| 00:29.43 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 00:30.07 | *** join/#asterisk war9407 (war@c-71-62-169-156.hsd1.va.comcast.net) |
| 00:38.11 | *** join/#asterisk jaybinks_ (~jaybinks@2402:c00:2:0:4a5b:39ff:fe45:8b8) |
| 00:45.24 | *** part/#asterisk nny (~Scott_2@cpe-174-107-201-103.sc.res.rr.com) |
| 00:53.34 | jaybinks_ | can someone give me a test call, to check my new pbx ... sip:asterisk-test@sip.itslenny.com:5060 |
| 00:57.22 | *** part/#asterisk k-man (~k-man@unaffiliated/k-man) |
| 01:00.14 | WiretapWork | jaybinks_, you'll need to provide an ISN or ENUM if you want most people here to bother :P |
| 01:03.48 | jaybinks_ | oh... how would you suggest I do that ? |
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| 01:10.16 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:25.53 | WiretapWork | jaybinks_, with the power of google? |
| 01:50.23 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 01:51.34 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 02:04.47 | *** join/#asterisk Shariff (~ask@zusjuh.xs4all.nl) |
| 02:04.49 | Shariff | Hi there |
| 02:06.10 | Shariff | My adsl modem is handling my home telephony. I would like to use asterisk for this, but the phones have a rj11 connection, and thus are connected to the modem. can I configure asterisk to handle those phones, even though I do not have the phones directly connected to the asterisk box? |
| 02:07.32 | WIMPy | Ask your provider for your account details. |
| 02:07.44 | WIMPy | But most won't tell you. |
| 02:07.48 | bdfoster | Shariff, are you using AT&T? |
| 02:08.12 | bdfoster | there are other ways of doing this as well, one of which is using an ATA and dropping service with your current provider |
| 02:08.15 | WIMPy | Maybe you can find some instruction with aunt google on how to extract them from the device. |
| 02:08.28 | Shariff | WIMPy: Actually I have my account details for my internet provider (also acts as a telephone service provider) |
| 02:08.38 | Shariff | bdfoster: Negative, I use a dutch isp |
| 02:08.51 | WIMPy | The SIP accounts, not your PPP acount. |
| 02:09.04 | bdfoster | yea, as long as it's sip |
| 02:09.14 | Shariff | WIMPy: Aye, I have my sip account details |
| 02:09.16 | WIMPy | Most probably. |
| 02:09.22 | bdfoster | ok, grab a few ata's |
| 02:09.42 | WIMPy | Ok, great. Then remove them from the IAD and put them in to sip.conf. |
| 02:09.45 | bdfoster | that'll take care of your house phones (you can use just one as well, but all are on the same extension |
| 02:09.46 | Shariff | bdfoster: doesn't my current adsl modem act as one? (fritz box 7340) |
| 02:09.58 | bdfoster | you'll probably not be able to use it |
| 02:10.18 | WIMPy | Ah, a FB, they are easy to manage. |
| 02:10.22 | bdfoster | plus ata's are so cheap you might as well grab a grandstream HT286 or SPA3102 |
| 02:10.32 | WIMPy | Well. Compared to if you don;t have access, that is. |
| 02:11.08 | WIMPy | Not that I'm a fan of AVM, but the FB might be the better choice. |
| 02:11.25 | Shariff | bdfoster: most likely I will not be able to manage my home phones using asterisk while the phones are connected to the fritzbox? |
| 02:11.43 | bdfoster | correct but I'm not familiar enough with fritzbox |
| 02:11.55 | bdfoster | usually it's not accessible |
| 02:12.06 | Shariff | WIMPy: Could you point me to some resources, how to go about this? I'm extremely new ;) |
| 02:12.11 | WIMPy | I don't see why that shouldn't work. |
| 02:12.14 | bdfoster | easy solution: buy a $30 HT286 |
| 02:12.28 | Shariff | bdfoster: checking into that now |
| 02:12.54 | bdfoster | I use them, I have subscribers of mine who use them, and I don't get complaints |
| 02:12.56 | WIMPy | The only cahallange I see is the usual NAT stuuf and not letting the FB interfere with Asterisks external communication. |
| 02:13.28 | Shariff | nods |
| 02:13.42 | WIMPy | Shariff: As I already wrote: You remove your providers SIP accounts from the FB and set them up in Asterisk instead. |
| 02:13.59 | Shariff | Isn't it also a bit silly.. the modem gets the calls (is capable of handling them) forwards them to asterisk, who in turn connects to the internet using the same modem? |
| 02:14.08 | WIMPy | You might have to use a non-standard port, I'm not sure on that, |
| 02:14.39 | WIMPy | Then you can set up local Accounts in Asterisk to connect the FB to. |
| 02:14.48 | Shariff | WIMPy: That would get the SIP working on the asterisk box... but what about phone control? they are physically connected to the FB, or am I misunderstanding you? |
| 02:15.08 | WIMPy | But you could also do ith the other way raund, and let th FB stay at the front and connect Asterisk via SIP to the FB. |
| 02:15.53 | Shariff | WIMPy: With that, the best way to go is probably to physically connect the phones to the asterisk box right? |
| 02:16.13 | WIMPy | What kind of phones? |
| 02:16.29 | Shariff | The ones now connected to the FB (home analog phones) |
| 02:16.44 | WIMPy | If you have hardware to do so, yes. |
| 02:16.49 | Shariff | nods |
| 02:16.58 | bdfoster | (which is expensive) |
| 02:17.33 | bdfoster | and the reason why you use ATA's |
| 02:17.35 | Shariff | If I do want to go down that road.. hooking up the phones to the asterisk box.. what would be the best pci card to do that? just for home use that is.. I've seen a LONG list of pci adapters.. so it's a kinda tree/forest thing? |
| 02:17.54 | Shariff | Ahh.. ata's solve the need for a pci-adapter, in simple usages? |
| 02:18.00 | bdfoster | yes |
| 02:18.07 | bdfoster | you dont want a pci card |
| 02:18.17 | bdfoster | you want an ATA |
| 02:18.18 | WIMPy | Your FB is (amongs others) an ATA. |
| 02:18.29 | bdfoster | correct ^^ what he said |
| 02:19.29 | Shariff | is processing info :) |
| 02:20.32 | bdfoster | ATA = analog telephone adapter |
| 02:20.36 | bdfoster | fwiw |
| 02:20.36 | Shariff | Do you know of any resources that might explain my case: managing my home analog phones trhough an ata ? |
| 02:20.51 | bdfoster | you need to be more specific |
| 02:21.01 | WIMPy | I'm sure the voip related forums are full of that. |
| 02:21.05 | Shariff | Sorry, I'm trying to be.. this is just very new to me ) |
| 02:21.23 | WIMPy | But I'm not so sure about the quality of the information found there. |
| 02:21.38 | Shariff | Basically, how I would need to set up asterisk in order to use the phones connected to an ata, and not to the asterisk box itself |
| 02:21.50 | bdfoster | are you talking about configuring the ATA to use asterisk? |
| 02:22.16 | bdfoster | Shariff, you're confusing yourself as well as all of us |
| 02:22.33 | Shariff | I'm really sorry, I'm not trying to confuse you :) |
| 02:22.52 | bdfoster | it's alright, just trying to translate |
| 02:22.56 | Shariff | :D |
| 02:23.27 | Shariff | I'm picturing the following setup: asterisk box <-> FritzBox Modem <-> home analog phone |
| 02:23.28 | WIMPy | You just divert the connection from the FB to your provider through Asterisk. |
| 02:23.48 | Shariff | I am looking for a resource describing that situation, to configure asterisk to use the home analog phone |
| 02:24.02 | bdfoster | ok the ATA is just going to be on your local network |
| 02:24.09 | Shariff | nods |
| 02:24.31 | Shariff | Ahhh and not connected to the internet (directly)? |
| 02:24.39 | bdfoster | so configuring the ATA to connect to * is about as trivial as connecting a softphone to * |
| 02:24.42 | bdfoster | correct |
| 02:24.50 | WIMPy | So instead of provider-FB-phone you go provider-Asterisk-FB-phone. |
| 02:25.05 | bdfoster | ^^what he said |
| 02:25.33 | Shariff | It's dawning :D |
| 02:25.45 | Shariff | Thanks a lot for the help! |
| 02:25.55 | bdfoster | there are some settings you will have to look up depending on the model you choose and that has to do with impedence and a few other settings since you are not in the USA |
| 02:26.04 | bdfoster | most are set up for NA settings |
| 02:26.24 | Shariff | NA? |
| 02:26.27 | bdfoster | also depends on the phones you use, you may not even have to do that |
| 02:26.31 | bdfoster | NA = north america |
| 02:26.37 | Shariff | Ahh ok |
| 02:26.49 | WIMPy | huh? He won't be changing anything at that end. |
| 02:26.57 | bdfoster | yea I just realized that lol |
| 02:27.08 | Shariff | So I can disregard that? :D |
| 02:27.22 | bdfoster | disregard the impedence and other settings |
| 02:27.38 | Shariff | ok then :) |
| 02:27.43 | bdfoster | that's if you are going to connect your analog land line to asterisk |
| 02:27.50 | bdfoster | but you dont have one so no need |
| 02:28.09 | bdfoster | Ive got a 3102 that I use for that, has both FXS and FXO |
| 02:28.24 | bdfoster | and the handytone but yea |
| 02:28.45 | Shariff | bdfoster: Nice.. I believe my FB also has FXS and FXO ports.. 2 FXS I think.. |
| 02:29.03 | bdfoster | prolly only 2 FXS |
| 02:29.09 | bdfoster | line one, line two |
| 02:29.13 | Shariff | yes! |
| 02:29.18 | Shariff | You know your stuff :D |
| 02:29.29 | bdfoster | I try lol |
| 02:29.42 | WIMPy | Usually 1 FXO and 2 (or seldom 3) FXS plus 1 or 2 S0. |
| 02:29.59 | bdfoster | with ATA's on an ADSL modem? |
| 02:29.59 | Shariff | nods |
| 02:30.06 | bdfoster | that's new |
| 02:30.07 | bdfoster | ... |
| 02:30.13 | WIMPy | Only the very latest models come without line interfaces. |
| 02:30.14 | Shariff | What's s0? |
| 02:30.28 | WIMPy | ISDN BRI |
| 02:30.33 | Shariff | Ahh duh |
| 02:30.42 | bdfoster | I'm in USA though, wacky europeans lol |
| 02:31.01 | Shariff | lol |
| 02:31.27 | bdfoster | ive got an AT&T VDSL modem/router/ATA here |
| 02:31.27 | Shariff | Thanks a lot for the help and patience.. sorry for the confusion :D |
| 02:31.28 | WIMPy | Ja, we already found out the hard way that we actually didn't want to go where you want to go :-) |
| 02:31.36 | bdfoster | hehe |
| 02:31.49 | bdfoster | yea well I can understand I used to live in Italy |
| 02:32.12 | Shariff | WIMPy: translating: you learned the hard way not to go where I am trying to go? |
| 02:32.28 | WIMPy | No, where you are. |
| 02:32.34 | Shariff | Ahh |
| 02:32.48 | bdfoster | I'm so confused |
| 02:32.49 | Shariff | Oh dear lord.. I just noticed the time.. no wonder my brain works at half speed |
| 02:32.57 | Shariff | it's 4.30 am |
| 02:33.01 | WIMPy | :-) |
| 02:33.21 | Shariff | Thanks and sleep well (when you get there) :D |
| 02:33.31 | bdfoster | Shariff, anyway, hope everything works out. come back when you get somewhere |
| 02:33.42 | Shariff | I will thanks! |
| 02:33.45 | Shariff | bye bye |
| 02:33.48 | bdfoster | cya |
| 02:34.09 | WIMPy | Lucky guy for having his account details. |
| 02:34.11 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:35.00 | bdfoster | yea, I can't get anything from AT&T, so I have to use the 3102 |
| 02:35.03 | bdfoster | which sucks big time |
| 02:35.22 | bdfoster | I'm gonna drop them as soon as I get the number ported |
| 02:35.28 | WIMPy | Same problem here. And it sucks big time. |
| 02:35.56 | bdfoster | funny thing is I have 9 trunks |
| 02:36.36 | bdfoster | but that damn landline is a) easy to remember and b) has been my phone number for too long to let it just go |
| 02:42.26 | WIMPy | I guess I should get a real land line again. |
| 02:43.35 | WIMPy | But that won't mix very well with my internet needs, tariff wise. :-( |
| 02:44.53 | bdfoster | only reason why I would need a landline anymore is a backup |
| 02:45.19 | bdfoster | even then, ill probably end up using GSM before I get another landline |
| 02:45.23 | WIMPy | I'd like something that just works. |
| 02:53.22 | bdfoster | I like problems, they keep me in business ;-) |
| 02:59.32 | ChannelZ | please swipe your credit card at the door |
| 03:01.47 | ketas | hmm |
| 03:01.57 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 03:03.37 | ketas | are they switching off analog lines there too? |
| 03:04.19 | WIMPy | guesses that happens everywhere. |
| 03:04.54 | ketas | voip over adsl2+ here |
| 03:05.16 | WIMPy | Same here. |
| 03:05.28 | WIMPy | But the cheapest possible IAD, off course. |
| 03:05.55 | WIMPy | It's not even up to what's in the contract :-( |
| 03:06.06 | ketas | well not exactly _here_ since i haven't asked for it |
| 03:06.11 | ketas | should maybe |
| 03:06.35 | bdfoster | u can get a real analog line here still |
| 03:06.37 | ketas | more asterisk hacking :P |
| 03:06.44 | bdfoster | most do |
| 03:07.31 | WIMPy | Only from the ex monopolist here. |
| 03:08.21 | WIMPy | I think none of the others have ever done analog. |
| 03:08.53 | ketas | here my isp provides iptv, i guess that all who want anything changed are switched to voip |
| 03:09.45 | WIMPy | That's again something I should have, but don't get. |
| 03:09.52 | WIMPy | But then, I'm not really interested. |
| 03:10.19 | bdfoster | yea iptv here |
| 03:10.38 | bdfoster | better than dish... |
| 03:10.45 | ketas | like when you have phone and get iptv+phone, your phone will be voip |
| 03:10.54 | bdfoster | same here |
| 03:10.57 | WIMPy | Interestinly enough iptv is only available where real phone lines were available. |
| 03:11.19 | bdfoster | real lines still availabe here |
| 03:11.25 | ketas | this is private house so only copper lines |
| 03:11.39 | ketas | no overhead fiber planned in next years |
| 03:11.49 | ketas | overhead fiber is pita anyway |
| 03:11.50 | WIMPy | From the carrier I'm with, they have never been available im my area. |
| 03:14.10 | *** join/#asterisk SithRee (~Sith@64-121-106-14.c3-0.smt-ubr1.atw-smt.pa.cable.rcn.com) |
| 03:14.41 | ketas | not really going to switch places because here it's only 16/1 and there, where people are packed tightly together it will be 100/20 |
| 03:15.19 | WIMPy | I'm just 50m away from VDSL. |
| 03:15.36 | WIMPy | The other end of the yard can get it. |
| 03:16.02 | ketas | hah! |
| 03:16.09 | WIMPy | But I'm not wired up to the same MDF, which is 750m away, but another one in the opposite direction which is 1200m away. Suxx. |
| 03:16.27 | ketas | workaround? |
| 03:16.29 | ketas | :p |
| 03:16.55 | WIMPy | Ja, I thought about putting a wire across the yard. |
| 03:17.43 | ketas | no vdsl here yet |
| 03:18.07 | ketas | i should be 500m away |
| 03:18.34 | ketas | not sure about length of wires |
| 03:39.16 | SithRee | I'm exploring the possibility of using asterisk for a small 5 user office, with I think 1 analog line, possibly 2 |
| 03:40.03 | SithRee | what hardware would I need to build a pbx/vm machine? |
| 03:41.35 | bdfoster | if they are really stuck on using analog lines, it will cost them |
| 03:41.58 | bdfoster | to me, it's not worth the cost, imho |
| 03:42.36 | pigpen | A small pc with an audiocodes MP-112 fxo/fxs sip box |
| 03:42.47 | bdfoster | ew |
| 03:42.49 | pigpen | use a polycom phone of your choice |
| 03:43.06 | bdfoster | from what I heard quality sucks on those |
| 03:43.15 | bdfoster | I would only use digium or sangoma cards |
| 03:43.26 | pigpen | cool. |
| 03:43.46 | pigpen | I have and I am running many digium cards. FXO/FXS and many PRI cards. |
| 03:43.48 | ketas | only analog? |
| 03:43.55 | ketas | no voip there? |
| 03:44.05 | pigpen | I have also deployed about 50 audiocodes....and have another 300 or so to go. |
| 03:44.10 | pigpen | quality is fine. |
| 03:44.13 | pigpen | no problems. |
| 03:44.17 | ChannelZ | ain't the MP-112 FXS? |
| 03:44.32 | pigpen | I think it is a dual...or maybe I am thinking of the MP114 |
| 03:44.56 | ChannelZ | hmm interesting |
| 03:44.57 | pigpen | I just deployed one...I'll look at the invoice. |
| 03:45.00 | ChannelZ | maybe it is dual |
| 03:45.45 | bdfoster | anyway, yea need ability to accept 2 FXO, then ip phones of your choice |
| 03:45.59 | pigpen | sorry: AudioCodes - MediaPack 114 - 2FXS, 2FXO (MP114/2S/2O/SIP) |
| 03:46.06 | SithRee | I need to check w/ the ISP about VoIP, so for now, just analog |
| 03:46.26 | pigpen | the only problem I have with audiocodes is disconnection. |
| 03:46.30 | bdfoster | SithRee, it would save you a bit if you can get away with voip |
| 03:46.47 | bdfoster | reliability goes down unless there's a backup |
| 03:46.48 | pigpen | it happens in certain areas. |
| 03:47.03 | pigpen | I am fighting three right now in San Antonio |
| 03:47.28 | pigpen | something about the switches in SA that disconnect doesn't detect the call drop. |
| 03:47.32 | bdfoster | yea I just can't do landlines anymore |
| 03:47.41 | bdfoster | I dont recommend them to my clients |
| 03:47.51 | bdfoster | and yes, I know that makes me sound like an idiot |
| 03:47.53 | bdfoster | lol |
| 03:48.01 | pigpen | bdfoster, I am all for PRI's.... :-) |
| 03:48.25 | bdfoster | yea I dont deal with anything needing something like that lol |
| 03:48.36 | bdfoster | anyone** |
| 03:48.40 | *** join/#asterisk tengulre (7d47d010@gateway/web/freenode/ip.125.71.208.16) |
| 03:49.03 | pigpen | but hopefully my latest purchase will help me figure out what dam ATT is doing.... Fluke TS52 Pro |
| 03:49.17 | bdfoster | ohh nice |
| 03:49.27 | pigpen | yeah, for $315 it better be. |
| 03:49.29 | joshaidan | pigpen: something not detecting the voltage drop? |
| 03:49.48 | pigpen | yeah. Audiocodes MP114 4 port FXO. |
| 03:50.04 | pigpen | I figure if I can figure one out, I can fix the other two locations. |
| 03:50.11 | ketas | pigpen: i thought you meant 1000V insulation tester at first... |
| 03:50.19 | ketas | surely that too can help att |
| 03:50.22 | ketas | :P |
| 03:50.27 | joshaidan | hmm... |
| 03:50.37 | pigpen | I was balancing a multimeter, holding the leads, a cell phone and watching the voltage and I saw the drop. |
| 03:50.40 | *** join/#asterisk radic (~radic@dslb-094-216-239-027.pools.arcor-ip.net) |
| 03:50.45 | pigpen | but it only dropped to about 1.2V |
| 03:50.53 | joshaidan | I seem to recall having a similar issue years ago with an audiocode where we'd get VM messages with dialtones on them |
| 03:50.55 | pigpen | I was thinking I would see 0V |
| 03:51.03 | pigpen | joshaidan, right. |
| 03:51.19 | joshaidan | I'll see if my brain remembers what I did to fix it |
| 03:51.23 | bdfoster | joshaidan, I get that on my spa3102 |
| 03:51.41 | bdfoster | user error maybe, or damn ATT ata sucks |
| 03:51.55 | pigpen | I have tried polarity reversal ( I know now it is not doing this), current disconnect, you name it, nothing works by itself or in combo. |
| 03:52.23 | joshaidan | Can you adjust the length of time of the voltage drop? |
| 03:52.43 | pigpen | I setup an absolute time out just in case, and it takes care of 99% of the issues, just long vm's. |
| 03:52.51 | pigpen | joshaidan, I didn't see any... |
| 03:53.02 | pigpen | device is rebooing. |
| 03:53.07 | pigpen | s/rebooing/rebooting |
| 03:53.52 | joshaidan | I remember once we delivered some POTS lines to a customer's PBX using an Adtran TA90x. Their PBX wouldn't drop the call because we weren't sending the disconnect (voltage drop). |
| 03:54.28 | WIMPy | Analog is evil! |
| 03:54.29 | pigpen | yeah, in call is like 48v, then it drops. |
| 03:54.34 | pigpen | WIMPy, yes, it is. |
| 03:54.42 | joshaidan | That took a long time to resolve because I thought it was a setting on the adtran when it turned out to be an MGCP field wasn't being sent to signal the disconnect. |
| 03:54.55 | ketas | analog lines are just too crappy to debug |
| 03:55.23 | bdfoster | which is why bdfoster likes voip lol |
| 03:55.32 | bdfoster | one reason anyway |
| 03:55.47 | WiretapWork | debugging analogue lines requires a good ear |
| 03:55.53 | ketas | :P |
| 03:55.56 | pigpen | I think that testing analog lines takes knowledge of the local telco, the right test equipment (I hope), and the equipment conf'd correctly....in therory. |
| 03:55.57 | joshaidan | Our problem is that we think too much in terms of protocols rather than electricity and physics :) |
| 03:56.01 | bdfoster | debugging voip requires good eyes |
| 03:56.25 | WiretapWork | I have been tshooting analogue lines a lot longer than I have voip :P |
| 03:56.29 | ketas | voip is pretty high layer too |
| 03:56.41 | bdfoster | and I'm still young |
| 03:56.53 | bdfoster | so, no, don't really deal much with analog |
| 03:57.00 | pigpen | hmm..I see "Reorder Tone Duration" in seconds, set to 255 |
| 03:57.24 | joshaidan | I don't think that would be the voltage drop setting you're looking for |
| 03:57.36 | pigpen | yeah, I think that is for overseas. |
| 03:58.02 | pigpen | I wonder if "Answer Supervision" has anything to do with "Disconnect Supervision".....bad translation? |
| 03:58.03 | joshaidan | I think that signals the "your call failed now hangup the stupid phone" sound you here |
| 03:58.13 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-xmitogejitmcyvjl) |
| 03:58.13 | joshaidan | Disconnect supervision may be it |
| 03:58.59 | pigpen | oddly enough, anything with "Disconnect and Answer Supervision" is under "Sip Advanced Parameters" |
| 04:00.23 | pigpen | yeah, nothing about disconnect time. |
| 04:00.46 | ketas | adsl link to moon |
| 04:02.52 | pigpen | <PROTECTED> |
| 04:04.00 | ketas | nah, quote from fbsd ipfw manpage |
| 04:04.05 | ketas | somehow came up |
| 04:08.14 | pigpen | CurrentDisconnectDuration |
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| 04:12.07 | pigpen | but, found this in an old post, cannot find it in the newer firmware |
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| 06:25.35 | wdoekes2 | good morning |
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| 06:56.26 | dinesh___ | hi all, it seems that my mobile provider is blocking data from my mobile -> asterisk if I do a Dial() forwared prior to Answer() the call |
| 06:56.51 | dinesh___ | i tried with mobile provider and the exact same extension actually worked |
| 06:57.07 | dinesh___ | I read that Dial() would automatically answer the channel |
| 06:57.35 | dinesh___ | but i would actually like to try a kind of "background Dial()", that would not send anything to the caller |
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| 06:58.08 | kaldemar | dinesh___: you read wrong. Dial does not answer. |
| 06:58.54 | dinesh___ | well it's on http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 06:59.07 | dinesh___ | "The originating channel that triggered this Dial command is then Answered, if necessary, and the two channels are connected together ("bridged") allowing a conversation to take place between them. " |
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| 06:59.51 | dinesh___ | i'll try to find the authoritative doc |
| 07:00.31 | dinesh___ | https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 07:00.32 | kaldemar | dinesh___: the previous sentence is relevant. |
| 07:00.56 | dinesh___ | "As soon as one of the requested channels answers, the originating channel will be answered, if it has not already been answered." |
| 07:01.02 | kaldemar | "The first channel that answers "wins", and all the other outgoing channels are hung up." <--- |
| 07:01.24 | dinesh___ | yep, but in my case i'm only calling a single channel |
| 07:01.40 | dinesh___ | no parallel ring |
| 07:01.55 | kaldemar | when the destination channel answers, the caller is noticed and the channels are bridged. |
| 07:02.24 | kaldemar | if the destination channel does not answer, the calling channel does not get an answer indication. |
| 07:04.13 | dinesh___ | yep right, but actually my problem is that with Swisscom (mobile -> asterisk does not let any audio go through), this bridge is only one-way, wheras with Orange it works in both ways |
| 07:04.54 | kaldemar | is early media what you're trying to achieve? |
| 07:05.40 | kaldemar | early media as in the caller gets audio from asterisk before the call is answered? |
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| 07:09.48 | dinesh___ | well what i am actually trying to achieve is a simple call redirection |
| 07:10.45 | dinesh___ | I call number A (which is my local SIP landline number), which then in turn calls number B through a cheap SIP provider |
| 07:11.16 | kaldemar | and the issue is? |
| 07:11.59 | dinesh___ | the issue is that, with one mobile provider it works perfectly fine, but with another one, I can only hear what B says, but B can't hear me |
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| 07:12.42 | kaldemar | use directmedia=no for the peers in sip.conf. |
| 07:17.22 | Russ | directmedia is teh awesome! |
| 07:17.27 | Russ | oh, wait |
| 07:17.32 | Russ | I'm thinking of earlymedia |
| 07:21.14 | sxpert | that's because of stupid NATs all over |
| 07:21.42 | sxpert | also, directmedia=yes tends to make police listen-in systems less efficient ^^ |
| 07:24.59 | kaldemar | sends sxpert some foil |
| 07:25.23 | cneb3000 | good morning sunshine |
| 07:25.44 | Russ | thats why I use IAX2, that way the police aren't put through any undue stress or trouble and can have time to make a donut run |
| 07:25.59 | Russ | police who have time to make donut runs are in better moods |
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| 08:12.59 | dinesh___ | thanks kaldemar, directmedia=no fixed the issue |
| 08:14.08 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 08:14.11 | Polysics | hello |
| 08:14.32 | irroot | TLS + SRTP is also police friendly ;) |
| 08:14.43 | Polysics | SIP call goes up, asterisk says packet2packet bridging up, no other warnings, but no audio on both sides |
| 08:14.49 | Polysics | what could i start looking at? |
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| 08:15.34 | cneb3000 | Polysics: What did each side negotiate their codecs as? |
| 08:15.53 | Polysics | cneb3000, that's the information i wanted to look for - where is it, please? |
| 08:16.02 | Polysics | sip debug? asterisk is 1.6.2 |
| 08:16.25 | cneb3000 | personally i'd run a wireshark trace. but i think sip debug might contain it. |
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| 08:17.02 | cneb3000 | benefit of a wireshark trace is it will capture the RTP stream (the audio bit) so you can play it back if there is any |
| 08:17.03 | kaldemar | matter of preference, sip debug will show the SIP headers and SDP too. |
| 08:17.12 | Polysics | i do not have access to the side that has problems (100 km away) - will try with sip debug |
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| 08:17.48 | Polysics | i never got how/if it is possible to get the debug for a single call separately |
| 08:18.21 | kaldemar | Polysics: enable debug by ip or by peer |
| 08:18.45 | kaldemar | Polysics: write sip set debug and hit tab, you'll get the options |
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| 08:22.23 | dinesh___ | when can I expect to see the dialplan function ACG() (http://www.asterisk.org/docs/asterisk/trunk/functions/agc) available in AsteriskNOW ? |
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| 08:32.19 | kaldemar | dinesh___: when asterisknow has asterisk 1.6.1 or newer. don't know what 1.6.X branch it has at the moment, but i'd be suprised if it was not 1.6.2. |
| 08:33.14 | kaldemar | dinesh___: also assuming that asterisknow istallation has func_speex.so which provides AGC. |
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| 08:40.26 | Polysics | if two clients can't agree on any codec, does the call still get bridged? |
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| 08:40.40 | Polysics | i got 15k lines of sip debug to sift through :-D |
| 08:41.22 | cneb3000 | it does still bridge, but just no media |
| 08:42.22 | WIMPy | So what gets bridged then, if it's not media? |
| 08:51.12 | cneb3000 | the two streams of media :) |
| 08:51.20 | cneb3000 | so.. lets say one side wants g729 the other g711 |
| 08:51.27 | cneb3000 | eventually they wont agree - but they just send it anyway |
| 08:51.31 | cneb3000 | they just wont hear each other |
| 08:53.49 | irroot | loving this "micro" machines small atom box with asterisk |
| 08:56.52 | Polysics | so that is probably the problem i have here |
| 08:57.00 | Polysics | ok, i sent the sip debug to the experts |
| 08:57.03 | Polysics | i will read it later too |
| 08:57.13 | Polysics | but i still do not know what i am looing for :-D |
| 08:59.01 | cneb3000 | honestly... Wireshark is the only way |
| 08:59.02 | cneb3000 | http://www.linuxjournal.com/article/9398 |
| 08:59.03 | cneb3000 | ;) |
| 08:59.39 | cneb3000 | although, i'm sure a lot of people will favour the text based logging of SIP in asterisk, you can make it visual in wireshark. just makes it quicker me thinks,. |
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| 09:15.08 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 09:15.26 | nicola_pav | hello. i have a lots of those messages: Unknown RTP codec 126 received from '0.0.0.0 |
| 09:15.34 | nicola_pav | how do i stop them please? |
| 09:15.49 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 09:16.12 | cneb3000 | nicola_pav: which version of Asterisk are you running? |
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| 09:16.23 | nicola_pav | cneb3000: 1.4 |
| 09:17.25 | cneb3000 | to anyone smarter than me... wasn't that a bug in 1.4 |
| 09:17.26 | cneb3000 | ^^^ |
| 09:17.55 | cneb3000 | "recieving unkown RTP Codec from unkown soures' |
| 09:21.09 | nicola_pav | cneb3000: so its a bug? |
| 09:21.24 | cneb3000 | nicola_pav: Maybe.. see --> http://lists.digium.com/pipermail/asterisk-bugs/2010-August/085286.html |
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| 09:24.38 | nicola_pav | cneb3000: ok, thanks a lot |
| 09:25.11 | cneb3000 | nicola_pav: No problem! try upgrading asterisk to a newer build of 1.4 (1.4.11.5?) and see if it helps |
| 09:25.47 | cneb3000 | sorry that was dumb - 1.4.41* |
| 09:25.55 | cneb3000 | face palm |
| 09:26.23 | nicola_pav | i will try and see how it goes |
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| 09:26.35 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:26.38 | schmidts | hello |
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| 09:29.53 | cneb3000 | howdy schmidts |
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| 09:56.33 | *** join/#asterisk CareBear\ (peter@stuge.se) |
| 09:56.51 | CareBear\ | hi all! is there some nice way to add extremely simple messaging into asterisk? |
| 09:56.53 | CareBear\ | over SIP |
| 09:56.57 | CareBear\ | I am thinking |
| 09:57.32 | CareBear\ | or, well, to begin with I just want to parse out a particular header from a server that my asterisk registers with |
| 09:57.48 | CareBear\ | passing that on will be a later issue |
| 09:59.26 | kaldemar | asterisk doesn't support SIP messaging outside a call yet. |
| 10:01.31 | CareBear\ | okey |
| 10:01.41 | CareBear\ | first step - grabbing the special header? |
| 10:03.00 | kaldemar | modify chan_sip.c if you're talking about a registration dialog :P |
| 10:03.23 | kaldemar | or set up a proxy in between. |
| 10:03.32 | CareBear\ | yes it might be that it only comes back on REGISTER replies |
| 10:03.35 | CareBear\ | hm proxy |
| 10:03.37 | CareBear\ | like what? |
| 10:04.10 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-195-199-7.zone9.bethere.co.uk) |
| 10:06.51 | kaldemar | like kamailio or opensips |
| 10:07.02 | CareBear\ | I talked to kamailio at LinuxTag |
| 10:07.07 | CareBear\ | well, people from :) |
| 10:07.18 | CareBear\ | it seems like not the right thing |
| 10:07.23 | CareBear\ | how thin is opensips? |
| 10:08.13 | kaldemar | opensips is another fork of openser, like kamailio. |
| 10:08.21 | kaldemar | take a peek at this: http://images.tmcnet.com/expo/astricon/presentations/thurs/Thursday_CiraB_S3.pdf |
| 10:09.19 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:09.27 | CareBear\ | .pdf sounds like not so thin :) |
| 10:09.44 | CareBear\ | I was rather thinking 250 lines of C code |
| 10:09.45 | CareBear\ | ;) |
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| 10:14.02 | cneb3000 | CareBear: Why are you worried about it being thing? OpenSIPS is pretty capable ! |
| 10:16.08 | cneb3000 | thing..? thin* :) |
| 10:16.30 | CareBear\ | I don't want another service |
| 10:16.39 | CareBear\ | I just want to tap this one header |
| 10:19.46 | kaldemar | what do you want to do with the information in the header? |
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| 10:23.27 | cneb3000 | requested a fax for asterisk license twice now and it's not been emailed through >_< |
| 10:23.38 | kaldemar | reading a header would be quite a small change to handle_response_register in chan_sip.c |
| 10:36.16 | CareBear\ | kaldemar : sounds perfect. |
| 10:36.54 | CareBear\ | is there also already some common outlet for $stuff that I could use? |
| 10:37.20 | CareBear\ | and which is a bit configurable so that I can get this particular $stuff into a particular $outlet? |
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| 10:59.19 | cneb3000 | Is fax a big deal is the USA? Do businesses actually use it much? |
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| 11:08.13 | Chainsaw | cneb3000: Finance departments like faxes, yes. |
| 11:08.38 | Chainsaw | cneb3000: Because you automatically have an indisputable written record that your message was printed on the other end. |
| 11:08.57 | Chainsaw | cneb3000: It's as good as a registered letter, and you can call in an expert witness from say... Canon, to authenticate the TX report. |
| 11:09.41 | cneb3000 | I see! |
| 11:09.45 | cneb3000 | Thanks |
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| 11:11.25 | Chainsaw | cneb3000: Also, pardon me for being blunt, accountants hate change. So your best bet is to get T38 passthrough working so that the existing fax machine coexists with VoIP, rather then implementing it as an e-mail gateway. |
| 11:12.50 | cneb3000 | that was the plan! |
| 11:13.08 | *** join/#asterisk devyll (~devyll@thpallady.net.hostway.ro) |
| 11:13.44 | CareBear\ | In Germany you must send fax to register a domain name under .de. |
| 11:14.15 | irroot | cneb3000 you welcome to try the T.38 gateway im working on |
| 11:14.31 | cneb3000 | if you want me to give it a 'test run' im more than happy? |
| 11:16.21 | irroot | svn co http://svn.digium.com/svn/asterisk/team/irroot/t38gateway-1.8 |
| 11:16.35 | cneb3000 | I'm on 1.8.4, thats fine right? |
| 11:16.53 | irroot | 1.8.4 has some bugs in it still |
| 11:17.30 | irroot | 1.8.5-rc1 will have the fixes |
| 11:19.18 | irroot | the issues #19251 and #19521 are required |
| 11:19.48 | irroot | the issues #19251 and #19215 are required |
| 11:19.53 | irroot | typo sorry |
| 11:20.21 | devyll | hello guys. suddently randome ongoing colls are interrupted due to strange errors (notice: pri got event: HDLC Bad FCS; and Warning: Yellow alarm on channel 1); extensive logs at http://asterisk.pastebin.ca/2069378; does anybody have any ideeas ? |
| 11:20.36 | cneb3000 | irroot: I'll have a play and let you know how I get on. Thanks. |
| 11:21.04 | irroot | look at t38modem / gnugk to work with ooh323 |
| 11:21.21 | irroot | have hylafax working fine with it |
| 11:21.37 | irroot | also linksys 2102 T.38 |
| 11:22.04 | irroot | no other devices tested by me but should work if compat with chan_sip |
| 11:23.16 | cneb3000 | i'll try tommorow buddy and let you know :) |
| 11:23.27 | irroot | thx its queued for 1.10 |
| 11:24.02 | irroot | mnicholson is helping test it think we very close |
| 11:24.31 | irroot | ill maintain the 1.8 branch once its released in 1.10 as i have customers using it |
| 11:24.51 | cneb3000 | anything I can do to help :) |
| 11:26.49 | irroot | its most appreciated |
| 11:27.04 | devyll | so.. any ideeas ? |
| 11:27.29 | devyll | I mean... any known issues ? if not I will stop bugging ... |
| 11:27.45 | *** join/#asterisk jkroon (~jkroon@41.216.197.151) |
| 11:28.16 | devyll | it is strange that no modifications were made anywhere in the configuration of the server or the configration of asterisk/dahdi .. and suddenlty it started behaving like i described above :( |
| 11:28.17 | cneb3000 | devyll: When you say 'interupted'.. do you mean terminated? |
| 11:28.33 | jkroon | is there someone from digium that can check up for me on why an order is being held up? |
| 11:28.56 | devyll | cneb3000 yes. sorry. the call is terminated |
| 11:29.13 | cneb3000 | jkroon: I just had to wait a couple of hours for a free fax for asterisk code to come through.. if it's something like that maybe there's a delay? |
| 11:29.25 | cneb3000 | jkroon: I was told customer services open in about 1 - 2 hours though. |
| 11:29.39 | jkroon | *sigh* |
| 11:32.26 | cneb3000 | devyll: google says a 'yellow alarm' is what is generated when the side you're talking to looses 'loss of signal'? |
| 11:33.29 | devyll | cneb3000, and can that be related to the notice above the Yellow Alarm ? the one with GDLC Bad FCS ? |
| 11:33.49 | devyll | because, I can't know if it's the ethernet board, the cable, the E1 provider, .. |
| 11:34.08 | irroot | devyll when i see that type of error 99% chance its telco |
| 11:34.11 | jkroon | cneb3000, yellow => we receive signal from remote end but they're not responding to our local sync requests. |
| 11:34.25 | devyll | great |
| 11:34.28 | devyll | I'll give them a call |
| 11:34.42 | cneb3000 | jkroon: thanks :) |
| 11:34.43 | irroot | devyll restart the NT the box that the E1 is connected too |
| 11:34.52 | jkroon | devyll, ??? could that affect BRI too? |
| 11:35.09 | irroot | and make sure you discon the battery backup pack |
| 11:35.18 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 11:47.45 | irroot | customer logs a ticket wants to know what "std deviation" is [on the ACD reports we gen from queue_log ... cruel bugger i am sent him the wikipedia link |
| 11:49.08 | cneb3000 | lazy! |
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| 11:51.06 | leifmadsen | irroot: that was probably the right move |
| 11:52.04 | ectospasm | one, two or three sigma? |
| 11:52.17 | ectospasm | I failed statistics the first two times around |
| 11:52.40 | irroot | simple rule if you do not understand the report its not for you |
| 11:53.25 | irroot | ectospasm besides we all know smoking is the leading cause of stats !!! |
| 11:53.44 | ectospasm | I thought it was drunk driving |
| 11:54.01 | ectospasm | ...maybe drunken smoking? |
| 11:54.15 | irroot | now that we need to study :P |
| 11:54.32 | irroot | get a UN pannel together and spend some $$$$ |
| 11:55.53 | *** join/#asterisk tfs (~username@108.115.71.111) |
| 11:58.12 | coppice | standard deviation is probably a statistic you'd derive in phone sex call centre |
| 11:59.43 | irroot | coppice lol they a medical aid / insurance call centre |
| 12:00.00 | irroot | so yeah bout the same someone getting screwed :P |
| 12:02.23 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 12:02.34 | *** join/#asterisk irroot (~gregory@dsl-185-122-118.dynamic.wa.co.za) |
| 12:03.45 | cneb3000 | http://xkcd.com/903/ |
| 12:03.45 | cneb3000 | <--- how relevant! |
| 12:04.04 | tuxx- | hehe :-) |
| 12:07.04 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-wfuakinlimxqglel) |
| 12:18.35 | QuantumSchema | Aha!! I wonder if this would work.... |
| 12:18.51 | QuantumSchema | The whole queue thing with not dialing an agent that is on a call.... |
| 12:19.08 | QuantumSchema | How would the queue react if I set the agent's number to DND? |
| 12:26.08 | fish-bulb | QuantumSchema: should be the same way as if they were on a call |
| 12:26.23 | fish-bulb | I forget the message, but it's something like Busy Here |
| 12:26.54 | QuantumSchema | Nice! I think I'm gonna give it a go. |
| 12:27.30 | QuantumSchema | I just wish there was a command like "QUEUESTATUS(agent,status)" similiar to "DEVSTATE()". |
| 12:31.48 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:32.17 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 12:32.17 | fish-bulb | QuantumSchema: what are you needing it for? |
| 12:34.02 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 12:35.01 | QuantumSchema | I've got Asterisk queues with external agents that dial in to login/out & pause/unpause. The queue knows that when an agent is called from a queue call to set it's queue state to "In Use". If the agent calls out side or recieves a non queue call, the agent's queue state stays "Available". I'm trying to manipulate that part. |
| 12:35.44 | QuantumSchema | All calls from external agents are still routed through Asterisk so it's fully aware of the calls so that's where I'm trying to get the dialplan to manipulate the queue status for the agent. |
| 12:39.24 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 12:42.22 | fish-bulb | I don't know of a way offhand to set the queue status non-queue calls, but could you add another step in the dialplan that set the device state? |
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| 12:48.27 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:49.58 | irroot | what to do with a customer who looses his RSA cert's password .... |
| 12:55.16 | *** join/#asterisk wonderworld (~ww@port-92-201-117-240.dynamic.qsc.de) |
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| 13:00.17 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:00.33 | russellb | irroot: charge a fee? |
| 13:01.13 | irroot | should charge a stupid tax :) |
| 13:02.16 | irroot | russellb good to see you think SVN needs a kick |
| 13:02.23 | russellb | k |
| 13:05.16 | *** join/#asterisk imox1234 (~imox1234@88.240.100.215) |
| 13:05.38 | russellb | irroot: stop breaking it |
| 13:05.40 | russellb | :-p |
| 13:05.59 | irroot | O:-) |
| 13:06.10 | *** join/#asterisk MariusAgon (~aa@79.142.116.89) |
| 13:07.02 | *** join/#asterisk saisoma (~saisoma@client105.jdcc.edu) |
| 13:09.06 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 13:09.40 | russellb | fixed |
| 13:09.43 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 13:09.47 | imox1234 | Sending fake auth rejection for device"""""" whats that for an error? |
| 13:10.24 | russellb | asterisk got a request that it didn't match to any entry in sip.conf |
| 13:10.33 | l2trace99 | man I just joined an my problems are fixed this group is awesome |
| 13:10.36 | russellb | it pretented it found one and requested authentication anyway |
| 13:10.42 | imox1234 | russellb: hmm ok thanks |
| 13:10.46 | russellb | that way it prevents someone from scanning for valid accounts |
| 13:12.18 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 13:12.43 | irroot | imox1234 russellb a good thing to do is have a iptables "tarpit" ratelimit sip packets from a ip |
| 13:13.14 | imox1234 | irroot: ?? |
| 13:13.26 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:14.01 | irroot | use RATELIM to help protect from a brute force attack |
| 13:14.48 | irroot | /etc/rc.d/rc.firewall:/sbin/iptables -A TARPIT -j RETURN -m state --state ESTABLISHED -m limit --limit 2/s --limit-burst 5 |
| 13:14.49 | irroot | /etc/rc.d/rc.firewall:/sbin/iptables -A TARPIT -j RETURN -m state --state NEW -m limit --limit 2/s --limit-burst 5 |
| 13:14.49 | irroot | /etc/rc.d/rc.firewall:/sbin/iptables -A TARPIT -j LOG -m recent --rcheck --seconds 30 --hitcount 20 --name RATELIM -m limit --limit 6/minute --limit-burst 1 --log-prefix "RATELIM " --log-level debug |
| 13:14.49 | irroot | /etc/rc.d/rc.firewall:/sbin/iptables -A TARPIT -j DENY -m recent --name RATELIM --update --seconds 30 --hitcount 20 |
| 13:14.49 | irroot | /etc/rc.d/rc.firewall:/sbin/iptables -A TARPIT -j RETURN -m recent --name RATELIM --set |
| 13:29.41 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
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| 13:32.19 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 13:33.16 | BMJ | Asterisk Developer Call at 10:00 EDT today. Info here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Developer+Conference+Call |
| 13:38.50 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:42.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:42.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:47.49 | m4xx | to prevent bruteforce on my system i use syslog then pipe the syslog to a perl script checking for failed registration strings, store the ip after that ip has x failed attempts in a period of y it firewalls them. can anyone see a reason not to do it this way? |
| 13:48.18 | BMJ | Asterisk Developer Call starting in 10 minutes. Bridge is open. Info here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Developer+Conference+Call |
| 13:48.54 | errr | m4xx: there is an app that does that.. I hope you didnt rewrite psad or fail2ban or one of the other solutions did you.. |
| 13:49.23 | m4xx | no i addopted my old sshd/mail/ftp script |
| 13:49.31 | m4xx | which does the same thing |
| 13:50.00 | m4xx | and i thought fail2ban worked on a cron |
| 13:53.14 | *** part/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 13:55.53 | cneb3000 | Can I join that call just to hear what goes on? or is it strictly Dev only? |
| 13:56.15 | cneb3000 | or would that be considered a pain in the back side |
| 13:56.56 | irroot | cneb3000 its all porn shhh dont tell em i let the secret out |
| 13:57.26 | BMJ | cneb3000: come join us. the more the merrier. |
| 13:57.42 | m4xx | apparently i wrote my script befor fail2ban existed ;D |
| 13:57.49 | cneb3000 | cool, thanks guys. and if there's no explicit material I'll be very upset |
| 14:01.36 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 14:03.07 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
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| 14:11.35 | justdave | I'm on 1.8.4.1... for purposes of this problem I have three servers, A, B, and C. Call originates on A, and is destined for C, but has to go through B to reach C. |
| 14:12.25 | justdave | actually, not even getting that far... |
| 14:12.45 | justdave | the call is made from a phone connected to A, routes to B over SIP |
| 14:13.01 | justdave | and dies there, because it tries to use the phone's username instead of server A's username when connecting to B |
| 14:13.17 | justdave | even though there's a username= entry in the sip.conf entry for B in A's config |
| 14:13.40 | justdave | so it gets an auth reject because the phone's username doesn't exist on B |
| 14:14.28 | cneb3000 | justdave: what does it say under the 'type=' for each of the references in server B and As sip.conf files? |
| 14:14.40 | justdave | friend |
| 14:14.49 | Chainsaw | leifmadsen: Is 1.8.4.1 out? |
| 14:14.59 | leifmadsen | yes |
| 14:15.04 | leifmadsen | as posted at asterisk.org |
| 14:15.30 | Chainsaw | leifmadsen: Cool, could you update the topic please? |
| 14:15.35 | Chainsaw | (That's really the only thing I look at) |
| 14:16.10 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.1 (2011/05/24), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 14:16.29 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:16.45 | leifmadsen | downloads page updated too |
| 14:17.09 | justdave | <PROTECTED> |
| 14:17.10 | justdave | [May 26 07:15:49] WARNING[23221]: chan_sip.c:13674 check_auth: username mismatch, have <237>, digest has <paris_sip> |
| 14:17.12 | *** join/#asterisk imox1234 (~imox1234@88.240.100.215) |
| 14:17.49 | leifmadsen | justdave: sounds like you have a configuration issue -- you probably need to look at 'fromuser' |
| 14:18.56 | justdave | leifmadsen: yup, that fixed it, thanks. |
| 14:19.01 | leifmadsen | np |
| 14:19.13 | justdave | I knew there had to be something like that missing. :) |
| 14:19.31 | leifmadsen | :) |
| 14:26.12 | *** join/#asterisk bent_screwdriver (~UserNick@74.255.249.66) |
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| 14:36.25 | Dynamicfail | Why isn't 1.8 in the ubuntu repos? |
| 14:37.48 | *** join/#asterisk espent (~espent@46.80-203-219.nextgentel.com) |
| 14:41.14 | *** join/#asterisk neurosys (~neurosys@69.199.128.254) |
| 14:41.20 | neurosys | Ugh I hate customers. |
| 14:42.27 | neurosys | Took a guy off a hosted and onto SIP *, Cisco 79**, He says Call waiting used to work on 1 line without using the second line, alowing him to use 4 lines total. Now it doenst do that. I have CW set in the cisco.. but it always rolls over. Anyone know anything about this? |
| 14:42.38 | *** join/#asterisk m_tadeu (~quassel@89.180.67.125) |
| 14:44.44 | m_tadeu | hi...what things can influence in the sound quality of a call? (well at least the most common) |
| 14:45.53 | m_tadeu | I'm only using SIP |
| 14:46.18 | m_tadeu | and from one day to another, the sound quality got sluggish |
| 14:46.21 | jacc0 | codec conversion |
| 14:46.48 | jacc0 | qos |
| 14:46.51 | jacc0 | reinvite |
| 14:47.11 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:47.24 | jacc0 | network load |
| 14:48.08 | *** join/#asterisk cneb3000 (~ben.cropl@gateway.magneticnorth.com) |
| 14:48.20 | jacc0 | distence between the speaker and the mic |
| 14:48.27 | jacc0 | speaker and mic volume |
| 14:48.29 | *** join/#asterisk Bart- (~bart@DanBUK-2-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 14:48.31 | Bart- | hi guys |
| 14:48.41 | jacc0 | hi bart |
| 14:48.46 | m_tadeu | hi |
| 14:49.27 | Bart- | i have a queue problem |
| 14:49.34 | Bart- | i get always [May 26 16:48:49] WARNING[5440]: app_queue.c:4176 queue_exec: Unable to join queue 'q-1' |
| 14:49.37 | Bart- | any idears? |
| 14:49.41 | Bart- | i reload the module already |
| 14:49.42 | leifmadsen | Bart-: not enough info |
| 14:49.45 | Bart- | ok |
| 14:49.53 | Bart- | the thing is i have a hotline |
| 14:50.01 | leifmadsen | could be anything really... invalid queue, joinempty=no and the queue is empty.... |
| 14:50.07 | leifmadsen | throws out random ideas |
| 14:51.16 | jaytee | picks up the random ideas that Leif threw away and claims them as his own |
| 14:51.31 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 14:51.36 | jaytee | "I invented the Internet!!!" |
| 14:51.54 | leifmadsen | jaytee: that's impossible! I invented the internet! |
| 14:52.36 | cneb3000 | you're both wrong. the internet has always been here. |
| 14:52.57 | cneb3000 | >_> |
| 14:53.02 | m_tadeu | jacc0: thanx...I'm checking those things |
| 14:53.49 | Bart- | ok sorry |
| 14:54.00 | m_tadeu | because I invented it like......forever :) |
| 14:54.13 | jaytee | On a related note, I've found that "An Inconvenient Truth" by Al Gore is a much cheaper cure for insomnia than Ambien or Lunestra and much, much cheaper with fewer side effects (mild nausea at the sound of Al's voice has been reported in < 20% of cases) |
| 14:55.45 | jacc0 | Yeah!! it has been laying around on a dvd here on my desk for years |
| 14:55.54 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:56.15 | cneb3000 | Al Gore sent me on a road of paranoia ¬_¬ |
| 14:56.30 | cneb3000 | to paranoia* |
| 14:57.23 | leifmadsen | http://www.lyrics007.com/Harvey%20Danger%20Lyrics/Paranoia%20Lyrics.html |
| 14:58.36 | anonymouz666 | if we are using real time sip peers |
| 14:58.48 | anonymouz666 | without rtcachefriends=yes there's no way to know the SIP device state |
| 14:58.51 | anonymouz666 | is that correct? |
| 15:00.02 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:00.37 | anonymouz666 | and if I have both servers 1 and 2, the queue is on 1 server, and the member is registered on 2 server with rtcachefriends=yes I will create a problem if the member on server 1 is SIP/ |
| 15:02.13 | leifmadsen | anonymouz666: right because the device has to be in memory in order to track the device state |
| 15:02.26 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 15:02.27 | leifmadsen | anonymouz666: you need distributed device state if the peers are remote |
| 15:02.36 | leifmadsen | either use openAIS or the XMPP method (in 1.8) |
| 15:02.56 | leifmadsen | e.g. peers are on one asterisk box, but the queue resides on a separate asterisk box |
| 15:03.49 | anonymouz666 | yeah but Queue() will have to call Local/ members not SIP/. Because SIP/ will search in memory of the queue separate box. |
| 15:04.10 | anonymouz666 | and If I use Local/ will lost pause states etc. |
| 15:05.17 | leifmadsen | anonymouz666: that's not a problem either, because you just tell the Local channel to use the state of the SIP device associated with the Local channel |
| 15:06.41 | anonymouz666 | I don't know if I understand |
| 15:07.06 | leifmadsen | like in queues.conf sample: ;member => Local/1000@default,0,John Smith,SIP/1000 |
| 15:07.43 | m_tadeu | if I could restrict the coded choice to only one coded, which should I choose? |
| 15:08.04 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 15:08.45 | anonymouz666 | ahhh understand it ! |
| 15:09.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 15:11.06 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 15:12.50 | leifmadsen | m_tadeu: codec you mean? |
| 15:13.06 | leifmadsen | m_tadeu: that depends on your bandwidth requirements and such, but I'm a fan of G.711u |
| 15:14.15 | m_tadeu | leifmadsen: *coded in deed :) well I have a limitation of 3 codecs[gsm/ulaw/alaw] |
| 15:14.33 | leifmadsen | I like ulaw if your network will allow it |
| 15:14.37 | leifmadsen | I don't like the sound of gsm |
| 15:15.06 | Chainsaw | leifmadsen: Particularly with hold music. |
| 15:15.11 | leifmadsen | amen |
| 15:15.23 | Chainsaw | leifmadsen: Vio*white noise*lin. |
| 15:15.25 | leifmadsen | I mean, I use G722 quite a bit, but not everything supports it yet |
| 15:15.46 | m_tadeu | Chainsaw: hehe nice description |
| 15:16.01 | leifmadsen | I've actually had customers complain about the use of g722 in their network because calls "sound weird" |
| 15:16.09 | leifmadsen | I respond with, "that's because it sounds better...." |
| 15:16.18 | leifmadsen | people are funny |
| 15:16.24 | m_tadeu | :) |
| 15:16.30 | Chainsaw | leifmadsen: Well, there is a failure mode with G722 where you get the wrong sample rate. |
| 15:16.39 | Chainsaw | leifmadsen: Which does sound very robotic and artificial. |
| 15:16.41 | leifmadsen | I expected the reaction to be, "omg! this new phone system sounds amazing!" |
| 15:16.42 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:16.48 | leifmadsen | Chainsaw: oh that's not the problem at all though :) |
| 15:16.53 | leifmadsen | they don't like how "clear" it sounds |
| 15:17.00 | Chainsaw | leifmadsen: Oh, right. |
| 15:17.15 | leifmadsen | it was my *facepalm* moment |
| 15:17.17 | Chainsaw | leifmadsen: I can't seem to find a proper way to enable it here, for handset-to-handset calls. |
| 15:17.22 | coppice | leifmadsen: there are genuine negative issues with G.722. the background comes over much better, which sometimes conveys embarrassing things |
| 15:17.47 | leifmadsen | coppice: true enough, but in this situation it's all open cubicals |
| 15:17.53 | leifmadsen | that was already a bit of an issue |
| 15:18.15 | leifmadsen | it was just an odd reaction I didn't expect ;) |
| 15:19.02 | m_tadeu | what are your thoughts about alaw? |
| 15:19.11 | coppice | many people will insist that G.711 is as hi-fi as you can get, and that its not compressed |
| 15:20.54 | anonymouz666 | m_tadeu: alaw/ulaw is nice to use over LAN |
| 15:20.59 | anonymouz666 | that's my opinion |
| 15:22.10 | coppice | its 2011. over a LAN you should be looking at something better than G.711 |
| 15:22.26 | anonymouz666 | speex then |
| 15:23.53 | anonymouz666 | coppice: I should be looking something that the clients supports hehe |
| 15:24.44 | *** join/#asterisk VenomX (~venomx@187.122.74.23) |
| 15:25.04 | coppice | then look at clients which support G.722. there are plenty of them now |
| 15:25.39 | anonymouz666 | why I should look for a codec that there are genuine negatives issues ? |
| 15:25.41 | anonymouz666 | :P |
| 15:26.16 | coppice | because the genuine positive issues are much stronger |
| 15:26.51 | anonymouz666 | good answer |
| 15:27.00 | *** join/#asterisk clayd (~Oscar@67.41.43.206) |
| 15:27.12 | m_tadeu | so right now I'm getting a bit noisy sound using ulaw on moh...it's the only call running on the system |
| 15:28.09 | clayd | Any recomendations (or pointer to good information) on what phones work best with an Asterisk system? |
| 15:34.37 | *** part/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 15:37.29 | MrNemus | would anyone know why sip channels would stay open ? |
| 15:38.19 | Chainsaw | MrNemus: It can happen for PUBLISH dialogs in 1.8 |
| 15:38.33 | Chainsaw | MrNemus: Does it show as "Rx: PUBLISH" in sip show channels? |
| 15:40.28 | MrNemus | http://pastebin.com/T1YLiJ5Q |
| 15:40.32 | MrNemus | this is what I see |
| 15:44.01 | QuantumSchema | Hay all! Is != valid in an ExecIf for Not Equal To? |
| 15:44.51 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:44.53 | leifmadsen | QuantumSchema: yes |
| 15:46.54 | *** join/#asterisk maxagaz (~maxagaz@123.118.82.76) |
| 15:47.01 | maxagaz | hi |
| 15:48.08 | m_tadeu | I'd like an advice on how to setup an mp3 for moh...which application should I use and how? |
| 15:48.10 | *** join/#asterisk FreeInfo (debian-tor@gateway/tor-sasl/freeinfo) |
| 15:48.35 | *** part/#asterisk FreeInfo (debian-tor@gateway/tor-sasl/freeinfo) |
| 15:49.16 | leifmadsen | m_tadeu: why not just convert the MP3 to something native asterisk can play? |
| 15:49.24 | leifmadsen | it's just silly to play actual mp3s |
| 15:50.21 | m_tadeu | and probably the origin of my noise :) |
| 15:50.37 | coppice | some people think all sound needs to be in MP3 |
| 15:51.07 | leifmadsen | speaks in mp3 |
| 15:51.16 | _Corey_ | I've had to go back to MP3 at a couple customers who've complained about hearing the "same five seconds" of a song everytime they go on hold... ;) |
| 15:51.32 | coppice | white man speaks with forked codec |
| 15:52.36 | coppice | _Corey_: its OK to put the whole of Wagner's ring on there in 16 bit linear |
| 15:53.05 | jaytee | even if they only get to hear the first five seconds |
| 15:53.30 | MrNemus | so does anyone know why the sip channels would stay open ? the carrier says they are sending the rtp hang up request it only happens when the person on the other end hangs up |
| 15:53.34 | jaytee | personally I prefer March of the Valkeries |
| 15:53.34 | coppice | but they won't hear the fat lady sing |
| 15:54.02 | _Corey_ | at least with MP3, they had some chance of hearing the fat lady sing :) |
| 15:57.01 | coppice | I prefer the Kinks "Tired of Waiting" |
| 15:57.10 | m_tadeu | well...does it really need to be mp3, so the clients don't get only the beginning of the music? |
| 15:57.55 | fish-bulb | ah, the Kinks. I could stand any queue if they were the on hold music |
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| 15:58.16 | _Corey_ | when playing in 'files' mode, it does a new thread per MOH session if I'm not mistaken... hence always starting at the beginning of a song |
| 16:00.34 | m_tadeu | is there a work around that? |
| 16:01.20 | _Corey_ | it's more of a preference than a problem, I'd say |
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| 16:10.36 | Maxxed | yip |
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| 16:20.44 | florz | Bart-: are there any agenty in the queue? |
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| 16:38.40 | d_preston215 | How would I go about sending IP phones through a VPN? |
| 16:40.01 | jpsharp | build the VPN, route appropriately, just as you would regular nonvoip traffic. |
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| 16:48.14 | diatonic | Can anyone comment on how service is from VoicePulse? The seem to be the only provider that has DIDs in a particular rate center I need. Struck out with Vitelity & Flowroute. |
| 16:51.45 | KavanS | diatonic, I use it, it's quite reliable |
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| 16:52.06 | KavanS | no real complaints with them - other than their international dial out policy, you need to provide them with the exact countries you intend to dial if you plan on using them for overseas dialing |
| 16:52.27 | KavanS | so if you do any "real" outbound overseas dialing, I'd use someone different for that aspect of your operations ;) |
| 16:53.06 | QuantumSchema | Is there any response I can provide or action I can take in a dialplan that would not ring an agent but not be listed as "RINGNOANSWER"? |
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| 16:53.55 | diatonic | KavanS, thanks |
| 16:54.27 | KavanS | overall quite reliable, have had very little downtime if any |
| 16:54.36 | KavanS | our guys monitor it with nagios and I've heard no complaints |
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| 17:10.55 | QuantumSchema | Bueller? |
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| 17:42.11 | jpsharp | SIP + Nat = bad things. SIP + NAT + NAT = seriously broken things. |
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| 17:43.44 | Freeaqingme | jpsharp: you're missing a step |
| 17:43.52 | Freeaqingme | NAT = horror |
| 17:44.27 | jpsharp | There is that. |
| 17:46.27 | saisoma | hey guys, question regarding a PRI and ss7. i am having issues calling AT&T toll free numbers. |
| 17:46.46 | saisoma | AT&T says that I need to "open the voice path early" and they are sending me ss7 messages to do that |
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| 17:47.55 | saisoma | my chan_dahdi.conf has nothing about ss7 in it. i did install it, but do i need configuration for it? |
| 17:47.55 | saisoma | http://pastebin.com/KS7Lf49N |
| 17:50.34 | MrNemus | would anyone know why sip channels are not closing ? or a setting I could set to close inactive channels |
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| 17:51.50 | jpsharp | saisoma: According to your config, you're not using SS7, just plain ISDN. |
| 17:52.23 | JonathanRose | is assaulted by a thousand heavily escaped SIP messages |
| 17:53.09 | saisoma | jpsharp: that's what i thought. what info do i need from my provider to configure ss7? i assume ss7type, maybe pointcode (do i make one up?), adjpointcode, defaultdpc? |
| 17:53.32 | saisoma | jpsharp: thanks for answering too btw. ss7 is totally new to me |
| 17:54.10 | WIMPy | SS7 and "national" are rather different things. You need to agree with your telco which one you use. |
| 17:54.59 | saisoma | WIMPy: *nod* is it possible that using national instead of ss7 could cause other issues, such as local echo (extraordinarily loud sidetone)? |
| 17:55.27 | WIMPy | It will cause other issues, like nothing working at all. |
| 17:55.36 | jpsharp | Yeah, it shouldn't even sync up. |
| 17:57.30 | jpsharp | and 'voice path early' sometimes means "inband call progress". You can try turning that stuff on in the dahdi config. |
| 17:57.34 | saisoma | WIMPy: that's what is strange, since we work no problem except for calling AT&T toll free numbers. the tech said they see ss7 messages going to us. that's why i was confused. i'll be contacting them for additional info, thanks guys. as always, everyone here gets me info that i pull teeth trying to get elsewhere |
| 17:57.48 | saisoma | jpsharp: ohhh . rgr. will find that setting, thanks |
| 17:58.41 | WIMPy | Yes, you might want to experiment with early medai settings. |
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| 17:59.05 | WIMPy | But that might require experimenting with software versions. |
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| 18:03.57 | sunfone | Does anyone have any troubles with Dahdi channels (on RBS T1) getting stuck in "ringing" state? |
| 18:04.12 | maxagaz | hey, there's something I don't understand about E1 line |
| 18:04.29 | maxagaz | E1 is an ethernet cable ? |
| 18:04.48 | sunfone | E1 is a TDM circuit (non US/Japan) with 30 channels I believe |
| 18:05.26 | sunfone | Handoff is an RJ45, though, so it *looks* like an ethernet cable :) |
| 18:05.48 | maxagaz | sunfone, ok but, when I plug it on my server, it can be plugged to an ethernet card or has to be plugged on a special PBX card ? |
| 18:05.58 | sunfone | PBX card |
| 18:05.58 | maxagaz | like PCI-E card |
| 18:06.15 | sunfone | Many folks make them - Digium of course, Sangoma, Rhino, etc |
| 18:06.52 | sunfone | I use Sangoma A10Xd |
| 18:08.25 | jpsharp | Plugging an E1 RJ45 into an ethernet card will possibly let the magic smoke out of the Ethernet card. |
| 18:09.32 | irroot | jpsharp but not too impressive :) |
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| 18:29.08 | WIMPy | Wouldn't the other way be more likely? |
| 18:29.48 | axilla | i have a question, i'm having an issue with inbound calls coming in as an unknown peer |
| 18:29.52 | axilla | asterisk 1.6 |
| 18:30.07 | axilla | only way i can get inbound calls to come in is by turning on anonymous inbound calls allowed |
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| 18:30.35 | mducharme-work | afternoon |
| 18:31.27 | mducharme-work | we want to connect our alarm system into an FXS port on our asterisk server to go through our PRI instead of paying for the monthly fee for an analog line for the alarm system - is that advisable? |
| 18:31.32 | WIMPy | evening |
| 18:31.51 | irroot | good night |
| 18:32.15 | WIMPy | mducharme-work: You might want to ask your insurance company about that. |
| 18:32.17 | mducharme-work | we have the appropriate ports and they are all configured.. we do faxing over voice over ip right now with no problems so I assume a dial up modem should work |
| 18:32.49 | sunfone | mducharme-work: its not always a modem - sometimes DTMF driven |
| 18:32.49 | mducharme-work | our alarm system fails over to a cellular connection if the dial up isn't working |
| 18:32.58 | axilla | Received incoming SIP connection from unknown peer to xxxxxxxxxxxxx |
| 18:33.17 | mducharme-work | so it's not as big of a risk as if it didn't have that fail over option |
| 18:33.29 | WIMPy | mducharme-work: Then I guess you might go ahead. |
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| 19:56.09 | jtrimmer | I have a custom extension setup so that when I dial 1099 it is suppose to execute a system command and broadcast callerid information. When I execute the command from the command prompt it works fine. when asterisk does it nothing happens that I can tell. here is the extension and only thing I can see from asterisk http://pastebin.com/KY9hvQpf |
| 19:59.28 | sezuan | Is it possible to define a template from which a sipeer/sipuser entry in ldap gets his defaults? |
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| 20:43.19 | GreatSUN | re |
| 20:43.35 | USCooler25 | I am looking for some help with my asterisk server. Am I in the right place? |
| 20:43.52 | WIMPy | ~ask |
| 20:43.53 | 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. |
| 20:45.35 | USCooler25 | Sorry, I have a server that periodically goes unresponsive to our phones and t1 line. The server appears to be running, I can connect to it and run commands on the CLI, but it will not do anything. It doesn't appear to be running any out of controll processes, and the only way to get it working again is to reboot the computer. Restarting asterisk doesn't seem to work. I have Elastix 2.0.3 |
| 20:46.28 | WIMPy | What kind of phones? |
| 20:46.44 | USCooler25 | Is there anything I can look for in the logs to trace what is happening. I have traced back several minutes before the event happens, but I can not see anything out the ordinary. |
| 20:47.15 | USCooler25 | polycom |
| 20:47.23 | USCooler25 | som 550s, and some 301s |
| 20:47.30 | USCooler25 | *some |
| 20:48.15 | USCooler25 | it is like the server is offline, but I can ping it, connect to it over ssh. |
| 20:48.24 | WIMPy | If you get issues both with T1 and SIP at the same time and restarting Asterisk won't resolve the issue, there must be something really strange going on. |
| 20:48.41 | WIMPy | Or do you have the T1 on a SIP gateway? |
| 20:48.51 | USCooler25 | sometimes it will go days with no problems, then some days I have to reboot it a few times |
| 20:49.07 | USCooler25 | No, the T1 is using a digium card to the phone provider |
| 20:51.23 | WIMPy | Even if you have issues with the card at a lower level, I can't imagine how that could prevent SIP phones from working after a restart of Asterisk. |
| 20:53.06 | USCooler25 | It seems to me like the system is not responding to any requests, like it is overloaded or something, but everything I look at says the system is pretty much idle when it is happening. |
| 20:53.49 | USCooler25 | It doesn't drop calls that are in progress, it just won't let any new calls initiate. |
| 20:54.45 | WIMPy | Ok, so what exactely happens, when you try to place a call? Both from the T1 and from the SIP phones. |
| 20:55.43 | WIMPy | Does the system have a high load? That can happen, even when it's idle. |
| 20:56.18 | USCooler25 | Sorry about the akf |
| 20:56.22 | USCooler25 | *afk |
| 20:56.47 | USCooler25 | When a call is placed from the phone, it just sits there and does nothing. |
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| 20:57.48 | USCooler25 | From the outside I am not sure exactly what happens. |
| 20:58.09 | USCooler25 | I will have to try to call in when it happens the next time. |
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| 21:02.36 | kaldemar | USCooler25: what is the asterisk version that you're using? |
| 21:02.50 | USCooler25 | Just a sec |
| 21:02.52 | *** part/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 21:03.13 | USCooler25 | 1.6.2.13 |
| 21:08.32 | kaldemar | there have been bugs that made asterisk not close UDP sockets. that might stop SIP from working. |
| 21:08.47 | kaldemar | https://issues.asterisk.org/view.php?id=17255 |
| 21:09.42 | kaldemar | someone found that in 1.6.2.16.1, it only happens when SIP session timers are in use. |
| 21:10.14 | WIMPy | But that situation would be cleared by a restart. |
| 21:11.15 | wdoekes2 | USCooler25: next time, get a backtrace from the running asterisk (if you don't have core show locks): gdb -p `pidof asterisk` -ex 'thread apply all bt full' -ex detach -ex quit > output.txt |
| 21:11.27 | USCooler25 | could paging to several extensions cause over use of the ports? |
| 21:11.36 | wdoekes2 | if you cannot restart asterisk, you're probably looking at some deadlock |
| 21:11.50 | wdoekes2 | 'core show locks' is useful too, but you need to have that compiled in |
| 21:12.26 | wdoekes2 | mm.. the udp leaks never caused it to not restart, afaik |
| 21:14.45 | WIMPy | USCooler25: Does it run again after you restart it? |
| 21:15.42 | USCooler25 | after i reboot, yes |
| 21:15.49 | wdoekes2 | and does killing asterrisk with -KILL work? |
| 21:16.04 | USCooler25 | i didn't try that, i tried amportal stop and start |
| 21:16.16 | WIMPy | And what happens if you just restart Asterisk? |
| 21:16.37 | USCooler25 | i only tried it once, and it didn't change anything. |
| 21:16.38 | WIMPy | Hmm, whatever that might do. |
| 21:17.11 | wdoekes2 | USCooler25: if it works again after a -KILL, the card is not to blame, if it doesn't the card probably is |
| 21:17.28 | WIMPy | Next time try 'core restart now' in the *CLI or if that won't do, kill it. |
| 21:17.30 | USCooler25 | the t1 card? |
| 21:17.34 | wdoekes2 | restarting a computer should alsmost never be necessary |
| 21:18.01 | USCooler25 | i'll try that |
| 21:18.17 | USCooler25 | but I would still like to figure out what is causing the instability |
| 21:18.28 | ChannelZ | global warming |
| 21:18.32 | USCooler25 | lol |
| 21:18.48 | wdoekes2 | get the gdb output when it hangs |
| 21:18.49 | ChannelZ | It causes everything! |
| 21:18.52 | USCooler25 | global warming is a myth |
| 21:19.13 | WIMPy | You could check for leaked sockets with netstat. |
| 21:19.15 | Freeaqingme | yup |
| 21:19.23 | Freeaqingme | I think central heating has been proven way better |
| 21:27.25 | USCooler25 | Thank you for your help guys. Hopefully I can get closer to finding the problem. |
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