| 00:00.10 | gruetzkopf | naah |
| 00:00.34 | robmal | Polen still has the same problem but switching to mail2fax is adapting very well. |
| 00:01.45 | gruetzkopf | its not like its ALWAYS that crowded on the line, but i couldnt resist looking at the pabx console when i saw 30 red "channel occupied" leds on my phone |
| 00:02.51 | robmal | Switch to mail2fax/fax2mail, disable 3way confs on the phone and switch to meetme/confbridge. |
| 00:03.14 | WIMPy | You use LEDs on the phone to monitor channels? |
| 00:03.43 | WIMPy | Using meetme or confbridge won't use less channels. |
| 00:04.01 | gruetzkopf | those peak time usages are actually confbridge calls |
| 00:04.40 | robmal | Depends if the calls are internal or external. |
| 00:05.07 | gruetzkopf | i do on my office phone, because it was really helpful to debug this mix of a philips KX-TDA100 and a asterisk box for anything more complicated than call incoming->phone rings |
| 00:05.24 | gruetzkopf | like conferencing and the iaxmodem | hylafax bank |
| 00:05.43 | WIMPy | robmal: How so? |
| 00:06.34 | robmal | WIMPy: If the calls are looping via the analog line they'll be blocking channels since using the external numbers but they might be internal for asterisk. |
| 00:07.36 | gruetzkopf | the philips box drops calls back to internal if you try to call our office via outgoing call |
| 00:07.44 | gruetzkopf | (from inside) |
| 00:07.50 | WIMPy | 1. I didn't read anything about anything analog and 2. I have no idea what kind of construction you're thinking of. Do you want to make internal calls via telco? |
| 00:08.20 | robmal | gruetzkopf: If that's how it reacts - great. |
| 00:08.38 | WIMPy | gruetzkopf: So Asterisk is under the Philips thing, not vice versa? |
| 00:09.01 | gruetzkopf | telco - E1 - philips - E1+4S0 - asterik |
| 00:09.04 | gruetzkopf | +s |
| 00:09.22 | robmal | WIMPy: Internal call goes to an external context, creates a channel which loops via Local/* to another context but the channel on the external still exists. |
| 00:09.27 | WIMPy | Interestingly enough, even some IADs catch external calls to your own numbers. |
| 00:10.22 | robmal | Still, 8 users creating 30 concurrent calls is an achievement. |
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| 00:10.42 | gruetzkopf | yep. and its only going to get worse |
| 00:10.56 | gruetzkopf | we'll soon have more people than phone lines |
| 00:11.05 | gruetzkopf | (like normal companies) |
| 00:11.08 | robmal | So maybe it's time to drop E1 and migrate to a normal SIP trunk? |
| 00:11.32 | chris349 | When I view my CDR each call has more than one entry. Is there any way to only have 1 CDR entry per call? |
| 00:11.38 | WIMPy | You get very good discounts for multiple E1. |
| 00:11.51 | robmal | chris349: Upgrade above Ast12 |
| 00:12.09 | robmal | WIMPy: Still paying way more than for a sip trunk ;-) |
| 00:12.12 | WIMPy | Depends on what you do to those calls. |
| 00:12.50 | WIMPy | Yes, but you also need a internet connection with enough bandwidth. And that's probably costing a lot more than a bunch of E1. |
| 00:12.55 | gruetzkopf | yeah, especially as our telco already put the frame relay->n*E1 equipment in out rack |
| 00:13.11 | gruetzkopf | we need the bit-transparent b-channels for legacy equipment |
| 00:13.29 | gruetzkopf | and 1*E1 was cheaper than 3*S0 |
| 00:13.34 | robmal | You need it just for fax lines to avoid the trouble. |
| 00:13.53 | gruetzkopf | yeah, fax at 9600bd is no fun |
| 00:13.58 | WIMPy | Wow |
| 00:14.11 | robmal | Well, if you're considering S0 than you do need a sip trunk. |
| 00:14.17 | WIMPy | Where did you get a single E1 that cheap? |
| 00:14.26 | WIMPy | Huh? |
| 00:15.17 | chris349 | robmal, Is there any setting that needs to be changed? |
| 00:15.48 | gruetzkopf | netcologne |
| 00:16.31 | WIMPy | Nice. Not an area where I'm active, unfortunately. |
| 00:16.33 | gruetzkopf | (compared to DTAG pricing, especially as team magenta tries to turn off their ISDN stuff by 2018) |
| 00:17.25 | gruetzkopf | whereas netcologne actually builds new ISDN equipment |
| 00:17.38 | WIMPy | Well, originally they wanted to switch off the PSTN 2010. Postponed over and over again. A real success story. |
| 00:17.58 | robmal | chris349: No, whole CDR/CEL backbone was rewritten at Ast12 and the guys here did a great job simplifying the parsing so that's my hint. |
| 00:18.05 | WIMPy | Versatel is the option here. |
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| 00:49.52 | Blackthorn | what distrubution is asterisknow built on? |
| 00:50.19 | WIMPy | CentOS |
| 00:50.50 | Blackthorn | well good :) thats what i'm using |
| 00:58.02 | chris349 | robmal, I am on Asterisk 13 and if the call rings 3 extensions and then forwards somewhere else after no answer I have 4 entries in the CDR with the same uniqueid |
| 01:00.02 | WIMPy | Maybe you should disable cdrs for unanswered calls. |
| 01:06.55 | robmal | Go slow. Dial(SIP/1&SIP/2... or some queue? |
| 01:10.52 | robmal | Also go quick, it's past 3am here so. |
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| 01:32.38 | knightr | WIMPy, I could not reply this morning before leaving for work and now my scrollback buffer has eaten the reply I got about Asterisk sound filesbut I believe it was from you... If you, thank you, if not, sorry for the trouble... |
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| 03:29.39 | ketas | damn, _[+0-9]. should also match to "9", no? |
| 03:37.54 | igcewieling | no. |
| 03:38.06 | igcewieling | it would match 9 and one of any other character |
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| 03:54.21 | ketas | igcewieling: well, but _. is bad, isn't it? :P |
| 03:55.35 | ketas | i basically want \d+ |
| 03:55.49 | ketas | but there's no pattern for that... |
| 03:57.14 | igcewieling | you need two patterns. a singel digit patter and a multi digit pattern. Ugly, thats life. |
| 04:00.08 | [TK]D-Fender | depends |
| 04:00.26 | [TK]D-Fender | if that is being dialed direct from say a SIP phone then no, he can do it in one |
| 04:00.40 | [TK]D-Fender | If it's in an auto-attendant, then no. |
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| 04:51.45 | cmendes0101 | I'm recording a call with MixMonitor but then trying to use Chanspy to play a recording. Chanspy works but MixMonitor doesn't record the Chanspy audio. Anyone have suggestions to get that to record? |
| 04:51.58 | cmendes0101 | Chanspy is using params bBqE |
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| 05:13.06 | _pepo_ | Hi friends. Within the dialplan: How I can know how long it lasted a SIP call? |
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| 05:15.04 | _pepo_ | Hi friends. Within the dial plan: How I can know how long it lasted a SIP call? |
| 05:18.49 | [TK]D-Fender | CDR <- |
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| 06:53.57 | _pepo_ | Thanks so much |
| 06:54.07 | _pepo_ | CDR is ok!!! |
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| 07:58.20 | carrar | LETS CDR!! |
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| 08:13.36 | stefan27 | is there any way to tell Dial to copy all variables AND their scope to the target channel? So that if I have SIP_A with Set(_VAR1=foo) and Set(VAR2=moo) and Dial(LocalX...) then LocalX would have VAR1 set to foo on _ scope and VAR2 set to moo on normal scope? |
| 08:13.48 | frenk77 | Hello i have some problem with gdb and core dump. If i use gcore -o file PID, then open in gdb it is ok, i see every function any "?? ()" after bactrace adresses. But problem is when asterisk crash by call abort in code or by kill -SIGABRT PID. I open gdb with coredump and it writes many "warning: .dynamic section for "/usr/lib64/asterisk/modules/app_cdr.so" is not at the expected address (wrong library or version mismatch?)" then i have 90% of backtraces w |
| 08:14.41 | stefan27 | Your last message cut off at "...backtraces w" |
| 08:16.48 | frenk77 | then i have 90% of backtraces with "?? ()" so i cant debug :( What can cause this ? |
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| 09:43.27 | nunne | I get Handshake failure 40 when issuing "openssl s_client -connect provisioning.officenetwork.se:443 -cipher "DES-CBC3-SHA" -tls1". Any ideas? I have added DES-CBC3-SHA to my cipherlist in apache. And it shows up at the bottom of the cipher lists on ssllabs test. Is it because I use FS perhaps? |
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| 11:11.18 | xpheres | Hello, I managed to make TLS work, but zoiper does not allow me to make outbound calls, I receive this error: [2015-10-01 12:06:24] NOTICE[9330][C-00000005] chan_sip.c: Failed to authenticate device "Raul 103"<sip:103@xpheresserver.hopto.org;transport=TLS>;tag=87eaa77f |
| 11:11.24 | xpheres | with other softphones works |
| 11:11.28 | xpheres | anyone knows why? |
| 11:13.42 | frenk77 | sip set debug peer 103 |
| 11:13.46 | xpheres | ok |
| 11:14.23 | frenk77 | then try call and paste output |
| 11:15.08 | xpheres | how do I set debug off of the peer? |
| 11:15.30 | frenk77 | sip set debug off |
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| 11:15.39 | xpheres | ok |
| 11:16.38 | xpheres | http://pastebin.com/mTsb0HM8 |
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| 11:18.09 | xpheres | should I have a client certificate for every peer / extension? |
| 11:18.22 | velus | hello is there a web based billing system other than a2billing that will allow did billing and also minutes billing in php please? |
| 11:19.35 | xpheres | or should I have a client certificate that is the same for every extension? |
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| 11:20.43 | carrar | [2015-10-01 12:14:56] WARNING[9330][C-0000000f]: chan_sip.c:16506 check_auth: username mismatch, have <107>, digest has <103> |
| 11:21.03 | velus | xpheres, i would try with same cert for every client peer/extension then if that doint work seperate |
| 11:21.22 | xpheres | but I can not call from zoiper from one extension to another |
| 11:21.36 | xpheres | why |
| 11:21.43 | frenk77 | add sip show peer 103 |
| 11:21.53 | frenk77 | and sip show peers |
| 11:22.47 | xpheres | http://pastebin.com/mTsb0HM8 |
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| 11:24.10 | xpheres | please check again |
| 11:24.10 | xpheres | http://pastebin.com/mTsb0HM8 |
| 11:24.14 | xpheres | I added sip show peerrs |
| 11:25.15 | frenk77 | 103 and 107 has same ip. can you add your sip config files ? :) |
| 11:25.53 | frenk77 | this can cause bad peer matching by ip |
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| 11:26.15 | xpheres | they have same ip because one is in my computer and the other in my mobile both connnected to my router |
| 11:28.00 | xpheres | ok solved |
| 11:28.11 | xpheres | I can call from my mobile internet , but not from local network |
| 11:32.21 | xpheres | should be media and signal from R port on in zoiper? |
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| 11:58.09 | Puck` | I'm curious if any of you guys used any speech syntesis software to generate .wav files to be played through an automatica .call file? |
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| 12:12.38 | flo291288 | Hello, Is possible to set one variable in an other channel like Set(foo=123, SIP/1234), to dont use SHARED ? |
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| 12:15.13 | Rico | hi |
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| 12:21.15 | [TK]D-Fender | Puck`: I'm sure many have used those pieces in any number of combinations. This specific one isn't really special vs any other |
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| 12:50.26 | frenk77 | xpheres: can you send sip config files ? it is problem only when bad configured :) |
| 12:51.30 | Puck` | [TK]D-Fender: I was just wondering which software was tried and works for english, sure, I can run through all, but thought asking would help me |
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| 12:59.03 | [TK]D-Fender | Puck`: I haven't heard of any that don't work with English. |
| 13:00.04 | carrar | AT&T Natural voice is very nice |
| 13:00.20 | carrar | http://www.wizzardsoftware.com/text-to-voice.php |
| 13:01.33 | Puck` | thank you carrar (: |
| 13:02.02 | Puck` | I need a linux shell software so I can generate the wav on the fly |
| 13:03.07 | [TK]D-Fender | https://www.google.ca/#q=asterisk+tts <- takes 2 words in a google search to turn up a half dozen quick solutions |
| 13:03.52 | carrar | What is this thing called google you speak of!! |
| 13:05.43 | Puck` | [TK]D-Fender: you are really in a helpfull mood, aren't you? (: |
| 13:06.42 | lvlinux | Puck`: Festival can be made to do some really nice tts. |
| 13:06.47 | lvlinux | Puck`: http://www.cstr.ed.ac.uk/projects/festival/morevoices.html |
| 13:07.01 | lvlinux | But it's a royal pain to set it up with the best voices. |
| 13:07.19 | [TK]D-Fender | Last I heard Festival was at the bottom of the list and Lumenvox was the "go-to" choice |
| 13:07.23 | Puck` | lvlinux: I was looking in to festival, saw some other examples as well, thus the reason I came here, seeking advice from some professionals who have used something like this before (: |
| 13:07.35 | Puck` | [TK]D-Fender: now, thank you! That's the advice I was looking for |
| 13:08.21 | lvlinux | [TK]D-Fender: the reason Festival is considered to be the bottom is probably because out of the box the voices are totally garbage 1980s robot sounding. But the new HTS voices are very very good. |
| 13:08.55 | lvlinux | But like I said it's a pain to setup to use them. |
| 13:09.27 | [TK]D-Fender | Sounds qualified... |
| 13:09.36 | lvlinux | Puck`: check out that demo page and you'll see some really nice natural sounding output. |
| 13:09.55 | Puck` | thank you very much lvlinux |
| 13:10.11 | lvlinux | np good luck :-) |
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| 14:44.35 | megazloj | hello everyone! |
| 14:45.04 | megazloj | can i ask for advice? |
| 14:46.08 | newtonr | Yup |
| 14:46.22 | newtonr | Don't take any wooden nickels |
| 14:46.52 | megazloj | ok, i have asterisk installed from repositories (11 version) on ubuntu. also i've setted up few sip peers |
| 14:47.12 | megazloj | and when i'm typing commang sip show channelstats, it show all zeros |
| 14:47.24 | megazloj | but why? |
| 14:48.12 | megazloj | <PROTECTED> |
| 14:48.12 | megazloj | <PROTECTED> |
| 14:48.56 | newtonr | Hmm |
| 14:49.37 | megazloj | rtp is 100% passes asterisk |
| 14:49.51 | megazloj | because of directmedia=no |
| 14:52.10 | newtonr | I think problems with RTCP could explain some zero'd out fields, but I don't think all of the fields rely on RTCP values |
| 14:52.25 | newtonr | what version of 11 are you on? |
| 14:52.36 | newtonr | "core show version" |
| 14:52.39 | newtonr | asterisk -V |
| 14:52.52 | megazloj | Asterisk 11.7.0~dfsg-1ubuntu1 built by buildd @ lamiak on a x86_64 running Linux on 2013-12-24 06:02:10 UTC |
| 14:53.34 | megazloj | also i've found a topic on asterisk forum for same problem, but no answer was given there |
| 14:53.43 | megazloj | http://forums.asterisk.org/viewtopic.php?f=1&t=90881 |
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| 14:56.29 | newtonr | That is pretty old, try 11.19 and see if the issue still occurs. If it does then file an issue on issues.asterisk.org/jira with all your details, configuration and logs of a call along with the 'sip show channelstats' output |
| 14:59.17 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-yhakoanphizevihd) |
| 14:59.29 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-avdqvthedhiarpgf) |
| 14:59.40 | megazloj | ok, i'll try to upgrade |
| 14:59.58 | megazloj | for cat /etc/issue |
| 14:59.58 | megazloj | Ubuntu 14.04.3 LTS \n \l |
| 15:00.11 | megazloj | 11.7.0~dfsg-1ubuntu1 - this is last provided in repos |
| 15:01.51 | igcewieling | megazloj: build it yourself. |
| 15:01.56 | [TK]D-Fender | Then ditch their repos |
| 15:01.58 | [TK]D-Fender | 11.19.0 (2015/08/07) |
| 15:02.12 | hjf | my itsp accepts all sorts of codecs: alaw, ulaw, g729 etc... |
| 15:02.25 | hjf | but i want to stick with either alaw or ulaw |
| 15:02.44 | hjf | because some clients MAY try to use credit card terminals on this line |
| 15:03.01 | megazloj | ok, thanks for advice |
| 15:03.02 | hjf | but i'm not sure if my country uses alaw or ulaw? i want to avoid transcoding |
| 15:03.42 | hjf | does it matter? |
| 15:04.15 | hjf | i mean I'd like to "keep voice packets intact" from the client's device to the phone company |
| 15:04.19 | glNito | ulaw for US and alaw for EU |
| 15:04.21 | igcewieling | hjf: if your customer tries to use CC machines on a VoIP line they will fail. |
| 15:04.32 | hjf | igcewieling: i just tried one and it worked! |
| 15:04.39 | igcewieling | hjf: tried it just once?> |
| 15:04.52 | hjf | 3 times: tried a charge, cancellation, and batch close |
| 15:05.26 | igcewieling | hjf: congrats, you made it work under test conditions. Let us know how it works in actual production. |
| 15:05.42 | hjf | i mean this isn't a "voip line" |
| 15:06.02 | hjf | this is a phone line, from an actual phne company, that just happens to be provided over voip |
| 15:06.16 | glNito | I've used CC machines on ATAs |
| 15:06.33 | glNito | POTS line from AT |
| 15:06.44 | glNito | AT&T with an ATA |
| 15:06.54 | hjf | i mean if it's G.711 it's the same as the phone company's line |
| 15:07.01 | hjf | no? |
| 15:07.19 | hjf | aren't all analog lines converted to digital at the central office, using alaw or ulaw anyway? |
| 15:07.22 | igcewieling | hjf: no, it isn't. It is packetized g711 with variable latency. |
| 15:07.25 | stefan27 | Hmm I've never really used sip show channelstats... It's based off rtcp repors? Is it reliable? |
| 15:08.14 | hjf | igcewieling: i see. well in this case i'm getting the data over a fiber connection in its own VLAN so maybe they're doing some sort of priorization |
| 15:08.59 | igcewieling | hjf: if you mean FIOS, that isn't really VoIP. |
| 15:09.35 | hjf | no |
| 15:09.38 | hjf | it's GPON |
| 15:09.49 | igcewieling | so nothing I said applies. |
| 15:10.10 | hjf | 30/6mbit data on a VLAN, voip "trunk" on another VLAN |
| 15:10.25 | igcewieling | everything I said applies to CC over SIP (well RTP) |
| 15:10.31 | hjf | it is SIP |
| 15:10.44 | igcewieling | Try it and get back to us. |
| 15:11.00 | hjf | i have an ATA connected to an Asterisk server, which is connected to said VLAN |
| 15:11.10 | hjf | wanna make it worse? ok here is the full setup: |
| 15:11.11 | igcewieling | might want to google people's experiences trying to run credit card over VoIP. |
| 15:11.19 | glNito | so just register the line in your ATA |
| 15:11.39 | hjf | i'm at my test lab, 30ms away from the router, connected over L2TP |
| 15:11.50 | hjf | from there, i'm forwarded to the Asterisk server |
| 15:11.57 | hjf | inside vmware esxi |
| 15:12.09 | hjf | which then sends the data over to the provider |
| 15:12.16 | hjf | on the same interface, on a different VLAN |
| 15:12.36 | hjf | to the trunk located in Buenos Aires 1000km away from me |
| 15:12.41 | hjf | and yet, somehow, it all works |
| 15:15.07 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 15:26.11 | *** join/#asterisk glNito (6ca1d582@gateway/web/cgi-irc/kiwiirc.com/ip.108.161.213.130) |
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| 16:28.23 | Kobaz | axeterisk |
| 16:31.17 | litn | same |
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| 17:37.06 | Reapster | heya, im wanting to redirect a channel (Say PJSIP/1-xxx) to another extension via ARI, maybe yyy@from-internal..but I see the only parameters are channelId and endpoint (Which has to be an Asterisk/same as channel endpoint?). Am I being stupid and should I instead create a bridge and add the existing channel + originate a new one to the yyy@from-internal extension and add it? |
| 17:37.29 | file | is the channel in your Stasis application? |
| 17:37.42 | Reapster | yeah all should be in Stasis app |
| 17:37.58 | file | continue can be used to send it back into the dialplan |
| 17:38.05 | file | but it leaves your Stasis application |
| 17:38.24 | file | otherwise you have to use a Local channel and do what you said |
| 17:40.15 | Reapster | Local/yyy@from-internal as an 'endpoint'? |
| 17:40.20 | file | yes |
| 17:40.30 | Reapster | ah perfect, sorry I see that is pretty clear on the wiki now |
| 17:40.39 | file | it has to be answered though, originated channels currently can not have anything done to them until answered |
| 17:43.47 | Reapster | fair enough, thanks so much for your help with this |
| 17:44.22 | file | goooooood luck |
| 17:44.29 | Reapster | hah thanks |
| 17:52.00 | hjf | i'm running with sip set debug on, but i don't see inbound caller id from the trunk... do they just not send it, or do i need to set an option to request it somehow? |
| 17:56.44 | robmal | core show channel xxx |
| 17:56.52 | [TK]D-Fender | If you can't see it in the INVITE, then they're just not sending it. |
| 17:57.04 | *** join/#asterisk eofster (~eofster@cologne.eofster.com) |
| 17:59.02 | hjf | [TK]D-Fender: yes, the invite is just from/to same number, not my external id |
| 17:59.24 | [TK]D-Fender | Lets see a call |
| 18:00.33 | hjf | [TK]D-Fender: http://pastebin.com/RBus8t0S |
| 18:05.00 | [TK]D-Fender | almost the same # twice |
| 18:05.05 | [TK]D-Fender | just with a prexi |
| 18:05.09 | [TK]D-Fender | prefix* |
| 18:05.15 | [TK]D-Fender | And... it is what it is... |
| 18:08.18 | hjf | [TK]D-Fender: maybe i have to call them and ask for caller id... |
| 18:09.29 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 18:11.31 | thiagoc | hi, i'm having a problem with asterisk where I make a "channel originate SIP/100 application musiconhold" and the audio have some glitch at every 2-5 seconds |
| 18:12.26 | thiagoc | on rtp debug I see this => http://pastie.org/10454537 |
| 18:18.49 | thiagoc | various "Got RTP packet from" |
| 18:19.50 | thiagoc | so I think that Asterisk isn't processing rtp properly |
| 18:48.45 | bipolar | I'm trying to find out if I can access a variable that shows up in the asterisk log file. In this line: [2015-10-01 14:45:43] VERBOSE[11345][C-0000c5a6], I'm trying to find out if the 'C-0000c5a6' is accessable as a variable in the dial plan. Does anyone know? |
| 18:55.45 | *** join/#asterisk vandyk (~vandyk@191.187.221.171) |
| 19:00.33 | rmudgett | bipolar: AFAIK it is not accessible by the dialplan |
| 19:09.04 | *** part/#asterisk velus (~Velus@cpc8-know14-2-0-cust829.17-2.cable.virginm.net) |
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| 19:29.11 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 19:29.15 | bipul | Hello |
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| 19:55.26 | igcewieling1 | bipolar: Your best bet is to use, I thnik, ${CHANNEL(linkedid)}, that is as close as you'll get to a unique string per call |
| 20:06.44 | robmal | As long as you use ast >12 |
| 20:07.09 | robmal | Earlier odd things happened during transfers. |
| 20:13.36 | igcewieling1 | I believe it would be ast >= 1.8 |
| 20:15.53 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 20:17.59 | robmal | Maybe so, but a lot of good things happened with asterisk 12. |
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| 21:08.06 | igcewieling1 | I expect to jump from Asterisk 11 to Asterisk 15 when the time comes. |
| 21:17.40 | *** join/#asterisk gk (~gk@unaffiliated/gk) |
| 21:19.45 | gk | hello, when doing multi tenant setup can (and how?) I have two same logins (for example their internal numbers) for two different sip phones in two different companies? every company has it's own sip domain |
| 21:20.33 | igcewieling1 | gk: people who do such things use a prefix aka siteid as part of the login, etc. |
| 21:20.34 | gk | can I set up a phone in asterisk with for example [12@company1] section title? or some other way? |
| 21:21.26 | gk | yeah, but can I do it without prefixing the login, something like virtual hosting for HTTP? |
| 21:21.53 | igcewieling1 | gk: no, you CANNOT have multiple SIP devices with the same login ID. |
| 21:22.44 | gk | igcewieling1: ok, thank you |
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| 21:43.45 | Puck` | on asterisk 1.8.13 when I do core show settings, I don't see a sounds directory, is that okay? I'm trying to play a file which is placed in a .call file, but asterisk reports that it's not found, I placed it in /var/lib/asterisk/sounds |
| 21:46.47 | [TK]D-Fender | because it isn't a separate thing |
| 21:46.55 | [TK]D-Fender | <PROTECTED> |
| 21:46.56 | [TK]D-Fender | ^ |
| 21:47.11 | [TK]D-Fender | If it says it's not there.. then the file is not there and readable in the form it should be |
| 21:47.17 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
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| 21:48.33 | Puck` | fixed, sound files are in /usr/share/asterisk/sounds/ |
| 21:49.25 | igcewieling1 | you must have installed from package? |
| 21:49.30 | Puck` | yes |
| 21:49.31 | Puck` | debian |
| 21:49.51 | igcewieling1 | the default is /var/lib/asterisk/sounds |
| 21:50.01 | [TK]D-Fender | Well.. that * compiled standard |
| 21:50.08 | [TK]D-Fender | distro packaging takes a few liberties |
| 21:50.08 | socomm | Can anyone recommend a good physical SIP phone to use for my home office? |
| 21:51.21 | *** join/#asterisk bluemerlin (~bluemerli@office-nat.westpoint.gradwell.net) |
| 21:51.21 | [TK]D-Fender | Polycom is best on quality. Aastra wins out on "lazy config" and are nicer with regards to speed-dials, etc |
| 21:58.04 | ChkDigit | How are the Digium phones stacking up? |
| 22:00.52 | [TK]D-Fender | I've got my gripes. |
| 22:01.03 | [TK]D-Fender | They come in maybe 4th or so on my list |
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| 23:23.48 | *** part/#asterisk kharwell (kharwell@nat/digium/x-greqakdedbpwkocy) |
| 23:25.40 | ketas | hmm, i wonder how to make my dialplan ( http://ketas.si.pri.ee/asterisk/extensions.conf ) less crazy |
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