IRC log for #asterisk on 20150209

00:11.35*** join/#asterisk malachi_constant (~root@unaffiliated/malachiconstant/x-837457)
00:11.40malachi_constantHey there folks.
00:13.01malachi_constantI'm getting 401 unauthorized in response to registration requests by my channel bank. Username and host= in my sip.conf match: http://pastebin.com/eWQtYnbn
00:14.02malachi_constantAny ideas?
00:14.13[TK]D-FenderYou have TWO devices with the same IP
00:14.52[TK]D-FenderFirst bad thing
00:15.06[TK]D-FenderSecond is a 101 grade mistake.
00:15.18malachi_constantEeep. I do.
00:15.21WIMPyYou register with the device? Isn't that the wrong way round?
00:15.22[TK]D-FenderDevices with a fixed host are not ALLOWED to register
00:15.34malachi_constantThe device registers with asterisk.
00:15.40malachi_constantOooh.
00:15.43[TK]D-FenderWhich is not allowed
00:15.45malachi_constantSo I need host=dynamic.
00:16.02[TK]D-FenderThe point of registration is to tell the other side where to find you.  If you're fixed you don't get to tell them where.
00:16.09WIMPyIf you know the IP, there's no need to register.
00:16.50malachi_constantGot it.
00:16.56malachi_constantThanks guys.
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08:25.02skirmishahi guys
08:25.26skirmishaany workaround to make asterisk match correct peer on inbound calls?
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08:25.48skirmishawithout using registration or username
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08:30.01skirmishaany ideas?
08:41.53Guggeskirmisha: sure, match on ip/port
08:48.09skirmishathat's the problem
08:48.19skirmishai have multiple peers with same ip and port
08:48.28skirmishaand its matching first found
08:49.02skirmishastrange thing is that behavior is not seen in older ver of asterisk
08:49.25skirmishai am currently testing on 13.1 and i am stuck with this issue and cant control incoming calls
08:52.19*** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou)
08:52.26skirmishais there a field that has priority and it is checked first?
08:52.36skirmishathen i can set under peer how to match it
09:00.58ChannelZHow can you have multiple peers on the same IP AND port?  How would you expect the traffic to get to the right one in the first place?
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09:02.50ChannelZAnd otherwise how, if you don't want to match by IP/port, OR without registration or username, do you expect it to work? Magic?
09:04.49skirmishaok lets say i have different users with different needs, but using same trunk for outgoing calls. Therefore i have separate peer for each user, but they are all with same ip. However incoming calls are matched to just wrong peer as there are many of them with same ip for outgoing calls, Thus i am not managed to control incoming calls. I have created separate peer for the incoming calls and pointing to correct context, but it is not matched as * match
09:04.50skirmishaagainst one of the outgoing peers/trunk set for users
09:07.11*** join/#asterisk CustosL1men (~CustosLim@unaffiliated/cust0slim3n)
09:09.03skirmishai have tested everything from type peer, user, friend, default ip. nothing is working as expected. on the outgoing trunks/peers there is only ip and type set nothing else
09:09.54AnonGirlskirmisha: reconsider your entire setup
09:10.01ChannelZAsterisk can match by IP AND port.  Many peers can have the same IP (if they are behind NAT for instance) but they will all have to have unique port numbers, otherwise it's impossible for Asterisk to send a call to the proper peer (the NAT wouldn't have any idea which device on the local side of the network to send the packets to.)
09:10.55ChannelZThe easiest way to do this is with registration, so the peers will get the IP and port for each peer, but if you reeeeally don't want to do that, then you've got to assign static port numbers yourself and put them in for each peer in sip.conf
09:11.25skirmishathere is no nat and why should i reconsider my setup. same setup with ast ver 1.4 works with no problem, now with latest 13.1 ver it does not. No patches made
09:14.00ChannelZWell like I said, you either need to make them register, or hardcode the IP & port yourself for each one
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09:17.55ChannelZWhat type were you using before?  (type=xxx in sip.conf)
09:22.00skirmishasame on all is peer
09:22.05skirmishaalso on ver 1.4
09:22.28skirmishathere is no nat and port is same as the ip
09:22.42ChannelZok.. so then was host=dynamic or did you have them statically configured?
09:23.06skirmishanope, host is ip of my peer and type is peer
09:23.15ChannelZAnd again, it's not possible to have multiple devices on the same IP *and* port
09:23.32skirmishabecause peer sends incoming calls from same ip it is matching one of the outgoing peers/trunks instead the one i configured
09:24.11ChannelZThat doesn't make sense.
09:24.26ChannelZWe'd really need to see some configs to figure out what the heck you've got going on
09:27.37skirmishaconfig is very simple - user a , normal user - > going to context with use peer/trunk 1, user 2 -> second context using second peer/trunk. Both trunks are identical - only host and type set. Difference is done in the context of users , So far no incoming call, that's only outgoing
09:28.27skirmishathird peer/trunk configured for incoming calls and forward to correct context - provider send call to us, asterisk match peer/trunk of first user
09:28.54skirmishai tried to allow anonymous incoming calls but that is not matched
09:29.21skirmishaon ver 1.4 that do the trick if i do not specify it, anonymous is matched and call flow is managed that way
09:29.34ChannelZThen you have something really screwed up for calls from your provider to be matching the peer of one of your devices.
09:30.39ChannelZAllowing guest calls means any call coming from someone that can't be matched to a configured user/peer will still be accepted (into a separate context) which again suggests your config is not correct.
09:31.01skirmishait is so called - asterisk multiple peers same ip
09:31.34skirmishaissue is that asterisk match against first peer/trunk set for user 1 outgoing calls
09:31.37ChannelZSo you're getting calls from your provider but have peers from your provider too?  That makes no sense.
09:34.03ChannelZShow us a sip debug of one of these incoming calls from your provider that is matching the wrong peer in your config.  And then show us the entry in sip.conf that it's *supposed* to be matching to.
09:34.23ChannelZ(just delete the secret=xxx)
09:34.38ChannelZ(if it even has one)
09:34.58*** join/#asterisk Darkerr (~Libor@static-84-42-235-44.net.upcbroadband.cz)
09:36.30Darkerrhello, is it possible to limit number of sip accounts (extensions) in asterisk?
09:36.44skirmishaone sec
09:37.15ChannelZDarkerr, errr... sip accounts and extensions are two totally different things
09:38.10ChannelZAnd in either case, you 'limit' them by only configuring the amount you need... ?  It's kind of a backwards question, can you elaborate?
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09:44.06DarkerrOk, Elastix web interface little confused me with Creating SIP account under Extensions menu, but its ok... I'd like to limit maximum Extensions that user can made in Elastix web interface... Since elastix doesnt have this feature I am looking for any possible solution
09:45.19ChannelZthat's an Elastix question
09:46.14ChannelZThere are no configuration limits within asterisk
09:46.36skirmishaok here are my observations, asterisk match first peer found in config that match against IP and port
09:47.01DarkerrNo limits in asterisk, ok that's what I needed to know. Thank you.
09:50.23ChannelZskirmisha, yes. Two physically different devices cannot have the same IP AND port.
09:50.29ChannelZThey can have the same IP and DIFFERENT ports
09:51.30eirirsor different IP and same port
09:51.40ChannelZgoes without saying
09:51.49eirirsnot for everyone hehe
09:52.35ChannelZConsider the port number as a child of the IP.  They go together, and the IP is matched first.
09:52.57skirmishagod damn
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09:55.49ChannelZ?
09:59.03skirmishathanks guys
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09:59.20skirmishalooks like i need to have this in mind when configure my setup
09:59.43skirmishaone question, can i match based on user agent?
10:00.12ChannelZno. That would be terrible anyway
10:03.41AnonGirlwut
10:09.14ChannelZSee http://burner.com/asterisk-primer/configuring-sip/ for a summary explanation of sip peer matching (with chan_sip, doesn't apply to PJSIP)
10:11.41ChannelZgoes to bed
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10:22.48pawieckii have a problem with pickupgroups. I have like 46 of them, and some of them are not working. What can be a problem?
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10:43.05catphishdoes asterisk still need dahdi on systems where no hardware is present?
10:43.36catphishor has all the software timing been moved into asterisk?
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12:21.14catphishseems its still needed for meetme
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13:30.03pawieckii've got a noob question: how can i forward port 5060 from remote host, to register softphone on my pc. I don't need voice transmission, just registering on server.
13:30.35[TK]D-FenderWhat do you mean FROM a remote host?
13:31.45[sID]pawiecki: iptables?
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13:50.15pawieckii have softphone on my pc, and server in remote location, i can access it via ssh. I want to register softphone to the server. Maybe i wrote it wrong.
13:52.00[TK]D-Fenderpawiecki: You should have SIP & RTP open on the server
13:52.02catphishi don't think you can forward udp over ssh, you could get asterisk to listen on tcp
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13:57.01[TK]D-Fenderpawiecki: if the only think you have open on your server is SSH : https://www.google.ca/#q=vpn+over+ssh
13:58.00catphishoh yeah, ppp over ssh is a good option if your client can support that
14:00.56pawiecki[TK]D-Fender: catphish: thanks, checking it out.
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15:07.02qakhanhi all, is there any way we can create 50 exts using a script
15:07.42[TK]D-FenderNo, the limit is 49
15:08.16qakhan[TK]D-Fender are you joking with me :)
15:08.27[TK]D-FenderYou'll have to wait for Asterisk 15 since 14 is already in lock-down.....
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15:09.33qakhanok what is that script
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15:10.59[TK]D-FenderWhat script?  Configuring Asterisk is YOUR job.
15:11.23[TK]D-Fenderqakhan: How many years have you been using asterisk at this point?  about 4 years based on a quick search...
15:11.51[TK]D-Fenderqakhan: Asterisk is configured by a bunch of dumb text files.  You cen generate those however you feel like and Asterisk won't care about the "how".
15:12.03MaliutaLap[TK]D-Fender: you know _the_ script ... it's called vim ;)
15:12.33[TK]D-FenderMaliutaLap: http://xkcd.com/378/
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15:13.32MaliutaLapqakhan: the issue with software as complex as asterisk is that there are so many different ways of doing things, and so many complex options that no script or GUI can hope to be actually useful to configure the whole thing
15:14.12[TK]D-FenderMaliutaLap: You mean like FreePBX? :)
15:14.30qakhan[TK]D-Fender so are you telling me we can have only 49 exts in an asterisk?
15:14.35[TK]D-FenderMaliutaLap: And yes you can script whatever you feel like and it'll be as useful as the peron writing it.
15:14.41MaliutaLap[TK]D-Fender: I know, I know ... but http://ars.userfriendly.org/cartoons/?id=20001111&mode=indexed
15:15.03MaliutaLap[TK]D-Fender: peron? or peon? ;)
15:15.07[TK]D-Fender[10:11][TK]D-Fenderqakhan: Asterisk is configured by a bunch of dumb text files. You can generate those however you feel like and Asterisk won't care about the "how". <------------
15:15.25[TK]D-Fenderqakhan: Generate config files.  Reload.  DONE.
15:16.03MaliutaLapwell generate config files. reload, test, tinker, rinse, repeat ;)
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15:17.32qakhan[TK]D-Fender and MaliutaLap i got you. but [TK]D-Fender said there is limit of 49 exts
15:17.55[TK]D-Fenderfacepalms
15:17.59MaliutaLappretty sure that's not something out Fender would say
15:18.14MaliutaLap[TK]D-Fender: can I borrow you palm?
15:18.32MaliutaLappilot if you got one ;)
15:18.37[TK]D-FenderMaliutaLap: I don't like where this is going...
15:19.31MaliutaLap[TK]D-Fender: it's just for my face - it's that dire that my palm->face alone isn't enough
15:20.12MaliutaLap[TK]D-Fender: if I could borrow the whole hand then a third hand typing might actually help
15:21.16MaliutaLapor I could use play it in poker - My hand is [TK]D-Fender's, the highest hand possible -> I win ... QED ;)
15:21.42[TK]D-FenderMaliutaLap: Face-high specifically...
15:22.00MaliutaLaptrue, true
15:22.17MaliutaLap[TK]D-Fender: why? what was your grotty mind thinking? ;P
15:23.02MaliutaLapconsiders the Zaphod Beeblebrox third arm. It might help me play guitar and keyboard betterer
15:24.12MaliutaLaptries for bed and sleep again - 40 hours is enough and the sillies are kicking in hard
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15:24.48[TK]D-FenderMaliutaLap: https://www.youtube.com/watch?v=iQ8ml7eENuI
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16:19.59ipalmerhello all, I'm trying to merge 2 conferences together, the conference names are dynamic so can be different every time, I know I need to use th originate application but am struggling to understand how to do it, any ideas anyone
16:21.07[TK]D-Fenderipalmer: Originate a Local channel that connects to the first side .... and on answer direct it to dialplan to connect to the other side
16:21.07WIMPyMaybe you should make the task a little harder and move the participants of the one conference in to the other.
16:22.07ipalmerD-Fender: thanks, that would allow all users in conference one to hear all users in conference 2?
16:22.35[TK]D-Fenderipalmer: It allows whatever you allow that channel to hear/say
16:22.50WIMPyI doubt it would be a good user experience to connect two conferences together. Unless one of them was muted.
16:24.01ipalmerD-Fender: ok, is this possible using AMI?
16:24.26ipalmerWIMPy: possibly not but it's what has been requested of me from the powers that be
16:25.19WIMPyOff course you can originate via AMI.
16:25.38WIMPyBut If you're already on AMI, you could also transfer the users.
16:26.15ipalmerWIMPy: I was toying with doing that but they want to be able to put them back into their original conferences afterwards
16:34.41DivideBy0in the ARI, I can start snooping, but to stop snooping, I should destroy the snooping channel, right?
16:34.55fileyes
16:35.05DivideBy0thanks
16:35.58*** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com)
16:49.27pabelangersnoop on to them, as they snoop on to us
16:52.59fileDivideBy0, how are you finding the mechanism for snooping?
16:57.45*** join/#asterisk areski (~areski@81.Red-83-53-30.dynamicIP.rima-tde.net)
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16:59.57*** join/#asterisk wanda__ (~xivo@modemcable094.94-70-69.static.videotron.ca)
17:00.13wanda__hi' IM looking for asterisk-fr?
17:00.33[TK]D-FenderAre you?
17:00.36WIMPyStart it.
17:00.45[TK]D-FenderYou seem to be about 3 charaters short...
17:01.09wanda__my profile http://valerie.dagrain.numerimoire.net/?page_id=25
17:02.10wanda__and currently here http://www.asterisk-france.org/threads/3430-Blog-XiVO-des-nouvelles-sur-des-tutos-et-contributeurs
17:02.25wanda__I need some advices about a presentation I prepare about XiVO
17:02.43wanda__I need advices from students in telecommunication :D
17:03.10wanda__and I would like to contact someone from Asterisk to thanks the community
17:03.13wanda__on the slide :D
17:03.15*** part/#asterisk catphish (~catphish@unaffiliated/catphish)
17:03.32wanda__that's it *____*
17:03.58DivideBy0file: still learning. I'm trying to snoop on a channel I just created before it enters my stasis app and haven't succeeded yet. I will report back more once I use it more
17:04.22wanda__So I listen your advices :)
17:05.28*** part/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com)
17:10.21stkochhi, I have a problem with asterisk 1.8.30.0 and voip calls with high packet loss. Sometimes a call works for 30 seconds, somtimes 120 seconds. After that the speech has dropouts and the call is aborted within 10 seconds. If there are dropouts an WARNING message comes with it: WARNING[2259]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to TAPI/1
17:10.25stkochHow to avoid call cancelation and let the call open to wait for lesser packet loss again?
17:10.57stkochI have tried a bit with jitter settings, it doesn't help.
17:11.31[TK]D-Fenderstkoch: use a codec that survives PL better like G.729 or iLBC, and adjust your rtp timeout.
17:11.34[TK]D-Fenderrtp.conf <-
17:11.38WIMPyThat means that TAPI/1 doesn't process the voice frames. Whatever kind of channel that is.
17:12.16stkochWIMPy: it's an lantiq soc in a router with openwrt
17:12.40stkochWIMPy: fxs port
17:14.09WIMPyYou probably need to tune your jitterbuffer.
17:18.04stkoch[TK]D-Fender: I have set rtptimeout=120; but the call terminates after 10 seconds
17:19.20stkochWIMPy: jbenable=yes; jbforce=yes; jbimpl=fixed;
17:21.47[TK]D-FenderI'm not even sure what TAPI technically uses for voice...
17:22.50stkoch[TK]D-Fender: https://dev.openwrt.org/browser/packages/net/asterisk-1.8.x/src/configs/lantiq.conf.sample?rev=32047
17:24.19[TK]D-FenderWhat hardware?
17:25.16*** join/#asterisk deranged (Bit@lolnerd.net)
17:26.01stkoch[TK]D-Fender: http://wiki.openwrt.org/toh/astoria/arv752dpw22
17:27.28stkoch[TK]D-Fender: lantiq danube
17:28.02[TK]D-Fenderstkoch: That's running * on OpenWRT itself right?
17:28.34[TK]D-Fenderstkoch: So the timeout is between the TAPI driver layer and * on the same device?
17:30.48stkoch[TK]D-Fender: on this router openwrt 14.09 is running. also on android softphone (connected to sipgate) there are dropouts after the same time but the call doesn't abort
17:31.12stkoch[TK]D-Fender: so could asterisk give the TAPI device dummy data as long the packet dropouts there?
17:31.31WIMPyIt's that TAPI device that can't handle jitter.
17:32.05WIMPyBut it's not the missing parts but the backlog thereafter that causes the problem. Well, or maybe not, but it's what causes that message.
17:32.30*** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de)
17:39.50[TK]D-Fenderstkoch: It shouldn't jitter being all internal.. That'd be a pretty serious CPU issue
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18:04.59rexwinwhen I run commands in cli it doesnot log in messages log, what needed to be done for logging output of asterisk cli output?
18:05.20*** join/#asterisk cyford (~allen@76.122.73.37)
18:06.33stkochWIMPy, [TK]D-Fender: it is possible to drop the backlog that causes the problem by asterisk? Have you seen the jitter settings in the lantiq.conf.sample link? Is it possible that asterisk does protect the lantiq fxs from jitter with silence sounds?
18:07.57[TK]D-Fender* doesn't do CNG
18:08.10[TK]D-FenderIt will buffer things a bit.. but no real concealment, iIRC
18:08.22[TK]D-Fenderrexwin: logger.conf <-
18:08.49[TK]D-Fenderrexwin: that will log OUTPUT.  Not sure you can log CLI COMMANDS you actually issue
18:10.12*** join/#asterisk ZerOlegend (~ZerOlegen@98-125-41-229.dyn.centurytel.net)
18:16.58crisedHas anyone used skype from a SIP phone_
18:17.00crised?
18:17.35stkoch[TK]D-Fender: but the beep beep beep sound comes right after call termination...
18:19.20[TK]D-Fendercrised: Probably not from a phone....
18:19.30[TK]D-Fendercrised: Better odds from a PBX
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18:21.36cyfordin the dialplan how do i get the name of owner of the extenstion
18:22.38mjordanthat concept doesn't exist.
18:23.10mjordanwhen you say "owner", what do you mean?
18:23.23[TK]D-FenderExtension = dialplan.  Nobody owns it.
18:23.57[TK]D-Fendermaybe "ls -la /etc/asterisk/extensions.conf" :)
18:24.05[TK]D-FenderThere's an owner :)
18:24.26mjordanheh
18:25.04[TK]D-Fenderconsiders looking up the regex to pull that back with a SHELL()
18:25.05cyfordallen = extenstion 200
18:25.25[TK]D-FenderYou're using those dangeroulsy vague terms again...
18:25.41[TK]D-Fenderdangerously*
18:25.59eirirsvague talk is just a trick to conceal the incompentence. :)
18:26.16cyfordi can run a system command to get it from sip_additional.conf
18:27.41[TK]D-Fendereirirs: It highlights it for me.  Showing no clue doesn't inspire to expect any more to come :)
18:27.56[TK]D-Fendercyford: You're in the WRONG CHANNEL for this......
18:28.16crised[TK]D-Fender: ok, Skype supports more phone numbers than voip.ms in my country, like local toll free numbers,
18:28.18cyfordno i was askin if there was a way in asterisk,
18:28.30[TK]D-FenderCyThose concepts are FREEPBX STRUCTURES
18:28.41cyfordbut after hearing there isnt i decided todo it in cli
18:28.46[TK]D-FenderThis is no anything inherently "Asterisk"
18:29.14[TK]D-Fendercyford: "database show" <----- it's in there
18:29.50[TK]D-Fendercrised: So go for it
18:30.28crised[TK]D-Fender: setup asterix... it takes work
18:30.33cyfordi thought it may be as easy as ${CALLERID(name)}   i just didnt know the variable,...  and when i dont know  and cant find info ,  i ask the pro's
18:30.38crisedI have a freebsd box where I could do it...
18:31.50[TK]D-Fendercrised: I didn't say you needed Asterisk for this
18:31.51WIMPycyford: We can't help you without knowing what you're actually looking for. But as [TK]D-Fender already said, it might not be an Asterisk question at all.
18:32.16[TK]D-Fender[13:29][TK]D-Fendercyford: "database show" <----- it's in there
18:32.19crised[TK]D-Fender: "not from a phone" ..
18:32.41[TK]D-Fendercrised: WE don't.  Doesn't mean you can't.
18:32.47crised[TK]D-Fender: oh you mean, just use skype... yeha it's an option, more uncomfortable
18:32.53[TK]D-Fendercrised: You're in #asterisk.  That's what we use.
18:32.55cyfordit was already answered,   i was looking for the name that is tied to an extenstion
18:32.59crised:)
18:33.16[TK]D-Fendercrised: Witha  Skype business acount you can access them via SIP.
18:33.26[TK]D-Fendercyford: And it's in there
18:33.30crised[TK]D-Fender: are they $$?
18:33.31WIMPycyford: Such a thing doesn't exist.
18:33.40cyfordin a dialplan instead of showing ${exten}  i want to show the name of it...
18:34.03cyfordyes i heard that, then i said i will do it through cli
18:34.12cyfordthanks
18:34.15[TK]D-Fendercyfordin a dialplan instead of showing ${exten} i want to show the name of it... <- This is a very poor way to try to express what you're looking for.
18:34.25[TK]D-FenderYou can do it DIRECT in the dialplan...
18:34.41WIMPyExtensions don't have names. The Extension is the "name".
18:34.41cyfordhow?
18:34.51[TK]D-Fender[13:32][TK]D-Fender[13:29][TK]D-Fendercyford: "database show" <----- it's in there
18:35.32cyfordok
18:35.35cyfordthanks
18:36.07[TK]D-Fenderprepares to re-paste it again for quadruple-nesting....
18:36.42cyfordi meant callerid of the exten WIMPy
18:36.52cyfordcallerid=Terri Jones <2020>
18:37.02WIMPyExtensions don't have caller IDs, either.
18:37.21WIMPyExtensions are destinations. Nothing else.
18:37.32[TK]D-Fendercyford: I hope you're not wasting time continuing to look at raw device config files....
18:37.53WIMPyMaybe they are something else in FreePBX, but that's not what it's about here.
18:38.25cyfordso u dont program a phone with an extenstion
18:38.38WIMPyWhat?
18:38.50cyfordso u dont program a phone with an extenstion
18:38.55stkoch[TK]D-Fender: it is possible it this error happens that asterisk restarts the channel?
18:39.12[TK]D-FenderxtI really have no experience with TAPI.
18:39.18[TK]D-Fenderstkoch: I really have no experience with TAPI.
18:39.26WIMPyPhones have users, not extensions.
18:39.33WIMPyExtensions can call phones.
18:39.47[TK]D-FenderMore accurately, phones have device configs.
18:40.11mjordanstkoch: Not trying to be discouraging, but you're using a third party channel driver whose site doesn't appear to have had an update pushed to it in the better part of three years. Your mileage is going to vary a lot.
18:40.20cyfordin the device config is a list of exenstions right lol?
18:40.24WIMPywas talking about phones, not Asterisk. But then we don;t really know what it's about.
18:40.43cyfordatleast all the phones i programed say it
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18:41.12WIMPyNo. Extensions are in extensions.conf and NOT in device settings.
18:42.07cyfordok,  i always calleded them dialplan exten.  vs user extens..    but ill change it sorry
18:42.27cyfordjust confusing going across platforms lol
18:43.21stkochso there is no solution that asterisk already provide? or the internet connection must good enough to have only little jitters...
18:44.17WIMPystkoch: It's really hard to say waht's going on there. I'd try to tune the JB settings. Especially a max size.
18:45.18cyford[TK]D-Fender  i can use the db lookup methode for sip and iax  and go the cli route for pjsips
18:45.38[TK]D-Fendercyford: Should work for ALL
18:45.48cyfordok thanks
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18:50.27drmessano"Asterisk - A big ball of ringy-dingy telly-welly..... stuff"
18:50.51drmessanoIf Digium marketing uses that, I want a some free something
18:53.40drmessano"BYOLOL - What do you mean you can't set up my e-mail on my Nokia 3360?"
18:56.24stkochWIMPy, [TK]D-Fender: csipsimple on android shows packet loss of 53.4 % !
18:56.55WIMPyWell, at that values, I wouldn;t waste any time on trying to make it work.
18:59.29fileraises eyebrow
18:59.46[TK]D-FenderGet a real system to run * on and a real FXO interface to match
18:59.55[TK]D-FenderSounds like this thing is jsut loaded to bits...
19:00.12WIMPyOr a working internet connection?
19:00.20drmessanoHow is my 12 yr old Linksys running OpenWRT not a real system?
19:00.25drmessanoScrew you guys, im going home
19:02.56drmessanoSomeone should make an adapter with a male<-->female 5.5mm/2.1mm coax plug that T's off to a pigtail with a regular in line and a mini USB on it for an RPi.. Then You could sell a router "upgrade" kit
19:03.08drmessanoregulator*
19:03.31WIMPyWhat?
19:03.31drmessano"Raspberry Pi Co-Processor"
19:04.19WIMPyLike the first ARMs were used as coprocessors to 6502s?
19:04.22drmessanoWIMPy, a Splitter that splits the DC going into a 12v router with a 5.5/2.1 plug, + regulator and USB to power an RPI
19:04.43WIMPyAh
19:04.54stkochWIMPy: I'm behind a proxy server with max. of 100 Mbit/s
19:05.57WIMPy100MBit should be good for 1000 calls. But you don't use a proxy for UDP, do you?
19:08.07*** join/#asterisk Alex_Bkash (7c06ecf5@gateway/web/freenode/ip.124.6.236.245)
19:09.28[TK]D-Fender~savemoney
19:09.28infobot<Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards.
19:09.43cyfordover 2000 calls for 100mbs full duplex
19:09.45*** join/#asterisk Navion (~Navion@rivendell/users/navion)
19:10.36WIMPyYou can get even more. Or maybe a little less. It all depends on your configuration.
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19:22.05stkochWIMPy: this connection is shared with hundred others with only one public IP for all
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19:42.26stkochThanks WIMPy [TK]D-Fender mjordan ...
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20:53.10DivideBy0I had an ARI originate hit my timeout and then hungup with a reason 16/normal, shouldn't it hangup with a reason of 18/expired or 102/timeout?
20:53.59WIMPyWhere/what timed out?
20:54.28DivideBy0the originated channel. you can give it a timeout when you originate it in the ARI
20:55.16WIMPyAs in it was ringing but not answered in the given time?
20:55.25DivideBy0yup
20:56.06WIMPyThat would be more like 19.
20:56.34DivideBy0agreed, 19 would be even better
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20:57.40DivideBy0I spent more time than I'd like to admit tracking it down. A better Hangup Cause would have made me look in the right place sooner
20:58.47WIMPyYou get used to that.
20:59.05overyanderCan you use t.38 udptl over a tcp session or do i have to use rtp for t.38 if i'm using tcp instead of udp?
21:02.04DivideBy0WIMPy: thanks. that's very reassuring
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21:09.27mjordanDivideBy0: interestingly enough, the core dial API looks like it attempts to set the hangup cause on the channel to 19. I'm not sure why that wouldn't have propagated up, as nothing else appears like it tries to set a different hangup cause on the channel on that path (although I did just do a cursory look).
21:09.37mjordanDivideBy0: what was the cause returned on the ARI event?
21:09.40rexwinI am trying to play an Auto-Attendant. but nothing happens
21:09.43rexwinhttp://pastebin.com/J9UqRafu
21:10.01mjordan"[Feb  9 07:06:12] WARNING[2373][C-00000000]: file.c:701 ast_openstream_full: File press-1 does not exist in any format"
21:10.06mjordanWhat do you think that means? :-)
21:10.44rexwinsome wav file is missing, i suppose
21:10.45WIMPyI'd prefer to play with a rea attendant rather than an automated one.
21:11.28rexwinWIMPy;-)
21:11.32mjordanrexwin: and where are the sound files located?
21:12.36fileoveryander, UDPTL is UDP and is a separate protocol from RTP.
21:13.27DivideBy0mjordan:  Received event: ChannelDestroyed cause_txt: Normal
21:13.48DivideBy0mjordan: I can create a simple reproducable test later tonight
21:13.58rexwingoogle says /var/lib/asterisk/sounds/e
21:15.19mjordanDivideBy0: that'd be handy. If you do, please attach the test case/logs to an issue at issues.asterisk.org, as that sounds like a bug to me
21:15.21DivideBy0mjordan: I looked in at my debug log, and there's no reason set. I only saw it on the ARI event
21:17.26DivideBy0ok. I'll post everything tonight
21:20.38Alex_BkashNot a room question. but any one can tell me how can i add javascript+ ajax effects on a existing SVG file??
21:21.45robmalLike svgjs.com ?
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21:24.39Alex_Bkashok let me check thanks
21:25.05overyanderfile, i'm having some issues with reliability. i was thinking that using TCP between my server and sip provider instead of UDP would increase stability, but I didn't know if I should use UDPTL or RTP for the t.38. UDPTL having 'UDP' in the name kinda implies that it's UDP and not TCP, but I wanted to confirm.
21:25.07DivideBy0mjordan: I tried to create a jira account, but nothing came through, and I cant reset my password. Do I need to be approved?
21:31.50rexwinthanks mjordan. it worked after I put a vm file
21:31.53wdoekesoveryander: all media is sent over udp; so yes, both rtp and udptl use udp.
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21:33.10overyanderso enabling tcp only applies to the sip packets and not the media packets?
21:33.15wdoekescorrect
21:33.27overyanderoh, didn't know that. thanks
21:33.28wdoekesif you want media over tcp, you'll need a tunnel (e.g. openvpn)
21:33.36wdoekesin rare cases, that can help
21:40.08overyanderthe t38pt_usertpsource option, does that set t.38 rtp to true? i just heard back from my provider, they only support rtp and not udptl. looking in my sippeers table (realtime sip) the only option related to t38 and rtp is t38pt_usertpsource.
21:40.29fileT.38 using RTP is not supported
21:41.30overyandereven with the FFA module?
21:41.45filecorrect
21:42.05overyanderwhat is the t38pt_usertpsource option for?
21:42.40fileUse the source IP address of RTP as the destination IP address for UDPTL packets
21:42.41file<PROTECTED>
21:42.41file<PROTECTED>
21:42.41file<PROTECTED>
21:43.19fileshamelessly copied from sip.conf.sample
21:46.03overyanderthanks file
21:47.13overyanderso, if my provider only supports RTP and not UDPTL and Asterisk only support UDPTL then why is it that some of the faxes are coming through using t.38? about 1 out of 20 are successful, the others time out.
21:48.14fileno idea - they may be incorrect
21:48.41fileor it's using standard RTP and not T.38 and by chance it's working
21:48.48filedunno!
21:53.25wdoekesoveryander: 'fax set debug on' and check the fax log configured in logger.conf
21:53.52filea SIP trace would also show what it is using
21:58.10wdoekesor that. but I bet the faxes come through over rtp, and the logs may help to figure out the reliability problems
22:00.16overyanderthe packet captures of the failed faxes and the one successful fax all show that it's UDPTL packets. I'm not sure what my provider is thinking.
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