| 00:06.00 | [TK]D-Fender | And 0-2. |
| 00:19.12 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 00:29.10 | *** join/#asterisk yaaago (~kresp0@gateway/tor-sasl/kresp0) |
| 01:04.06 | *** join/#asterisk D30 (~deo@203.177.9.66) |
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| 01:52.44 | starks | hi |
| 01:56.08 | starks | i came in here a few days ago about getting cisco 79xx phones working on switchvox/asterisk. got the registration done using siproxd, but there's no audio |
| 02:15.03 | starks | nvm |
| 02:15.10 | starks | got it, need to tweak siproxd |
| 02:16.22 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |
| 02:31.36 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@sm3e210fc5.cust.navigue.com) |
| 02:31.50 | eXcAliBuR | I hate to do this, but what is better asterisk or freepbx? |
| 02:32.36 | WIMPy | What is better? Oranges or cars? |
| 02:34.43 | eXcAliBuR | I like ice cream |
| 02:34.45 | eXcAliBuR | :} |
| 02:35.05 | eXcAliBuR | point taken |
| 02:35.11 | eXcAliBuR | they are different and can't be compared |
| 02:35.15 | eXcAliBuR | so |
| 02:35.20 | eXcAliBuR | how are they different? |
| 02:35.24 | eXcAliBuR | what makes asterisk better? |
| 02:35.43 | WIMPy | FreePBX uses Asterisk. |
| 02:35.50 | eXcAliBuR | I knew it |
| 02:35.54 | eXcAliBuR | i fuckin knew it |
| 02:35.56 | [TK]D-Fender | FreePBX CONFIGURES asterisk |
| 02:36.01 | [TK]D-Fender | ^ |
| 02:36.26 | eXcAliBuR | damn i'm smart |
| 02:36.57 | eXcAliBuR | brb, i'm going to find my nuts |
| 02:37.21 | [TK]D-Fender | What you asked is like asking "what's better, a star-shaped extrusion of Play-Doh, or a bucket of Play-Doh you can make anything you want out of" |
| 02:38.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 03:26.38 | igcewieling | Does anyone have experience with Asterisk 11 hangup handlers? I have some questions regarding how to determine which leg the handler is running on. |
| 03:51.21 | *** join/#asterisk damania (~damania@108-206-226-245.lightspeed.irvnca.sbcglobal.net) |
| 03:51.22 | damania | any good kvm ovr ip device recommendations out there? |
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| 04:12.46 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 04:38.06 | mnathani | I am seeing a strange IP in my sip debug 192.168.0.25 which should not be on my network |
| 04:41.18 | mnathani | http://pastebin.com/1WripDJs |
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| 05:16.16 | Psil0Cybin | Hello everyone :) |
| 05:17.00 | mnathani | No such command 'sip set debug on' |
| 05:17.20 | mnathani | I could have sworn I used the same command just a little while back |
| 05:21.01 | ChannelZ | then chan_sip is probably not loaded |
| 05:24.37 | *** join/#asterisk starks (~eric@remote.continuitycenters.com) |
| 05:25.20 | starks | just want to thank you guys for being awesome |
| 05:33.15 | ChannelZ | please insert cookie |
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| 05:44.17 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 06:23.01 | D30 | hi all, need some input from you guyz,, im going to use another TDM card for my asterisk box,.. the old card was actually damaged for some reason and i need to replace it.. its my first time doing it :) |
| 06:24.05 | D30 | i already inserted the card to the board, i can see that its detected.. |
| 06:24.28 | D30 | but dont know whats next :) |
| 06:25.13 | D30 | i am using a TDM400 card btw.. |
| 06:34.22 | ChannelZ | well you need to configure /etc/dahdi/system.conf and then /etc/asterisk/chan_dahdi.conf |
| 06:35.10 | *** join/#asterisk lorsungcu (~anonymous@216.160.1.107) |
| 06:36.00 | D30 | just that two configs ChannelZ ? |
| 06:37.19 | ChannelZ | more or less. /etc/dahdi/system.conf configures the hardware and creates channels, /etc/asterisk/chan_dahdi brings those channels into asterisk |
| 06:37.58 | ChannelZ | What configuration is the TDM400? 2 FXO/2FXS, a single channel, etc ? |
| 06:39.32 | *** join/#asterisk Tokeiito (~quassel@193.219.74.164) |
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| 06:43.56 | D30 | yes ChannelZ its 2FXS and 2FXO |
| 06:46.16 | D30 | http://codepad.org/UBgNCe9T |
| 06:48.38 | ChannelZ | what is that out of? |
| 06:49.58 | D30 | what u mean? |
| 06:51.00 | ChannelZ | What generated that text |
| 06:51.57 | D30 | ahh lsdahdi |
| 06:52.17 | ChannelZ | oh. I forgot about that one. |
| 06:52.26 | ChannelZ | Well generally you can run dahdi_genconf and it will make you system.conf |
| 06:52.55 | ChannelZ | then you can manually make your asterisk/chan_dahdi.conf |
| 06:53.52 | D30 | okay ill try that,,.. thanks ChannelZ |
| 06:57.01 | ChannelZ | wanders off for awhile |
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| 11:15.45 | toresbe | Howdy folks :) |
| 11:16.10 | mirela666 | aloha |
| 11:16.23 | toresbe | I was just wondering; are there any hardware options available that will connect an analog phone to Asterisk with any better voice codec than alaw/ulaw? |
| 11:17.14 | WIMPy | No. That doesn't make sense. POTS phones aren't made for more quality. |
| 11:18.27 | toresbe | Well, just because they aren't specifically made for more quality does not mean that they would be unable to represent a better signal than what T-carrier allows. I've been listening to a lot of old phreaking recordings, and I was amazed at the level of detail that would come through an analog line. :) |
| 11:19.41 | toresbe | For instance, when you reach an operator you can certainly tell if she was in the same building as the central office, because there really is plenty of high-frequency response |
| 11:19.47 | WIMPy | Well, I'm not saying it's impossible, but the difference would probably not be worth any extra effort. |
| 11:20.46 | WIMPy | And dahdi is G.711 only anyway. So you'd have to look for an ATA doing it. Bot I haven't heard of any. |
| 11:21.06 | toresbe | Dang. That's a real shame. T-carrier really sounds like crap compared to a lot of the old analog systems that were in use. |
| 11:23.16 | WIMPy | I've made my comments about not supporting G.722 a few times. |
| 11:24.30 | toresbe | G.722 to me has an unpleasant feeling to it that I can't put my finger on... what's the gist of your opposition to it? |
| 11:25.14 | WIMPy | Twice the quality for (max) the same bandwidth. |
| 11:25.28 | WIMPy | And free |
| 11:25.54 | toresbe | Oh, I thought you were against it. It's certainly more comfortable than G.711 for conversations, that's true. |
| 11:26.30 | toresbe | It's quite grating to be on a call to a mobile subscriber when handoff occurs from a AMR-WB-capable cell to one that is not. |
| 11:26.43 | *** join/#asterisk tedstriker (~tedstrike@host-135-196-33-208.lines.viateldsl.com) |
| 11:26.52 | WIMPy | No. Some people her argue that it has never worked anywhere. But that isn't true. It has just been hardly used. |
| 11:30.09 | toresbe | https://soundcloud.com/evan-doorbell/new-york-city-routings-1970-b#t=32:00 has one interesting comparison between a tone coming in from an ESS central office over wire trunks and T-carrier :) |
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| 12:42.55 | StaRetji | folks, I am sending Hangup(34) but my peer says call is hangup with 63 - Service or option not available, unspecified |
| 12:43.09 | StaRetji | I have no idea why |
| 12:43.56 | StaRetji | I've made a list of DNID numbers which I dont wan't to allow to call trough my switch ,but still I want to give them option to failover |
| 12:46.19 | WIMPy | Cause-SIP-cause mapping is not 1:1 |
| 12:46.24 | WIMPy | But why are you sending 34? |
| 12:48.19 | StaRetji | 34 sends normal circuit congestion |
| 12:48.45 | WIMPy | Why do you want to send a temporary failure? |
| 12:48.52 | StaRetji | so i thought if I hangup with 34 peer will get this code as circuit congestion and will failover to their second trunk |
| 12:49.02 | StaRetji | ehm |
| 12:49.28 | StaRetji | well, cause they are blocked temporary |
| 12:49.33 | StaRetji | tomorrow I might unblock them |
| 12:49.38 | WIMPy | Ok, would make sense. |
| 12:49.55 | StaRetji | only problem is they don't get 34 |
| 12:50.07 | WIMPy | If you blocked trhm, why don't you reject the calls (21)? |
| 12:50.16 | StaRetji | will they failover? |
| 12:51.04 | WIMPy | It will always depend on their configuration. |
| 12:51.47 | StaRetji | cause usually, if circuit is congested they would failover |
| 12:51.55 | Farkie | Hi guys, I explained my problem here yesterday - basically a 2nd call which comes in over SIP rather than the hard line, call is silent |
| 12:51.57 | StaRetji | at least I hope so lol |
| 12:52.13 | Farkie | when it happens, this is the log that we get |
| 12:52.31 | Farkie | http://pastebin.com/NRCm0XqS |
| 12:52.34 | WIMPy | You could also try 42 |
| 12:52.52 | StaRetji | let me test, there is no other way to find out ;) |
| 12:57.34 | StaRetji | I did, dude said 63 again |
| 12:58.26 | WIMPy | Well, you know the story about the source being available, I guess... |
| 12:58.41 | StaRetji | yeah |
| 12:58.48 | WIMPy | Although it's probably their end that messes it up. |
| 12:59.16 | StaRetji | exten => _X.,n,Hangup(42) |
| 13:00.05 | *** join/#asterisk illuder (illuder@41.160.33.43) |
| 13:00.21 | StaRetji | this is cli response: -- Executing [3366123432@filter:15] Hangup("SIP/7051206811-0000095b", "42") in new stack |
| 13:00.31 | StaRetji | so, I guess i am sending it |
| 13:00.47 | Farkie | What is the best way to diagnose a NAT issue |
| 13:01.13 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 13:01.45 | WIMPy | Well, SIP itself doesn;t use the cause codes. It's translated to com SIP response. And these translations tend to be debatable. |
| 13:02.29 | *** join/#asterisk vlad_starkov (~vlad_star@81.213.74.232) |
| 13:02.50 | StaRetji | could it be that they are not getting anything, that's why they moan about 63 |
| 13:07.33 | WIMPy | There should be some distinct mappings. |
| 13:08.21 | WIMPy | Are they maybe just using DIALSTATUS and no HANGUPCAUSE at all? |
| 13:08.47 | WIMPy | Assuming they are using Asteris, taht is. |
| 13:09.12 | StaRetji | I am trying to find out, you know, if I wanna die, I will send this guy to go for death lol |
| 13:09.40 | StaRetji | hehehe. they use some commercial switch |
| 13:09.58 | StaRetji | ha, weird, I removed and left only Hangup() |
| 13:10.02 | WIMPy | So where is that 63 comming from? |
| 13:10.06 | StaRetji | 57 - Bearer capability not authorized |
| 13:10.11 | StaRetji | this is what they get now |
| 13:10.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:10.28 | WIMPy | Sounds rather broken. |
| 13:24.25 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 13:29.06 | *** join/#asterisk vlad_starkov (~vlad_star@81.213.74.232) |
| 13:31.34 | StaRetji | Cause No. 57 - bearer capability not authorized. |
| 13:31.34 | StaRetji | This cause indicates that the user has requested a bearer capability which is implemented by the equipment which generated this cause but the user is not authorized to use. |
| 13:32.20 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 13:32.42 | StaRetji | I have this in my sip.conf |
| 13:32.44 | StaRetji | use_q850_reason=yes |
| 13:33.04 | StaRetji | is that okay or I should change to =no |
| 13:34.37 | *** join/#asterisk vassilux (~vassilux@LMontsouris-656-01-80-156.w82-127.abo.wanadoo.fr) |
| 13:41.42 | StaRetji | I've changed to Hangup(38) they are getting 41 |
| 13:41.44 | StaRetji | lol |
| 13:50.23 | *** join/#asterisk devil_evoxxx (~d3v1l@82.193.15.178) |
| 13:50.48 | devil_evoxxx | hi all guy, there is something new in hints on asterisk 11? i've upgrade from a 1.8 and i'm having trouble |
| 13:51.37 | WIMPy | No, yes gives them the chance to get the untranslated numerical cause. |
| 13:53.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:53.59 | Chainsaw | devil_evoxxx: I had trouble with them, and had to revert to the "old" way. |
| 13:54.48 | devil_evoxxx | Chainsaw: what's the old way? |
| 13:55.19 | devil_evoxxx | now i'm doing exten => _2XX,hint,SIP/int-{EXTEN} |
| 13:55.48 | [TK]D-Fender | I hope not... |
| 13:56.00 | [TK]D-Fender | Is that the actual line? |
| 13:56.02 | WIMPy | And what's the trouble? |
| 13:56.08 | [TK]D-Fender | lack of $ <------------- |
| 13:56.27 | StaRetji | WIMPy: okay, I am waiting for them to confirm if 41 will give them option to failover. Though I am doing Hangup(38) they get 41 - Temporary failure |
| 13:56.35 | StaRetji | hilarious |
| 13:56.38 | StaRetji | :) |
| 13:57.14 | devil_evoxxx | nono, i missed the $ writing here |
| 13:57.18 | devil_evoxxx | on the config the $ is present |
| 13:57.38 | devil_evoxxx | i cant see in asterisk cli "extensions changed..blablabl" |
| 13:57.58 | Chainsaw | devil_evoxxx: extenpatternmatchnew=no |
| 13:58.32 | WIMPy | Well, that's not specifically for hints. |
| 13:58.44 | Chainsaw | But the hints in my dial plan do not work unless I set that. |
| 13:58.53 | devil_evoxxx | Chainsaw: in general section? |
| 13:59.10 | Chainsaw | devil_evoxxx: Correct. |
| 13:59.13 | WIMPy | My whole dialplan acts ups up completely without that. |
| 13:59.41 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 14:03.03 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:03.38 | devil_evoxxx | still not working, |
| 14:06.36 | devil_evoxxx | my dialplan is this http://pastebin.com/ejv30RBk |
| 14:06.41 | devil_evoxxx | nothing complicated :( |
| 14:07.04 | devil_evoxxx | and in both context and subscribecontext on sip.conf i've setted internal-call |
| 14:07.31 | WIMPy | Do you have call-counters enabled? |
| 14:08.23 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:08.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:08.31 | StaRetji | WIMPy: I've set 31 now, they get 18 |
| 14:09.02 | StaRetji | they are definitely getting codes, but wrong ones |
| 14:09.07 | StaRetji | weird, right? |
| 14:09.42 | WIMPy | Yes, but perfectely normal. |
| 14:09.48 | [TK]D-Fender | devil_evoxxx: What ver of *? pastebin the actual subscription attempts, etc |
| 14:09.59 | devil_evoxxx | WIMPy: do you intend call-limit ? |
| 14:10.09 | StaRetji | hm, I am thinking to try random codes until they get 34 lol |
| 14:10.25 | WIMPy | devil_evoxxx: That's one way to enable them. |
| 14:10.41 | devil_evoxxx | [TK]D-Fender: Asterisk 11.7.0, i will paste the sip-sub asap |
| 14:11.25 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 14:12.50 | devil_evoxxx | WIMPy: i have to add call-limits=x ? |
| 14:14.03 | leifmadsen | pretty sure there is no call-limits setting in 11 (removed after 1.8 I think) |
| 14:14.08 | leifmadsen | callcounter=yes I believe is the setting |
| 14:14.17 | leifmadsen | if you're at all talking about device state |
| 14:15.01 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 14:15.44 | devil_evoxxx | i can confirm the callcounter = yes solveed |
| 14:15.55 | leifmadsen | bows |
| 14:21.59 | StaRetji | WIMPy: YES! |
| 14:22.00 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 14:22.01 | StaRetji | lol |
| 14:22.21 | StaRetji | I've set Hangup(3) and they are getting 34 |
| 14:23.03 | *** join/#asterisk serafie (~erin@nat/digium/x-uitgsuyeyemipalw) |
| 14:25.04 | *** join/#asterisk [Outcast] (~Outcast]@50-195-8-201-static.hfc.comcastbusiness.net) |
| 14:25.58 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 14:26.36 | *** join/#asterisk calum_ (~calum_@host109-151-195-197.range109-151.btcentralplus.com) |
| 14:27.07 | BeachBall | starts to bounce.... look who it is |
| 14:28.02 | WIMPy | StaRetji: Now make sure to document the type of equipment they use and how to trick it in to the right way. |
| 14:37.04 | StaRetji | WIMPy: this is going crazy mate, now my peers says that his peers is getting Unallocated (unassigned) number |
| 14:37.15 | StaRetji | while he gets 34 |
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| 14:50.25 | StaRetji | WIMPy: their switch is Avangard 2.0 |
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| 14:52.49 | jeffspeff | I'm having an issue with a phone registering. Here's the SIP debug logs. The phone in question is coming from IP 172.13.221.190 at extension 1225. And the traffic is coming from over the internet. http://pastebin.com/yd2NeTfw |
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| 14:56.58 | [TK]D-Fender | jeffspeff: Nowhere in there do I see that IP trying to register |
| 14:57.32 | [TK]D-Fender | jeffspeff: Or sendinging anything at all to your server |
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| 15:13.23 | jeffspeff | [TK]D-Fender, when the phone would reboot, i could see the console showing that the peer had registered then it would show the useragent/device for that exten. * would also do the necessary load from the realtime sip db. however, shoing the peer would state that it was unreachable. |
| 15:13.48 | [TK]D-Fender | jeffspeff: Here we see nothing. |
| 15:14.07 | jeffspeff | i pretty much gave up on it yesterday. i think it has something to do with the att uverse modem/router combo |
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| 15:35.24 | vader- | quick opinion question. We are currently running an older version of Asterisk like 1.2.x... All dial plans have been done by hand, all sip connections, etc. If you had to move to a GUI type interface for configuration what would you recommend? Our enviroment is about 70-100 SIP Phones, Digium TDM2400 and a Digium PRI card... Right now it's easy for me to provision a phone with a cisco config |
| 15:35.24 | vader- | file, and the asterisk settings, etc. But for other people in my department and anyone new we hire they need point and click... Just looking for any suggestions you guys might have. |
| 15:36.48 | garymc | What time do Schmooze open? |
| 15:36.54 | garymc | for business? |
| 15:42.46 | leifmadsen | Qwell: ^^^ ? |
| 15:44.19 | Qwell | Whatever time the website says. |
| 15:46.06 | leifmadsen | Qwell: :) |
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| 15:52.50 | BeachBall | if i am doing sip calling only over internet, i don't need libpri or dahdi? |
| 15:52.52 | BeachBall | right? |
| 15:53.17 | boom^time | BeachBall, right. |
| 15:54.31 | [TK]D-Fender | BeachBall: Last I heard, as of * 11 you still needed DAHDI for Meetme & Page |
| 15:54.42 | BeachBall | >:( |
| 15:54.53 | BeachBall | stops the ./configure |
| 15:55.16 | BeachBall | issues make clean |
| 15:55.17 | BeachBall | :/ |
| 15:56.39 | boom^time | BeachBall, all you said was you were doing sip calling over the internet. That's all I do and I don't require libpri or DAHDI |
| 15:56.43 | file | Page in 11 uses ConfBridge, so DAHDi is not required |
| 15:56.58 | file | Meetme will always require DAHDi |
| 15:57.07 | [TK]D-Fender | file: Good to know. |
| 15:57.38 | [TK]D-Fender | file: in what version was non-dahdi timing usable for IAX Trunking? |
| 15:58.04 | file | 1.8 for sure |
| 15:58.30 | leifmadsen | prefer timerfd over pthread |
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| 16:06.02 | BeachBall | so confbridge doesn't need it... great |
| 16:06.07 | BeachBall | well better installed than not i guess |
| 16:06.57 | BeachBall | hugs file |
| 16:07.12 | file | ConfBridge has never needed it |
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| 16:13.47 | BeachBall | ummm your spose to hug back |
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| 16:38.50 | BeachBall | i did this and broke asterisk /usr/sbin/safe_asterisk |
| 16:38.54 | BeachBall | :{ |
| 16:39.16 | BeachBall | now i get this error /usr/sbin/asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 16:39.45 | navaismo | ldconfig |
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| 16:42.48 | ghost75 | how can i find out needed t.38 settings for provider |
| 16:43.58 | ghost75 | res_fax.c:1934 sendfax_t38_init: channel 'SIP/personalvoip_out-00000000' refused to negotiate T.38 |
| 16:44.04 | ghost75 | res_fax.c:2020 sendfax_t38_init: Audio FAX not allowed on channel |
| 16:44.37 | ghost75 | i use sendfax with z option |
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| 16:53.22 | *** join/#asterisk terretz (4526a036@gateway/web/freenode/ip.69.38.160.54) |
| 16:54.15 | terretz | Howdy |
| 16:55.35 | terretz | I have a question about the salesforce integration with Asterisk running on a Switchvox and I'm wondering if anyone has experience with customizing the salesforce panel in switchboard. |
| 16:57.58 | [TK]D-Fender | terretz: This isn't a Switchvox or Salesforce support channel... |
| 16:59.43 | terretz | my apologies - this is my first dive into this level of asterisk software - I looked through the developer forums and found a few others with my inquiries but there was no direction. I'm really just reaching out for any guidance |
| 16:59.58 | terretz | What route should I go? |
| 17:00.24 | terretz | I don't know who controls the salesforce panel - because the switchvox is running asterisk - this is where I started |
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| 17:02.21 | [TK]D-Fender | terretz: Switchvox is a commercially support product only. You'd have to contact Digium or some other qualified service provider |
| 17:02.46 | bsdice | <PROTECTED> |
| 17:03.09 | terretz | Ok - I'll contact digium to see what kind of support they can give me. |
| 17:03.17 | terretz | thank you |
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| 17:04.02 | ghost75 | dont know nuttin about asterisk too :> |
| 17:04.31 | terretz | bsdice - I know ENOUGH - but my company is asking for something a little out of my knowledge scope |
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| 17:07.23 | newtonr | terretz, btw, there are Switchvox forums here http://forums.asterisk.org/, but yeah if you have active switchvox subscriptions you likely have some level of support from Digium |
| 17:07.57 | DevWork_ | Yea don't you buy commercial for support? |
| 17:08.27 | terretz | yeah |
| 17:09.46 | terretz | I checked the forums - I'll see what Digium can offer me |
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| 17:23.18 | illuder | hello, does anyone know if a2billing has an irc chatroom? |
| 17:25.29 | [TK]D-Fender | They don't. They do have a mailing list though |
| 17:25.43 | [TK]D-Fender | Go to their page. They'll list all of their support avenues |
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| 17:29.31 | illuder | tx, anyone has integrated a2b with asterisk here? |
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| 17:41.04 | bsdice | no a2b here, my users are freeloaders |
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| 17:42.53 | illuder | lol... |
| 17:43.17 | illuder | got it :) |
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| 18:15.11 | starks | is there any way to do shared lines with asterisk/switchvox? i want multiple phones to be able to dial out on a ring-all line |
| 18:15.22 | starks | with digium phones btw |
| 18:15.38 | starks | cisco seems to do this effortlessly |
| 18:16.09 | leifmadsen | starks: that's because Cisco phones are kind of dumb and controlled entirely from the CUCM |
| 18:16.19 | leifmadsen | however, you can do it with Asterisk SLA |
| 18:16.32 | paulc | starks: We did it with separate registrations for the ring-all number on each phone, setting caller ID on the peer |
| 18:16.40 | leifmadsen | +1 |
| 18:16.45 | paulc | Want to call out from 3499 Help Desk? Press that line button then dial out on it |
| 18:16.52 | leifmadsen | there are several ways to skin a cat, so yes, to answer your question, it is possible |
| 18:17.16 | paulc | The nice thing there is you can either have the name show as the actual person's name but with 3499 as the Caller ID number.. or put "Helpdesk" there for all registrations/phones. |
| 18:17.44 | starks | paulc, how? whenever i try to place the shared extension on multiple phones it unassigns the phone it was previously assigned to |
| 18:18.00 | paulc | You don't do it with shared extensions.. you have a separate registration for each phone.. |
| 18:18.11 | starks | explain |
| 18:18.20 | paulc | in my example, I've got helpdesk.paul, helpdesk.steve, helpdesk.john defined as peers in sip.conf |
| 18:18.48 | paulc | then in extensions.conf, 3499 calls a macro which uses SIP/helpdesk.steve&SIP/helpdesk.john&SIP/helpdesk.paul as a parameter for the destinations to ring |
| 18:19.07 | paulc | result: dial 3499 and all phones ring. First one to pick up gets the call. Other phones remain free for additional calls to 3499. |
| 18:19.11 | starks | i'm using a switchvox, i don't have that level of control unless i void my warrenty |
| 18:19.13 | starks | *warranty |
| 18:19.31 | paulc | Can you create a ring group in switchvox? |
| 18:19.38 | paulc | or are you using it to control phone provisioning too? |
| 18:20.07 | starks | switchvox controls provisioning unless i decide to use it solely for routing, which i think is insane |
| 18:20.56 | paulc | Ah.. Hmm.. Can you create multiple registrations/endpoints and assign each to a single phone? seems to me it should be doable, but I haven't had a ton of experience with Switchvox.. |
| 18:22.12 | starks | it might be possible to manually provision the lines in a way that avoids the switchvox bitching |
| 18:23.25 | starks | wouldn't mind doing that on a few d70s as opposed to hundreds of d50s |
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| 18:45.37 | DevWork_ | The D70's seem super priced compared to other ip phones on the market with what I believe is equivalent or even better features. |
| 18:45.45 | DevWork_ | Maybe the codec? |
| 18:47.19 | Penguin | A macro? To dial three phones? |
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| 18:59.38 | Katty | is there any big difference between using automon and automixmon? |
| 19:01.10 | Katty | maybe one only functions while the channels are bridged? |
| 19:04.26 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 19:11.50 | leifmadsen | Katty: different applications used for recording |
| 19:11.59 | leifmadsen | Monitor vs MixMonitor I believe |
| 19:12.03 | Katty | indeed |
| 19:12.11 | Katty | but how specifically are they different? |
| 19:12.11 | leifmadsen | use MixMonitor when you can |
| 19:12.16 | leifmadsen | Monitor is old and busted :) |
| 19:12.16 | Katty | because? |
| 19:12.22 | leifmadsen | less reliable, etc. |
| 19:12.36 | Katty | less reliable because? |
| 19:12.42 | Katty | peers at leifmadsen suspiciously |
| 19:12.47 | leifmadsen | because it was coded poorly? |
| 19:12.50 | Katty | because you cann adjust the volume and such? |
| 19:12.56 | leifmadsen | no, coding level issues |
| 19:13.01 | Katty | righto. |
| 19:13.08 | leifmadsen | like, crashy crashy boom boom |
| 19:13.30 | Katty | are there any scenarios in which i should be using monitor() instead of MixMonitor()? |
| 19:13.36 | leifmadsen | not that I have found |
| 19:13.40 | Katty | okies |
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| 19:43.52 | starks | DevWork_, rapid dial is worth it imho |
| 19:46.42 | DevWork_ | what is "rapid dial" |
| 19:47.39 | DevWork_ | Are those the BLF buttons? |
| 19:48.41 | WIMPy | Rapid dial must be the issue you get when you dial faster than Asterisk can cope and looses digits. |
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| 19:50.32 | Chainsaw | WIMPy: I had that. DTMF works better out of band. |
| 19:51.43 | WIMPy | Off course. |
| 19:51.43 | DevWork_ | What about phones like the GXP2160 from grandstream? |
| 19:52.03 | WIMPy | But it still happens, even on isdn. |
| 19:52.57 | WIMPy | On sip you only get the issue with feature transfers or disa. |
| 20:01.19 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-32-191.pn.at.cox.net) |
| 20:02.06 | igcewieling | Has anyone seen this message on a PRI (not BRI) and found a solution: "PRI Error on span 1: Received MDL/TEI management message, but configured for mode other than PTMP" PRI, NOT BRI |
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| 20:03.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:09.29 | WIMPy | I have never seen a PRI with TEI management. PTMP PRIs don't exist. Although they might be technically possible. |
| 20:10.02 | WIMPy | Who does that? |
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| 20:19.28 | igcewieling | WIMPy: CommOne, but I don't think they do that. |
| 20:19.36 | igcewieling | the circuit has been working for months |
| 20:20.21 | WIMPy | So what has changed? |
| 20:24.20 | Roland- | hello, is there a way to exclude failed calls from logs? |
| 20:28.14 | igcewieling | WIMPy: Nothing I can tell. time stamp on chan_dahdi.conf is June 25 of last year. I assume the carrier changed things, but I can't prove it. |
| 20:30.56 | WIMPy | It certainly sounds broken. |
| 20:31.10 | WIMPy | Do you have a trace for the curiosities collection? |
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| 20:34.58 | igcewieling | here you go http://pastie.org/8746548 |
| 20:37.49 | WIMPy | I don't see anything about TEI management there. |
| 20:38.18 | WIMPy | But it looks like you and the other end have different ideas about the lines state. |
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| 20:40.30 | igcewieling | it does not show up in pri debug, it shows up in the logs |
| 20:40.46 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 20:40.52 | igcewieling | there you go http://pastie.org/8746563 |
| 20:40.59 | WIMPy | It logs something it doesn't receive? |
| 20:41.15 | igcewieling | it logs something not shown in pri debug. |
| 20:42.03 | WIMPy | Anyway it looks like the other end tries to activate L2 while you think it already is activated. |
| 20:44.59 | igcewieling | I told them to insist the carrier prove out the circuit on-site with a T-BERD |
| 20:46.47 | igcewieling | Sounds like the answer to my question is "no" |
| 20:53.28 | WIMPy | A what? |
| 20:54.07 | WIMPy | Ah, a windows tablet. |
| 21:02.10 | igcewieling | a T-BERD is a handheld T-1/PRI protocol tester, they cost several thousand dollars and telco techs use them to verify ISDN protocol stuff. |
| 21:03.46 | WIMPy | Looks like it might be the successor of the Lite 3000. |
| 21:07.07 | ghost75 | somebody using sendfax from res_fax over t.38 ? |
| 21:08.41 | igcewieling | ghost75: never got that to work |
| 21:08.54 | ghost75 | got errors? |
| 21:08.58 | igcewieling | receive fax works just dandy for us, not so much sendfax |
| 21:09.14 | igcewieling | ghost75: It was 4 months ago, I don't recall the exact problem |
| 21:09.34 | ghost75 | i got message that provider refuses t.38 |
| 21:10.04 | igcewieling | the problem looks rather obvious than |
| 21:10.06 | igcewieling | then |
| 21:10.21 | ghost75 | but they say they do support |
| 21:10.45 | navaismo | word != fact |
| 21:11.10 | navaismo | good to see you igcewieling |
| 21:11.39 | ghost75 | but t.38 has also lot settings |
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| 21:32.58 | igcewieling | navaismo: I stopped coming around because none of *my* questions were getting answered. |
| 21:33.15 | navaismo | seriously? |
| 21:33.15 | igcewieling | anyway, ttfm |
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| 21:34.12 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 21:34.17 | ghenry | Hi all |
| 21:34.40 | navaismo | ok that was strange: searched for ttfm in google and it gave me->" To the fucking max" Im indeed Max |
| 21:34.41 | ghenry | I'm on 1.8 here and get incompatible destination on this http://paste.scsys.co.uk/308250 |
| 21:34.58 | ghenry | Asterisk doesn't look at the second audio offer in the SDP |
| 21:35.06 | ghenry | This used to work on Asterisk 1.4 |
| 21:35.37 | ghenry | Any pointers on a way to get Asterisk to look at the *whole* SDP before it rejects? |
| 21:36.05 | navaismo | can you paste the complete debug? |
| 21:36.58 | ghenry | navaismo: me? |
| 21:37.04 | navaismo | yes |
| 21:37.16 | ghenry | the SIP trace or Asterisk debug? |
| 21:37.43 | navaismo | run in the asterisk cli: sip set debug on |
| 21:39.21 | ghenry | yep, I've got that. two secs |
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| 21:41.03 | ghenry | navaismo: http://paste.scsys.co.uk/308254 |
| 21:41.20 | ghenry | If I set my ptime to 20 and not 30 then the audio offer gets merged and the call is OK |
| 21:41.48 | ghenry | but Asterisk never looks at the other audio line navaismo. 1.4 did |
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| 21:44.28 | navaismo | hmm have you tried with pedantic=yes|no if i recal that lloks for multiline headers, now not sure if that helps |
| 21:44.53 | navaismo | before 1.8 that settings was no by default |
| 21:47.50 | ghenry | cool |
| 21:47.52 | ghenry | will try |
| 21:51.01 | ghenry | navaismo: no difference |
| 21:51.31 | navaismo | with =no? |
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| 21:54.28 | ghenry | navaismo: yep |
| 21:57.44 | wdoekes | I don't think asterisk groks multiple m lines for different ptime values |
| 21:57.47 | wdoekes | see this: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086503.html |
| 21:58.52 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 21:58.57 | darkdrgn2k | What does "INVITE sip:2001@192.168.31.146 in-dialog | +SDP [5399.84s]" mean? |
| 21:59.12 | Qwell | darkdrgn2k: context required |
| 21:59.24 | darkdrgn2k | looking at a sip trace of a call |
| 22:00.52 | ghenry | wdoekes: weird as the same INVITE works on Asterisk 1.4 |
| 22:00.57 | darkdrgn2k | in the middle of a call |
| 22:00.59 | darkdrgn2k | i get this: |
| 22:01.00 | darkdrgn2k | http://pastebin.ca/2643509 |
| 22:01.01 | ghenry | It must either loop them or mobe on |
| 22:01.06 | ghenry | mvoe |
| 22:01.08 | ghenry | move |
| 22:01.08 | darkdrgn2k | actualy i get several sequences of that |
| 22:03.32 | ghenry | wdoekes: OK, I'll do it that way |
| 22:04.11 | darkdrgn2k | Qwell: any idea what that means? |
| 22:04.34 | darkdrgn2k | its an invite, trying, ok in the middle of a call |
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| 22:45.27 | navaismo | Is there a way to have the queue_log.log and the queue_log in the table using realtime, seems like i only can have one |
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| 22:56.07 | navaismo | ? |
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| 23:47.42 | JeffC_NN | Looking into solutions to get rid of quiet background noise on VoIP headsets (other agents talking). Would the dialplan function DENOISE() help? Does it work on ulaw codec? |
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| 23:51.46 | newtonr | JeffC_NN, I didn't know that function existed. I'd be interested to know if it works for you. |