed in registrations. REGISTER does that. what issue are you experiencing? |
| 14:54.24 | volga629 | I just see in log |
| 14:54.43 | volga629 | I am trying test build in client on Fedora 17 |
| 14:56.44 | kaldemar | volga629: look at the NOTIFY that causes that. |
| 14:57.13 | volga629 | SIP/2.0 405 Method Not Allowed I see this message and just trying find what settings on client cause it |
| 14:58.26 | *** join/#asterisk vlad_sta_ (~vlad_star@79.104.4.238) |
| 14:58.49 | kaldemar | volga629: maybe you should show the whole sip debug for that and the settings. |
| 15:00.18 | volga629 | I can print screen from settings |
| 15:04.02 | kaldemar | volga629: settings on asterisk side. |
| 15:05.11 | volga629 | yes |
| 15:09.11 | *** join/#asterisk mzb_ (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 15:09.47 | [TK]D-Fender | volga629, You are showing the answer that * gave ... and not showing us the complete dialog it is in response to. |
| 15:10.10 | [TK]D-Fender | volga629, Don't ask if the answer is right is you aren't showing us the question |
| 15:10.38 | volga629 | I am working to pull debug andd settings |
| 15:12.52 | kaldemar | actually the 405 was to asterisk from rakia. asterisk sent the notify. |
| 15:13.33 | kaldemar | probably just a MWI. |
| 15:16.11 | volga629 | http://fpaste.org/UQwx/ |
| 15:19.04 | *** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
| 15:19.46 | [TK]D-Fender | volga629, Where's the 405 in there? |
| 15:19.53 | kaldemar | it's not. |
| 15:20.04 | [TK]D-Fender | </rhetorical> |
| 15:20.08 | [TK]D-Fender | :) |
| 15:20.09 | kaldemar | and neither are the settings. |
| 15:22.37 | volga629 | I don't see it right now ??? :-( |
| 15:22.55 | volga629 | no I see again |
| 15:23.51 | volga629 | http://fpaste.org/0WLm/ |
| 15:25.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:25.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:27.47 | *** join/#asterisk MooJuice (41725fe2@gateway/web/freenode/ip.65.114.95.226) |
| 15:33.40 | volga629 | I will be right back |
| 15:33.52 | *** part/#asterisk volga629 (~volga629@host7.pythian.com) |
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| 16:04.54 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 16:05.55 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 16:09.01 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
| 16:11.17 | *** join/#asterisk volga629 (~volga629@host7.pythian.com) |
| 16:12.13 | volga629 | You think this message fro WMI ? |
| 16:12.25 | *** part/#asterisk charley (charley@epicboise.com) |
| 16:12.46 | [TK]D-Fender | volga629, Think? |
| 16:12.55 | [TK]D-Fender | volga629, No, we SEE the packet right up there. Don't you? |
| 16:14.55 | MooJuice | Anyone else seeing a problems with SIP call transfers in Asterisk 10.x? I seem to have sporadic handoffs where both parties hear MOH then the caller gets hung up on / never makes it to the transferee. I get the SIP-Notifiy OK in the logs. |
| 16:15.50 | MooJuice | Asterisk 10.11.1 seems to make the behavior happen much more frequently |
| 16:16.48 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 16:17.45 | volga629 | hmmm, yes wee see packet, very interesting who cause for 405 |
| 16:17.54 | [TK]D-Fender | MooJuice, you should be upgrading to * 11. 10 is no longer supported |
| 16:18.57 | MooJuice | [TK]D-Fender: Id love to.. easier said than done though in prod enviornments ;) |
| 16:19.28 | [TK]D-Fender | MooJuice, You know what's harder? Getting bug fixes that aren't coming :) |
| 16:19.56 | [TK]D-Fender | MooJuice, Guess the prioritization level just went up... |
| 16:20.44 | MooJuice | [TK]D-Fender: yea, I was more wondering if this was a known issue (as I can't find anything similar to it in jira) or if im the lucky winner... |
| 16:22.35 | [TK]D-Fender | MooJuice, http://www.funnypics.co.za/plog-content/images/general/demotivators/misfortune-1.jpg |
| 16:22.52 | MooJuice | hah |
| 16:23.35 | MooJuice | [TK]D-Fender: you have any good experiences with * 11? im a heavy confbridge user |
| 16:23.37 | *** join/#asterisk DarthExpeditor (~IceChat9@71.46.209.2) |
| 16:27.21 | [TK]D-Fender | ConfBridge got a LOT more powerful with * 11 |
| 16:27.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:27.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:27.47 | [TK]D-Fender | You should probably visit asterisk.org and check their wiki and read up on all the changes... |
| 16:32.39 | MooJuice | [TK]D-Fender: ive taken a gander already.. looks nice haven't heard of any success stories from the front lines though.. |
| 16:35.47 | MooJuice | [TK]D-Fender: building 11.1.2.. this better fix my problem.. :P |
| 16:38.15 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.83.238) |
| 16:42.51 | *** join/#asterisk elico (~Thunderbi@109.66.88.247) |
| 16:43.12 | Katty | beats vsphere |
| 16:43.22 | Katty | be good! or i hang you with yarn! |
| 16:44.24 | Katty | speaking of vsphere |
| 16:44.30 | Katty | how does asterisk do in a virtualized enviornment? |
| 16:44.40 | Katty | does it barf all over itself? |
| 16:44.49 | [TK]D-Fender | MooJuice, It's best to look for horror stories. Those are the important ones. |
| 16:44.56 | [TK]D-Fender | MooJuice, No news is good news. |
| 16:45.16 | [TK]D-Fender | MooJuice, Do read the UPGRADE docs though. |
| 16:47.13 | _Corey_ | Katty: It can work fine.. We've done some large deployments using ESXi |
| 16:47.19 | Katty | post 4? |
| 16:47.50 | _Corey_ | We've done it on 4 and 5 |
| 16:48.39 | chuckf | Katty: From what I've heard most of it is fine, except things that rely on cpu timers (such as confrencing) But I don't know if its better in v11 |
| 16:48.58 | Katty | _Corey_: how about conferencing? |
| 16:52.55 | _Corey_ | Yeah, no major issues. I don't know that I'd expect to scale huge though on conferencing |
| 16:53.16 | Katty | i may have to try that out sometime. |
| 16:53.30 | Katty | you do thin provisioning or thick? |
| 16:54.28 | carrar | medium slice! |
| 16:54.56 | Katty | a medium slice does sound much more reasonable. |
| 16:55.04 | Katty | unless it's chocolate. |
| 16:55.37 | carrar | Well you don't provision chocolate |
| 16:55.42 | carrar | You jsut site down and eat it all |
| 16:56.08 | Katty | pff. i provision my chocolate. |
| 16:56.18 | _Corey_ | Katty: you mean thin vs. thick on the virtual disk? |
| 16:56.23 | Katty | but in the past i did. |
| 16:56.27 | Katty | _Corey_: yes'r |
| 16:56.38 | Katty | _Corey_: do you reserve a block of space |
| 16:56.41 | _Corey_ | I don't think we've seen any difference to speak of |
| 16:56.52 | _Corey_ | ... as in, we've done both |
| 16:56.52 | Katty | no performance issues with call recording |
| 16:57.01 | Katty | or...other disk hogging things |
| 16:57.13 | Katty | fancy |
| 16:57.51 | _Corey_ | not really... we've got a largish call center customer running on ESXi doing all that stuff |
| 16:58.19 | carrar | SSD SAN! |
| 16:58.19 | Katty | what do you figure the call volume of this 'largeish' call center is |
| 16:58.21 | carrar | WTF |
| 16:58.33 | _Corey_ | come to think of it, I did a case study talk on it at Astricon this year... you can check out the video if you want |
| 16:58.37 | _Corey_ | 350+ agents now |
| 16:58.38 | carrar | err |
| 16:58.39 | carrar | FTW |
| 16:58.40 | carrar | heh |
| 17:00.03 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:00.25 | carrar | Is there dancing in the video Corey? |
| 17:00.46 | carrar | Asterist Style |
| 17:01.12 | carrar | s/Asterist/Asterisk/ |
| 17:01.43 | _Corey_ | carrar: Not so much in my sessions... ;) |
| 17:02.17 | Katty | well i may give it a go on vsphere 5 then |
| 17:02.25 | Katty | see if i can make it CRASH AND BURN |
| 17:02.31 | _Corey_ | 5 performs better than 4 if I remember correctly |
| 17:02.33 | Katty | which doesn't take much anyway |
| 17:03.52 | Katty | you guys have a favorite carrier? |
| 17:03.57 | Katty | we've always used bandwidth.com |
| 17:04.03 | Katty | but they're always cranky, it seems. |
| 17:04.31 | _Corey_ | Ast far as ITSPs go they're OK... had their ups and downs but seem to generally be alright |
| 17:04.59 | Katty | i guess if i worked on a support team like that all day i'd be cranky too ;) |
| 17:05.32 | Katty | i should hire carrar to be our carrier |
| 17:05.37 | Katty | he can run tin cans on a string between here and stl |
| 17:06.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:06.23 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 17:06.27 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:18.59 | *** join/#asterisk greenwolf (c087e3a3@gateway/web/freenode/ip.192.135.227.163) |
| 17:22.24 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 17:23.42 | SuperNull | i recall a servie called 'The Voice' (of asterisk) anyone know if that still exists? |
| 17:25.15 | [TK]D-Fender | SuperNull, http://theivrvoice.com/ |
| 17:26.13 | SuperNull | thanks TK. |
| 17:26.32 | SuperNull | damn she looks like she needs a big .. |
| 17:26.36 | SuperNull | script. |
| 17:28.40 | navaismo | ey! |
| 17:30.05 | navaismo | I need Siri voice for my asterisk or better the Bane's voice from D.K.R. |
| 17:32.11 | SuperNull | banes voice was retarded. |
| 17:32.18 | SuperNull | i stopped watching partially because of that. |
| 17:32.45 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 17:32.47 | *** join/#asterisk JohnnyAsterisk (~cianmaher@154.50.194.130) |
| 17:33.22 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:33.55 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.238) |
| 17:39.03 | greenwolf | are there any good available opensource voice to use with asterisk |
| 17:39.18 | greenwolf | i also need to have the voice be able to do text-to-speech |
| 17:40.08 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:40.10 | Merlin | cepstral is the easiest to work with, but you have to pay |
| 17:40.17 | navaismo | greenwolf, festival is free |
| 17:40.21 | navaismo | but sucks |
| 17:40.39 | greenwolf | yea i know but its terrible |
| 17:40.45 | greenwolf | anything else available that im missing? |
| 17:40.50 | navaismo | googletts is the best but i think we are breaking the TOS |
| 17:41.06 | Merlin | you could just use twilio and forget asterisk |
| 17:41.16 | navaismo | and maybe in the future we need to pay hahaha |
| 17:41.19 | greenwolf | u mean for the whole system ? |
| 17:41.33 | Merlin | just for the IVR |
| 17:41.58 | navaismo | greenwolf, try the googletts and compare |
| 17:42.37 | greenwolf | im not familiar with googletts? |
| 17:42.40 | greenwolf | is it a voice? |
| 17:42.54 | greenwolf | and Merlin does twilio work like asterisk? |
| 17:43.04 | greenwolf | do i get the same features as if developing with asterisk? |
| 17:43.26 | Merlin | that's a kind of big question to answer |
| 17:43.53 | Merlin | twilio has lots of info on their website, you should read it |
| 17:43.58 | Merlin | listen, i prefer asterisk |
| 17:44.08 | Merlin | but if you can't spend any money, you should look at all options |
| 17:44.58 | *** join/#asterisk timahvo1 (~rogue@197.180.179.215) |
| 17:45.02 | greenwolf | i think half the fun for me is to actually program in asterisk and being able to develop custom code |
| 17:45.06 | _Corey_ | Cepstral is VERY inexpensive |
| 17:45.08 | greenwolf | thats 99% of the fun for me |
| 17:45.31 | greenwolf | yea i was looking at cepstral how is the audio quality on that? has anyone used it before? |
| 17:45.40 | navaismo | yes but ceptrasl voice really sucks |
| 17:45.54 | greenwolf | don't they have many different voices to choose from tho? |
| 17:46.08 | _Corey_ | There are many voices, yes--including Allison |
| 17:46.09 | greenwolf | where can i get a really good voice if im willing to pay |
| 17:46.20 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 17:46.21 | greenwolf | but the voice has to have text-to-speech capibility |
| 17:46.37 | mchou | nuance |
| 17:46.46 | mchou | $$$$ |
| 17:46.53 | saint_ | hi all |
| 17:47.22 | navaismo | greenwolf, http://zaf.github.com/asterisk-googletts |
| 17:47.39 | saint_ | I just received my first digium phone.. I was able to set it up quickly , I can make / receive calls from the outside, but there is no voice path with other registered users. Is ulaw enought as a codec in the sip.conf for this phone ? |
| 17:48.18 | ChannelZ | Should be |
| 17:48.25 | greenwolf | question...is canreinvite depreciated |
| 17:48.43 | ChannelZ | greenwolf: it's called directmedia now depending on your version |
| 17:48.43 | navaismo | change to directmedia |
| 17:48.47 | greenwolf | if i wanted asterisk to re-write the SIP packets so that asterisk doesn't handle media |
| 17:49.10 | greenwolf | so i would use directmedia if i didn't want asterisk in the way or handling media during a sip call correct? |
| 17:49.36 | greenwolf | it would have to re-write the SIP headers with the location of RTP media which would be the sip phone or device location correct? |
| 17:49.38 | ChannelZ | Yes |
| 17:49.59 | greenwolf | does it work well? does it check it NAT is in the way and if it is does it continue to handle media then? |
| 17:50.19 | greenwolf | or will it force it anyway no matter what about NAT |
| 17:50.48 | ChannelZ | Well it's up to the endpoints to be able to actually talk to each other directly, something Asterisk doesn't necessarily know to either way NAT or not. |
| 17:51.34 | ChannelZ | * will stay in the media path if you ask it to do certain things like record the call or having to listen for inband DTMF |
| 17:51.43 | greenwolf | well isn't normally asterisk stay in the media path for music on hold and call features? |
| 17:52.11 | greenwolf | yes but normally and by default asterisk stays in the media path at all times |
| 17:52.17 | greenwolf | unless you tell it otherwise correct? |
| 17:53.25 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 17:53.57 | kuku | I have a lot of channels with "Rx: BYE" as the last message... and they are not going away. |
| 17:55.25 | kuku | 1.8.13.0 is the version |
| 17:56.19 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.238) |
| 17:56.49 | leifmadsen | they will go away eventually |
| 17:56.57 | kuku | they dont. |
| 17:57.01 | leifmadsen | you're talking about "sip show channels" I imagine |
| 17:57.08 | kuku | Yes. |
| 17:57.12 | leifmadsen | ignore that output basically |
| 17:57.16 | leifmadsen | core show channels is all you care about |
| 17:57.34 | greenwolf | i like both sip show and core show channels |
| 17:57.40 | kuku | I have 16 calls but I have 89 active sip dialogs |
| 17:57.40 | greenwolf | they each tell u different info |
| 17:57.44 | leifmadsen | unless you have hung channels from 'core show channels' the sip channel output will eventually clear itself up |
| 17:58.12 | kuku | They don't. And once there are about 150 active dialogs, it takes 30-60 seconds for the call to get sent to the trunk |
| 17:58.26 | greenwolf | i know this is a question that is shunned up and everyone hates it but i gotta ask |
| 17:58.56 | greenwolf | if i had a dual core processer XEON running 4GB ram how many active concurrent calls could i handle? |
| 17:59.03 | greenwolf | i know this is a question ppl hate when asking |
| 17:59.04 | leifmadsen | greenwolf: many |
| 17:59.09 | leifmadsen | because it doesn't have an answer |
| 17:59.16 | greenwolf | i mean roughly tho |
| 17:59.23 | leifmadsen | it is incredibly dependent on environment and what you're doing |
| 17:59.25 | leifmadsen | greenwolf: at least 10 |
| 17:59.28 | leifmadsen | probably up to 1000 |
| 17:59.28 | greenwolf | so i have an idea how many asterisk servers i should setup |
| 17:59.32 | leifmadsen | you have to test |
| 17:59.42 | leifmadsen | any answer someone gives you means absolutely nothing |
| 17:59.42 | greenwolf | reallly? asterisk can handle 1000 active sip calls at one time? |
| 17:59.47 | leifmadsen | sure |
| 17:59.52 | leifmadsen | depends what you're doing |
| 17:59.54 | greenwolf | deadlocks? |
| 17:59.55 | Qwell | only 1000? |
| 18:00.14 | leifmadsen | greenwolf: deadlocks are issues that are not dependent on number of channels |
| 18:00.30 | leifmadsen | more channels can certainly make it more obvious thougfh |
| 18:00.32 | greenwolf | someone who runs a different kind of software PBX told be that asterisk gets deadlocks and wouldn't be able to handle that many active channels |
| 18:00.43 | leifmadsen | but ya, people don't answer this question because it doesn't have any definitive answer |
| 18:00.43 | Qwell | greenwolf: He's also an idiot. |
| 18:00.51 | greenwolf | they said the way asterisk is built that it would be lucky to handle even 156 calls |
| 18:00.54 | leifmadsen | greenwolf: what Qwell said |
| 18:00.54 | greenwolf | so this is not true? |
| 18:00.55 | Qwell | greenwolf: That unnamed person. Who we all know. |
| 18:00.56 | leifmadsen | don't hire that guy |
| 18:01.03 | greenwolf | lol |
| 18:01.11 | leifmadsen | he's wrong |
| 18:01.14 | greenwolf | yea hes totally against asterisk and wanted me to come to freeswitch |
| 18:01.20 | leifmadsen | oh I know who that is |
| 18:01.24 | greenwolf | i told him no i love asterisk and have a passion for it |
| 18:01.25 | leifmadsen | he's an idiot |
| 18:01.41 | greenwolf | i love * way too much to leave plus i understand asterisk and its dialplan code |
| 18:01.55 | leifmadsen | you don't have to sell us on it |
| 18:01.58 | greenwolf | but honestly has anyone ever seen here 1 system handle over 1000 concurrent calls? |
| 18:02.21 | leifmadsen | greenwolf: yes |
| 18:02.27 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 18:02.29 | greenwolf | i just kinda need a general here because i have a big company that wants me to setup asterisk server for them and they have roughly 1000 internal extensions |
| 18:02.34 | leifmadsen | you have to test |
| 18:02.42 | greenwolf | and i needed to know how many asterisk servers i should run so i dont have any problems |
| 18:02.48 | greenwolf | i want to be safe you kno |
| 18:02.50 | leifmadsen | 1000 internal extensions doesn't correlate to any number of active channels |
| 18:02.54 | feeshon | Got SIP response 486 User Busy **IP of sip provider** |
| 18:03.03 | *** join/#asterisk carrar (tim@osburn.com) |
| 18:03.09 | greenwolf | i mean would 500 active outgoing calls at one time be alot for 1 asterisk server? |
| 18:03.11 | feeshon | In my asterisk console is this on the SIP providers end? |
| 18:03.16 | leifmadsen | greenwolf: then do your due diligence and test out the scenarios they are going to need, and find out how many machines you need |
| 18:03.18 | greenwolf | running XEON dual core at 4GB ram? |
| 18:03.49 | greenwolf | ok how do i test that many calls going thru my asterisk system without actually placing those calls? |
| 18:03.52 | greenwolf | is there a tool i can use? |
| 18:04.02 | leifmadsen | transcoding, recording, network capacity, conference rooms, paging, parking, transfers, etc all effect asterisk to various degrees |
| 18:04.04 | greenwolf | that can simulate a number of outgoing channels or calls |
| 18:04.07 | leifmadsen | sipp |
| 18:04.16 | greenwolf | its called sipp? |
| 18:04.20 | greenwolf | thats the program? |
| 18:04.24 | leifmadsen | no, I'm lying |
| 18:04.24 | carrar | opensource |
| 18:04.35 | carrar | http://sipp.sourceforge.net |
| 18:04.50 | greenwolf | oh wow nice |
| 18:04.51 | carrar | thats a fake URL too BTW |
| 18:05.03 | greenwolf | ok im gunna check this tool out i think this is exactly what im looking for |
| 18:05.12 | greenwolf | ok guys thanks im gunna go do some more research |
| 18:05.14 | leifmadsen | it is |
| 18:06.06 | greenwolf | but again do u think 1 asterisk server with nothing but outgoing active calls ,no conferencing, no parking, no ivr--none of that can handle about 500 calls? |
| 18:06.18 | feeshon | Can anyone help me with a dropped call issue? A called dropped and I luckily caught it in asterisk console. |
| 18:06.22 | leifmadsen | refuses to answer |
| 18:06.23 | greenwolf | is it possible and has it been done before? |
| 18:06.25 | feeshon | I got this error |
| 18:06.26 | greenwolf | come on |
| 18:06.31 | greenwolf | i need just a general here |
| 18:06.35 | leifmadsen | greenwolf: yes it has been done -- it has been done numerous times |
| 18:06.38 | greenwolf | i understand its a terrible question |
| 18:06.39 | leifmadsen | I told you the answer |
| 18:06.40 | feeshon | Got SIP response 486 User Busy |
| 18:06.42 | greenwolf | ok |
| 18:06.52 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 18:06.55 | carrar | greenwolf, why do you even need to run asterisk? |
| 18:07.07 | greenwolf | for a company PBX why else? |
| 18:07.08 | carrar | Just run a sip proxy |
| 18:07.16 | greenwolf | oh i see |
| 18:07.26 | greenwolf | well its needed as a gateway tho |
| 18:07.33 | greenwolf | how else am i gunna run outside sip trunks |
| 18:07.43 | leifmadsen | by using a sip proxy |
| 18:07.44 | carrar | rtp proxy? |
| 18:07.55 | greenwolf | no u need asterisk as a sip gateway |
| 18:08.01 | greenwolf | for the outside connectivity |
| 18:08.02 | leifmadsen | you do? huh |
| 18:08.05 | carrar | no, I don't |
| 18:08.15 | greenwolf | acoording to the ppl at opensips |
| 18:08.19 | leifmadsen | ok, this is getting kind of ridiculous, going to work :) |
| 18:08.29 | carrar | opensips and use rtpproxy or media proxy |
| 18:08.31 | carrar | can use |
| 18:08.40 | greenwolf | and i can link that to a sip provider |
| 18:08.46 | greenwolf | all without asterisk? |
| 18:08.57 | greenwolf | ok cool see i learn something new everyday |
| 18:08.59 | carrar | You need asterisk if you need a PBX |
| 18:09.01 | greenwolf | i see |
| 18:09.10 | kuku | leifmadsen: So those BYE messages don't go away but youre saying to ignore them even thought I have quality issues with the call when the number gets bigger? |
| 18:09.19 | feeshon | The fact that you want 500 outgoing calls on one sip server is worrysome, you may even max out the interface with all the over head if you don't have multiple |
| 18:09.24 | kuku | I recently switched from chanspy to bridge - and I'm thinking it has to do with that.. |
| 18:09.52 | greenwolf | ok i see |
| 18:09.59 | greenwolf | thanks guys for all your input here |
| 18:10.02 | greenwolf | greatly appreciated |
| 18:10.28 | greenwolf | i will take everything into consideration and do some more research..will come back later to explain what i did to achieve my goal |
| 18:10.30 | greenwolf | later guys |
| 18:10.51 | carrar | yeah, don't tell us now whatever you do!! :) |
| 18:11.20 | carrar | ASterisk is probably easier for him |
| 18:13.37 | navaismo | feeshon, User is busy |
| 18:13.51 | navaismo | thats not an error |
| 18:14.37 | Qwell | carrar: I, for one, will be on the edge of my seat awaiting his return. |
| 18:15.25 | navaismo | kuku, if you see calls in core show channels verbose that must be hanged at that time you can try the "hangup request <channel>" |
| 18:15.37 | kuku | wonders why a discission about someone who wants to put all the risk on one server deserves more attention then an actual issue with many channels not closing out... |
| 18:16.37 | kuku | navaismo: hangup is not a command available in the cli |
| 18:17.35 | saint_ | is there a way for a digium phone to be able to push the MSGS button and dial the voicemail without the voicemail system asking for mailbox number and password ? |
| 18:18.02 | *** join/#asterisk Dibbler (~Dibbler@host86-156-142-101.range86-156.btcentralplus.com) |
| 18:19.11 | feeshon | OK here is some better info I took out name/numbers |
| 18:19.15 | feeshon | http://pastiebin.com/?page=p&id=50ef060f5b262 |
| 18:20.40 | navaismo | kuku, which version of asteris are you using |
| 18:20.45 | kuku | 1.8.13 |
| 18:21.43 | navaismo | try with channel request hangup |
| 18:25.03 | navaismo | feeshon, user busy is not an error is a response, but you may check that the number exist and its not busy(in a call) then pb a SIP debug of the call |
| 18:25.34 | kuku | No such command 'channel request |
| 18:26.11 | feeshon | In the middle of a call it got a busy tone |
| 18:26.22 | feeshon | on our end and then he hung up and redialed the # |
| 18:26.43 | feeshon | If RTP traffic drops will it give you the busy tone? |
| 18:27.01 | feeshon | Seems like it hung up correctly |
| 18:28.56 | navaismo | kuku, you are executing that inside the asterisk cli right? |
| 18:29.41 | kuku | yes sir |
| 18:31.40 | [TK]D-Fender | feeshon, You don't have SIP debug enabled. We don't see if anything looks "proper" really. |
| 18:32.15 | feeshon | ok |
| 18:33.02 | navaismo | kuku, how do you installed asterisk? I have those commands in asterisk 1.8.11-cert6 |
| 18:34.33 | [TK]D-Fender | kuku, pastebin the actual attempts so we can see. |
| 18:34.48 | navaismo | core show help must show you the commands |
| 18:36.18 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 18:42.42 | *** join/#asterisk lorsungcu (~anonymous@h66-173-112-123.mntimn.dedicated.static.tds.net) |
| 18:42.52 | lorsungcu | what do |
| 18:43.03 | lorsungcu | having issues with NAT and a sonicwall |
| 18:43.18 | lorsungcu | anyone know anything about them? |
| 18:43.30 | lorsungcu | TZ205; inbound audio works, outbound does not. |
| 18:43.46 | lorsungcu | calls hang up after 30s |
| 18:43.48 | kuku | navaismo: I installed asterisk yes |
| 18:44.36 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:44.37 | kuku | aaa - I didn't specifyu the full command |
| 18:44.51 | lorsungcu | when calls are hung up, the other end is not disconnected |
| 18:46.35 | kuku | How do I aget the <channel>? sip show channels shows me call id |
| 18:47.51 | ghost75 | core show channels |
| 18:47.54 | [TK]D-Fender | lorsungcu, Disable all ALG options on it. Portforward as normal. Configure * as normal for functioning behind NAT |
| 18:48.13 | [TK]D-Fender | kuku, You don't |
| 18:48.22 | [TK]D-Fender | kuku, "core show channels concise". |
| 18:48.43 | [TK]D-Fender | kuku, ghost75's suggestion will get you truncated names which won't work. |
| 18:49.00 | [TK]D-Fender | kuku, As it's formatted. "concise" will give it in full |
| 18:50.30 | saint_ | Hey.. when I try to load res_digium_phone, I got the following error message in the console: Error loading module 'res_digium_phone.so': /usr/lib/asterisk/modules/res_digium_phone.so: undefined symbol: ast_msg_tech_register |
| 18:50.34 | saint_ | Can someone give me a hand with this ? |
| 18:54.11 | kuku | Thank you |
| 18:55.03 | navaismo | saint_, did you compiled with optimize memory? |
| 18:55.10 | ghost75 | thats strange, concise means like short in english or? |
| 18:55.16 | saint_ | navaismo: no |
| 18:55.28 | lorsungcu | yeah, tried that TK |
| 18:55.38 | lorsungcu | that's where I'm at now, with no outbound audio |
| 18:56.27 | [TK]D-Fender | lorsungcu, Perhaps you should confirm that it was done right |
| 18:57.43 | kuku | [TK]D-Fender: But that gets me active channels. Those are not the issues. |
| 18:58.13 | saint_ | BRB |
| 18:58.20 | [TK]D-Fender | kuku, which are? |
| 18:58.54 | [TK]D-Fender | kuku, because there is no "hengup" to unrelated SIP comms that aren't an actual active call channe;l |
| 18:59.40 | kuku | [TK]D-Fender: Then whey do the "BYE" channels stay up and not go away ? |
| 19:00.24 | [TK]D-Fender | kuku, AFAIK they linger for a little bit in case something else picks it up or it just expires, etc... |
| 19:00.39 | [TK]D-Fender | kuku, Do you have an actual problem due to this? |
| 19:01.14 | kuku | Yes - when there are too many of the bye channels ( like 100+ ) open, there is a delay in dialing out. |
| 19:01.23 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 19:02.47 | qakhan | i am using page app on asterisk, when i page someone and other person hangup call then my ext is still in use, it doesnt hangup |
| 19:03.22 | kuku | I have 12 active calls but 82 active sip dialogs - that is not normal |
| 19:03.41 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:03.41 | qakhan | here is my dialplan http://pastebin.com/DyDaqbgB |
| 19:03.53 | kuku | I'm using "Bridge" to play a message when the caller wants to play a message ( instead of saying the same things over and over 0 |
| 19:03.55 | kuku | I'm using "Bridge" to play a message when the caller wants to play a message ( instead of saying the same things over and over ) |
| 19:09.13 | *** join/#asterisk lorsungcu (~anonymous@h66-173-112-123.mntimn.dedicated.static.tds.net) |
| 19:09.48 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.4.238) |
| 19:10.18 | *** join/#asterisk lorsungcu (~anonymous@h66-173-112-123.mntimn.dedicated.static.tds.net) |
| 19:23.53 | *** join/#asterisk lorsungcu (~anonymous@h66-173-112-123.mntimn.dedicated.static.tds.net) |
| 19:29.48 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 19:32.11 | ghost75 | hey any1 using google voice with 1.8 ? |
| 19:37.38 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 19:39.27 | lorsungcu | TK, here's a pb of my sip.conf |
| 19:39.28 | lorsungcu | http://pastebin.com/Q4xPHKLj |
| 19:39.52 | lorsungcu | still no outbound audio |
| 19:42.39 | ghost75 | canreinvite is deprecated |
| 19:44.28 | Qwell | deprecated != removed |
| 19:44.47 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 19:46.55 | carrar | canhascatlasyereyes=YES |
| 19:47.38 | ghost75 | can haz cookie? |
| 19:51.46 | ghost75 | what dial prefix would you use for selecting different peer? |
| 20:11.28 | *** join/#asterisk ponyofdeath (~vladi@cpe-75-80-173-129.san.res.rr.com) |
| 20:14.04 | ponyofdeath | hi, if I have asterisk port 5060 open with iptables and allow established / related packets also have iptables modules nf_conntrack_sip nf_nat_sip loaded. Do I need to accept the RTP ports as well? |
| 20:15.28 | SuperNull | do you block all by default ? |
| 20:15.48 | *** join/#asterisk lorsungcu (~anonymous@h66-173-112-122.mntimn.dedicated.static.tds.net) |
| 20:16.13 | SuperNull | i dont believe conntrack does anything with iptables as far as adding allow rules using 'ALG' type methods. |
| 20:16.23 | SuperNull | RTP isn't nearly the problem tho that sip is. |
| 20:16.25 | Qwell | ugh |
| 20:16.28 | Qwell | disable those. |
| 20:16.31 | Qwell | Go away until you do. |
| 20:16.47 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.59.95) |
| 20:17.18 | ponyofdeath | SuperNull: yes I block all incoming packets unless allowed |
| 20:17.39 | SuperNull | be explicit with the range in the sip config and then unblock the whole range. |
| 20:17.50 | SuperNull | keeping 5060 udp secure is the key. |
| 20:18.02 | ponyofdeath | what do u mean keeping it secure |
| 20:18.50 | SuperNull | only worry really about firewalling 5060 |
| 20:19.04 | SuperNull | 5060 sets up the RTP stuffs anyway. so if it cant happen it wont get setup. |
| 20:23.09 | ponyofdeath | SuperNull: ok so i set an rtp range in rtp.conf and then allowed it with iptables rules |
| 20:23.24 | ponyofdeath | SuperNull: I am having issues with gtalk calls not coming in |
| 20:23.31 | ponyofdeath | after about an day or two |
| 20:23.59 | ponyofdeath | gtalk calls fail to be set up from my server |
| 20:25.24 | vlad_starkov | Question: Does anyone use OpenStack for VoIP services? |
| 20:27.42 | *** join/#asterisk mzb_ (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 20:32.11 | navaismo | ~book |
| 20:32.11 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:32.20 | ghost75 | (21:18:56) SuperNull: only worry really about firewalling 5060 <- what for |
| 20:32.22 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 20:32.26 | raden | hugs Katty |
| 20:33.15 | raden | on my aastra 480's i have a global extension, this will ring on line one if im on line one will ring line 2 etc..... |
| 20:33.35 | raden | i then have like 4 programmed as a different extension but if im on the line it goes right to VM |
| 20:33.47 | raden | is there a way that I can make like 4 go to 5 if it busy ? |
| 20:36.02 | *** join/#asterisk elico (~Thunderbi@bzq-79-183-197-51.red.bezeqint.net) |
| 20:37.31 | ghost75 | hunt group? |
| 20:39.19 | ChannelZ | Yes! let's kill some vermin |
| 20:43.37 | *** join/#asterisk rfnunezjr (~c0d3.d3m0@116.50.251.178) |
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| 20:51.48 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
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| 21:03.54 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 21:04.37 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
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| 21:13.16 | SuperNull | if i dont have a username set for a peer.. does it require a secret still (if secret is set) |
| 21:23.22 | *** part/#asterisk volga629 (~volga629@host7.pythian.com) |
| 21:26.58 | *** join/#asterisk lorsungcu (~anonymous@50.12.210.49) |
| 21:27.05 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 21:34.44 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
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| 21:44.20 | *** join/#asterisk lorsungcu_ (~anonymous@67-220-24-45.usiwireless.com) |
| 21:46.49 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 21:47.05 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 21:48.10 | rogers- | Registers are sent via port 5060 udp? |
| 21:49.08 | fprior | Hi all and happy 2k13. I need connect two locations, both have Asterisk behind NAT. What do you suggest: SIP or IAX trunking and why ? |
| 21:49.26 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 21:49.29 | kaldemar | rogers-: depends. by default, yes. |
| 21:52.13 | kaldemar | SuperNull: username and secret parameters are independent. |
| 21:52.17 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 21:53.59 | *** join/#asterisk blee (~blee@rrcs-24-227-58-250.se.biz.rr.com) |
| 22:00.37 | [TK]D-Fender | rogers-, They are sent *tp* 5060 |
| 22:00.50 | [TK]D-Fender | to |
| 22:06.14 | *** join/#asterisk blee (~blee@rrcs-24-227-58-250.se.biz.rr.com) |
| 22:13.10 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 22:15.59 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ocdnlqllyamdntrw) |
| 22:25.41 | rogers- | trying to register using BRIA iphone app to an asterisk box behind a firewall.. 5060 is forwarded, but no go |
| 22:25.56 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 23:01.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.23 | jacekowski | rogers-: tcpdump |
| 23:05.12 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:05.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 23:23.32 | phix | shouldvi use ael |
| 23:45.50 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:47.26 | *** join/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 23:47.44 | *** part/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 23:48.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.19.182) |
| 23:52.54 | *** join/#asterisk Kyosh (~whoa@pool-72-89-93-13.nycmny.fios.verizon.net) |
