| 00:00.42 | rrittgarn | Question to interject into your heated debate. CELGenUserEvent... do i have to register a custom type before it will actually get picked up and stored? or should USER_DEFINED just work? |
| 00:02.14 | MarkS- | [TK]D-Fender: thanks for pointing in what I should have discovered before asking |
| 00:02.45 | [TK]D-Fender | MarkS-: Remember... proof IS as easy as it seems. Got packets? No? You really wondering? :p |
| 00:03.56 | MarkS- | I should've trusted (and looked closer at) results from tools like ngrep for checking the direction of packets |
| 00:05.54 | [TK]D-Fender | no packets, no workee...... |
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| 05:07.19 | CRCinAU_ | Hai all. |
| 05:07.27 | CRCinAU_ | I'm trying to figure out the cause of this: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 05:07.47 | CRCinAU_ | this is on an outgoing call to a peer that I register with via a register line in sip.conf |
| 05:08.26 | CRCinAU_ | in the logs, I see this: |
| 05:08.28 | CRCinAU_ | <PROTECTED> |
| 05:08.32 | CRCinAU_ | <PROTECTED> |
| 05:08.35 | CRCinAU_ | [2014-01-22 16:03:26] WARNING[4699][C-000000f6]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 05:08.39 | CRCinAU_ | <PROTECTED> |
| 05:08.41 | CRCinAU_ | <PROTECTED> |
| 05:09.01 | CRCinAU_ | and interestingly, even though 'sip show registry' shows the peer as registered, a 'sip show peers' shows this for the peer: |
| 05:09.08 | CRCinAU_ | 72275/72275 (Unspecified) a 0 Unmonitored |
| 05:09.25 | CRCinAU_ | ie Host = (Unspecified). |
| 05:09.48 | CRCinAU_ | This doesn't seem to make sense, as 1) There is a host=blah in the peer config in sip.conf, and 2) I register successfully to the peer. |
| 05:10.05 | CRCinAU_ | This is using asterisk 11.7 - and seems to be a very recent problem |
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| 05:10.48 | [TK]D-Fender | your registering out has nothing to do with a peer not having a host. |
| 05:11.49 | [TK]D-Fender | If that is a peer for outgoing... you should have the host to reach them |
| 05:11.51 | CRCinAU_ | doesn't the Host get set with a successful registration? or is that only for incoming registrations? |
| 05:12.04 | [TK]D-Fender | the pieces you're describing don't fit you.... |
| 05:12.12 | [TK]D-Fender | yet* |
| 05:12.24 | CRCinAU_ | either way, there IS a host=blah in the sip.conf definition of the peer |
| 05:12.30 | [TK]D-Fender | you are registering to a provider? |
| 05:12.36 | [TK]D-Fender | show us |
| 05:12.53 | [TK]D-Fender | "unspecified" is what you gt with "host=friend" |
| 05:13.02 | [TK]D-Fender | host=dynamic |
| 05:13.04 | [TK]D-Fender | oops |
| 05:13.05 | CRCinAU_ | type=friend you mean? |
| 05:13.11 | [TK]D-Fender | may bad... dynamic |
| 05:13.19 | CRCinAU_ | I have type=peer, host=my.sip.server |
| 05:13.22 | [TK]D-Fender | that is the only time it should be "unspecified" |
| 05:13.43 | [TK]D-Fender | "sip show peer 72275" |
| 05:14.01 | [TK]D-Fender | also make sure you've actually reloaded any config changes... |
| 05:14.08 | CRCinAU_ | ToHost is set in that. |
| 05:14.13 | [TK]D-Fender | And prove your DNS is working on the server since you say you're using a hostname |
| 05:14.48 | CRCinAU_ | I can ping it from the server via DNS |
| 05:14.57 | CRCinAU_ | so that implies DNS + network is good. |
| 05:15.05 | CRCinAU_ | and as I said, I can register to the same host. |
| 05:15.20 | [TK]D-Fender | Addr ->_IP is what has to be set there |
| 05:15.31 | CRCinAU_ | ? |
| 05:15.44 | [TK]D-Fender | no to Host |
| 05:15.57 | CRCinAU_ | hmmmm |
| 05:15.57 | CRCinAU_ | <PROTECTED> |
| 05:15.58 | CRCinAU_ | <PROTECTED> |
| 05:16.03 | [TK]D-Fender | [00:14]CRCinAU_ToHost is set in that. <- not the relevant field |
| 05:16.10 | [TK]D-Fender | clearly bad |
| 05:16.28 | CRCinAU_ | so if it works for the system, why can't asterisk resolve it? o_O |
| 05:16.39 | CRCinAU_ | I've noticed if I restart asterisk, it works again |
| 05:16.51 | [TK]D-Fender | I'm not seeing actual configs and dumps |
| 05:17.00 | [TK]D-Fender | Which sounds liek "non-applied changes" |
| 05:17.08 | CRCinAU_ | so its almost like its forgetting the resolution, but not looking it up again |
| 05:17.18 | CRCinAU_ | I've done a 'sip reload' many times |
| 05:17.22 | [TK]D-Fender | I'm not going to guess that's what the difference is at this point |
| 05:17.52 | [TK]D-Fender | UNSPECIFIED has a very singular use. |
| 05:17.54 | CRCinAU_ | in fact, doing a 'sip reload' doesn't populate the Addr->IP field at all |
| 05:19.44 | CRCinAU_ | config snippets: http://fpaste.org/70549/03679741/ |
| 05:19.59 | CRCinAU_ | its a very basic config |
| 05:20.42 | [TK]D-Fender | Also change your syntax to the standard for using a peer entry |
| 05:20.50 | [TK]D-Fender | SIP/peer/numbertodial |
| 05:21.16 | CRCinAU_ | has that changed these days? I'm pretty sure that was the standard back when I first wrote this. |
| 05:21.25 | CRCinAU_ | and its been working since... well, a long time ago... |
| 05:22.04 | [TK]D-Fender | this is the clean and proper way since forever and noone should ever linger on using an @ unless it's a completly manual URI |
| 05:22.14 | CRCinAU_ | fair enough :p |
| 05:22.23 | CRCinAU_ | still, that isn't the cause of the problem... |
| 05:23.19 | [TK]D-Fender | Go check you haven't done other silly things like duplicate peer entries, etc |
| 05:23.44 | [TK]D-Fender | and prove you're working in the right files, in the right folder, etc |
| 05:25.07 | CRCinAU_ | well, everything is in /etc/asterisk - sip.conf and extensions.conf - which is the same I've used since 1.4.x |
| 05:26.02 | [TK]D-Fender | Ok, well you've made some mistake somewhere and it seems we aren't going to be able to help you find it. |
| 05:26.30 | CRCinAU_ | soooo - no config changes since moving from 10.x -> 11.x, now doesn't work, but its a config problem? o_O |
| 05:26.49 | CRCinAU_ | I haven't had to touch the configs for the sip providers in nearly a year |
| 05:26.50 | [TK]D-Fender | Peer basics are peer basics |
| 05:27.41 | CRCinAU_ | lets try again with: Why can't asterisk do a DNS lookup? |
| 05:27.57 | CRCinAU_ | yet when I do a 'core restart now' it can |
| 05:28.07 | CRCinAU_ | is the Addr->IP gets populated |
| 05:28.13 | [TK]D-Fender | I have no proof that's the issue. |
| 05:28.22 | CRCinAU_ | <PROTECTED> |
| 05:28.24 | CRCinAU_ | <PROTECTED> |
| 05:28.33 | CRCinAU_ | <PROTECTED> |
| 05:28.35 | [TK]D-Fender | if it is you should see it in core debug |
| 05:29.09 | CRCinAU_ | see, I have 3 sip providers, but it does this for all three. |
| 05:29.18 | CRCinAU_ | its not just a single provider that is having this issue. |
| 05:29.44 | [TK]D-Fender | look at core debug. If it's failing for that, it'll tell you |
| 05:30.02 | CRCinAU_ | configs are exactly the same for all three... and having only 5 lines in the config doesn't leave much room for error ;) |
| 05:30.19 | CRCinAU_ | the problem is though, it only happens at random |
| 05:30.23 | CRCinAU_ | it'll work fine for days. |
| 05:30.24 | CRCinAU_ | then stop |
| 05:30.34 | [TK]D-Fender | There's one there's there's always more room for than JELLO ... and that's errors |
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| 05:35.58 | CRCinAU_ | well, I guess try a single thing of disabling SRV lookups and see if it changes anything |
| 05:36.09 | CRCinAU_ | try one thing at a time until it no longer fails |
| 05:37.08 | [TK]D-Fender | Always a good start |
| 05:38.43 | CRCinAU_ | I figure it has to be DNS related... however it doesn't make sense to me that if Addr->IP is null, why it doesn't attempt to resolve the hostname |
| 05:38.58 | CRCinAU_ | one would think if you don't know what the hostname resolves to, it should be looked up |
| 05:39.18 | CRCinAU_ | and with DNS working at the system leve, there is no reason to think it should be failing. |
| 05:42.35 | [TK]D-Fender | unless DNS is flakey |
| 05:42.45 | [TK]D-Fender | or SRV is not working right |
| 05:42.53 | [TK]D-Fender | or that isn't the problem at all. |
| 05:43.03 | CRCinAU_ | maybe - but I don't have any further ideas? |
| 05:43.35 | [TK]D-Fender | Look at it. try to trigger it recovering. you should the difference if DNS is the case |
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| 07:28.48 | hrolf | Hi #asterisk, I'm having issues with the Record() application? Any ideas? |
| 07:29.00 | hrolf | Following are the console logs that appear: |
| 07:29.01 | hrolf | <PROTECTED> |
| 07:29.04 | hrolf | [Jan 22 12:23:41] WARNING[5818]: file.c:753 ast_readaudio_callback: Failed to write frame |
| 07:29.07 | hrolf | <PROTECTED> |
| 07:29.10 | hrolf | [Jan 22 12:23:41] WARNING[5818]: app_record.c:276 record_exec: ast_streamfile failed on SIP/212-0000001b |
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| 07:37.58 | hrolf | ? |
| 07:38.29 | hrolf | Any ideas what could be wrong here? |
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| 08:27.51 | Zogot | Ahoyhoy |
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| 08:37.09 | ChannelZ | Oy |
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| 08:59.43 | bulkorok | somebody using snom 300 ?! |
| 09:00.00 | bulkorok | I have one that re-registers every second or 4 seconds... a bit strange |
| 09:01.56 | mirela666 | I have |
| 09:02.03 | mirela666 | but not htat behaviour |
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| 09:11.11 | bulkorok | yeah.,.. deeper inspection shows that it's not the snom itself... must be a provisioning thing... |
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| 09:23.42 | skrusty | morning |
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| 09:29.46 | hrolf | Hi #asterisk, I'm having issues with the Record() application? Any ideas? |
| 09:29.50 | hrolf | Following are the console logs that appear: |
| 09:30.10 | hrolf | <PROTECTED> |
| 09:30.20 | hrolf | [Jan 22 12:23:41] WARNING[5818]: file.c:753 ast_readaudio_callback: Failed to write frame |
| 09:30.25 | hrolf | <PROTECTED> |
| 09:30.30 | hrolf | [Jan 22 12:23:41] WARNING[5818]: app_record.c:276 record_exec: ast_streamfile failed on SIP/212-0000001b |
| 09:34.26 | hrolf | Any help? |
| 09:36.31 | wdoekes | hrolf: enable debug logs and check if you can find more info surrounding these errors |
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| 09:40.18 | hrolf | wdoekes: Okay let me try that. |
| 09:41.56 | hrolf | wdoekes: Noping same error. No more info. Just "failed to write frame". |
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| 09:45.12 | wdoekes | no debug logs? or nothing you deem important? |
| 09:47.55 | hrolf | wdoekes: Nothing important regarding the error except the two lines already posted. No error/warnings etc. |
| 09:48.56 | hrolf | in debug logs. |
| 09:49.55 | wdoekes | I wasn't looking for more errors, I was looking for debug info |
| 09:50.49 | hrolf | wdoekes: Okay let me post it for you then. |
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| 10:02.17 | hrolf | wdoekes: http://pastie.org/8656579 |
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| 10:14.34 | wdoekes | hrolf: looks like there is something going on with codecs that switch. try allowing only ulaw or gsm and see what happens |
| 10:19.12 | hrolf | wdoekes: Okay what about the file format. I'm saving it as .wav? Should I change that to .gsm or only change codec in configuration? |
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| 12:50.50 | tompaw | I am confused as fock with the nat/rtp config. How should I have these configured if my asterisk has two NICs: one for LAN and one for public ip? |
| 12:51.35 | tompaw | I want it to sdp the external IP to the public ip leg and internal IP to the lan one. |
| 12:53.39 | tompaw | According to http://www.freepbx.org/forum/installation/dual-nic-setup it's not possible :/ |
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| 13:05.54 | tompaw | OK, solution found thjx :-) |
| 13:05.54 | *** part/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 13:15.12 | mirela666 | Hi one question, why pattern fe. _5XX matches _511 for example |
| 13:15.27 | mirela666 | CDR(dst) is _511 |
| 13:21.03 | *** join/#asterisk kaotiko (~0xFF@unaffiliated/kaotiko) |
| 13:22.23 | kaotiko | hi, a question, why when I have a Record(path/file:gsm), I can Reproduce it with Playback for example. But when I go to the path I cant see the file |
| 13:22.59 | kaotiko | My path is /tmp/ because if I use other path I get an error |
| 13:23.17 | wdoekes | kaotiko: did you see the 'k' Record() option? |
| 13:24.57 | kaotiko | But, when I do a playback I can listen it, but I cant see where is the file |
| 13:27.04 | kaotiko | exten => 5555,n,Record(/tmp/bienvenida:gsm) ; I do ls -la /tmp/ and the bienvenida.gsm doesnt appear |
| 13:27.46 | kaotiko | but when use Playback(/tmp/bienvenida) work well |
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| 13:33.53 | kaotiko | I change the path and now work better (/var/spool/asterisk/tmp/) |
| 13:36.09 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 13:36.52 | Tuju | has anyone here fought with new cisco 9-3-1SR3 firmware? |
| 13:37.33 | Tuju | i can get it to register, but it's not actually registered. and once it does it, it hangs around for a while and then restarts itself. |
| 13:38.27 | Tuju | mostly i've hard time understanding the USECALLMANAGER change in their configuration syntax. it doesn't make any sense. |
| 13:38.47 | Tuju | how you're supposed to distinguish lines and different proxies now? |
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| 14:34.10 | *** join/#asterisk markmcn (~markmcn@motoko.section9.ie) |
| 14:35.14 | markmcn | Hey all, I've been told this is a good place to find some help as I'm currently being driven crazy with an asterisk issue I'm hoping someone would be kind enough to help |
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| 14:35.55 | mirela666 | markmcn: what seems to be the officer problem? |
| 14:36.21 | markmcn | mirela666 I'll be a few min typing plase bear with me |
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| 14:39.51 | markmcn | I'm trying to setup a replacement asterisk system for a call centre, On the old system agents were static members of queue's and were logged in using AgentCallbackLogin() however this has been removed from the new version. All the alternatives i've seen push the sip endpoint into the queue rather then the agent which is an issue for queuementrics as our agents can move from ext to ext can you please suggest any alternatives to using AgentCallbackLogi |
| 14:41.37 | [TK]D-Fender | Markthis was changed a long while back. AddQueueMemeber, RemoverQueueMemeber is what you're looking for now |
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| 14:45.12 | markmcn | [TK]D-Fender: I've found those functions however all examples just use them to push sipendpoints into a queue. So I tried adding agests as static members but then how does asterisk find the agesnt if the agent doesn't log in. Sorry again if i'm not making sense or if my understanding isn't right |
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| 14:47.10 | tparcina | My colleague (web programmer) needs to program web page where it will be visible is user on the phone or not. What is the right way to check that info from asterisk? |
| 14:47.22 | Tuju | crap, i tried to mirror phone traffic into wireshark but now spotted my switch on windowboard. |
| 14:47.27 | mirela666 | markmcn: you can have static members and Pause them on call connect and unpause on disconect (PauseQueueMember,UnpauseQueueMember) |
| 14:47.31 | Tuju | no wonder it doesn't work. |
| 14:48.08 | tparcina | Should we use manager or something else? |
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| 14:49.12 | freckle | tparcina: I have a python script that runs continually. It opens up the AMI and listens for extensions connection, hanging up or ringing. It then updates a field in a table of extensions to show the state. The webpage refreshes every 2 seconds with the latest status. |
| 14:49.36 | freckle | It is pretty basic but works very weel |
| 14:49.40 | freckle | /well/ |
| 14:50.06 | tparcina | freckle: That sound great. |
| 14:50.20 | tparcina | freckle: Are you willing to share that script? |
| 14:51.09 | freckle | tparcina: https://plus.google.com/111016409913401452215/posts/7XsFrxw1VgW |
| 14:51.26 | mirela666 | tparcina: asterisk -rx "sip show inuse" |
| 14:51.46 | [TK]D-Fender | markmcn: Agents are GONE. |
| 14:51.46 | mirela666 | or to use manager events |
| 14:52.15 | [TK]D-Fender | markmcn: So you HAVE to push another channel-type. If you want to use the dialplan to process them then use a Local channel |
| 14:52.19 | markmcn | mirela666: at the min I have an agent defined as agent => 1001,,Mark then I add A1001,0 to a queue and call the cue however nothing rings. From what I can at the min I have no way of telling asterisk which sipendpoint agent1001 is sitting at |
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| 14:52.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.57 | tparcina | freckle: Thank you. What minimal rights I need to give to manager? |
| 14:53.43 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 14:53.46 | [TK]D-Fender | freckle: Actually.. that "polls" the status on interval, it doesn't "listen" :) |
| 14:54.07 | [TK]D-Fender | freckle: It's the same methodology I use to get llive queue stats on my CSR's phones |
| 14:54.12 | tparcina | mirela666: Thank you. What does this inuse 0/0/0 mean? |
| 14:54.20 | markmcn | [TK]D-Fender: REALLY ? if so what are callcentres now doing to replace agesnts |
| 14:54.26 | [TK]D-Fender | (login/pause status, queue waiting, held, voicemail count, etc) |
| 14:54.50 | [TK]D-Fender | markmcn: I just told you... you "log in" USING AddQueuememeber to put them in in the first place |
| 14:54.54 | freckle | [TK]D-Fender: yeah sorry, I forgot how it worked before I looked up the post I wrote |
| 14:55.06 | [TK]D-Fender | markmcn: AQM/RQM = AgentCallbacklogin (and out) |
| 14:55.27 | [TK]D-Fender | freckle: A listening daemon would be who things like FOP work |
| 14:55.30 | [TK]D-Fender | how* |
| 14:55.45 | freckle | [TK]D-Fender: yeah I know |
| 14:55.48 | mirela666 | tparcina: for example: 998 1/1/0 was ringing |
| 14:56.08 | mirela666 | after that 998 1/0/0 when answered |
| 14:56.17 | mirela666 | 0/0/0/ not in use |
| 14:56.46 | mirela666 | i'm not exacly sure twhat are the fields, but guessing |
| 14:57.06 | markmcn | [TK]D-Fender: Thanks I understand using AQM/RQM to add endpoints to queue. So working on what you've just told me that agents are gone. you add and remove sipendpoitns ? |
| 14:57.19 | mirela666 | tparcina: or you can try out http://monast.sourceforge.net/ |
| 14:58.05 | [TK]D-Fender | markmcn: Channels of a type that still exist. SIP is one option. I already spelled out the other |
| 15:00.22 | markmcn | [TK]D-Fender: Thanks for the help I kinda get what you mean i'll have to go back to the planning |
| 15:01.20 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:01.54 | [TK]D-Fender | markmcn: sounds good |
| 15:09.15 | *** part/#asterisk freckle (~jon@entagw100.enta.net) |
| 15:09.37 | tparcina | mirela666: Thank you for the link. I'll look into all three solutions and figure out what is best for my case. |
| 15:11.22 | mirela666 | np ;) |
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| 15:27.03 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 15:27.35 | Katty | morning |
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| 15:32.17 | dwayne | offers Katty a muffin |
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| 15:38.39 | markmcn | [TK]D-Fender: Just wanted to say thanks a bit of redesign has potentially solved the problem tip of my hat to you sir |
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| 15:39.05 | [TK]D-Fender | markmcn: You're welcome |
| 15:44.20 | pii3 | i search on asterisk website but i cannot find a compatible/supported devices like mitel, alcatel, cisco and ... |
| 15:45.03 | [TK]D-Fender | because it doesn't work like that |
| 15:45.19 | [TK]D-Fender | first there is no such thing as compatability with a brand... but rather specific MODELS |
| 15:45.48 | pii3 | so let me know please can you give me more light about this ? |
| 15:45.52 | [TK]D-Fender | pii3: Asterisk supports a number of PROTOCOLS that those makers also make gear that use. |
| 15:46.31 | [TK]D-Fender | pii3: SIP, IAX2, H.323 (iffy), MGCP ((iffy, and phones only, not trunks) |
| 15:46.42 | [TK]D-Fender | pii3: As for hardware PC interfaces that's another matter |
| 15:46.50 | [TK]D-Fender | pii3: For which you'd have to be more specific |
| 15:47.22 | [TK]D-Fender | pii3: Digium's card are the front runnings (being the authors of Asterisk), and many other manufacturers make cards supported by it |
| 15:47.36 | [TK]D-Fender | pii3: So what are you looking to actually do? |
| 15:47.38 | pii3 | so ip phone should be compatible with a protocol |
| 15:48.04 | [TK]D-Fender | IP phones speak a protocol. Which one matters as to how usable it might be, licensing costs, etc |
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| 15:50.19 | lnb | if call is missed and goes to vmail, is it possible to pull the call back and answer it? |
| 15:51.30 | [TK]D-Fender | Same way you would for any call |
| 15:51.58 | [TK]D-Fender | Hijacking another call is hijacking another call. |
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| 15:59.44 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:59.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:59.47 | [TK]D-Fender | YES! |
| 16:00.29 | Katty | dwayne: :> |
| 16:00.37 | Katty | noms muffin, offers dwayne apple slices |
| 16:00.53 | dwayne | eats apple slices with cinnamon |
| 16:00.56 | dwayne | thanks! |
| 16:01.15 | Katty | mm, cinnamoned apples |
| 16:01.41 | Tuju | yeah, it looks new xml config does not like multiple proxies in callmanager branches. |
| 16:02.38 | Katty | dwayne: i had a banana drizzled with chocolate sauce last night :> |
| 16:02.43 | Katty | dwayne: it was quite tasty |
| 16:02.52 | dwayne | ooooh! |
| 16:03.13 | Katty | not real chocolate sauce tho :< just hershey's. next time i'm melting semisweet choc chips |
| 16:05.10 | Tuju | any idea why cisco phone sends ip address in sip packages, instead of dns names? |
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| 16:31.58 | jaflong | Hi, Is 'video on hold' possible with confbridge? |
| 16:32.24 | tzafrir | When I set tonezone in dahdi to be mx, the dialtone played after a stutter tone is different from the original dialtone |
| 16:32.46 | tzafrir | Is that intended, or a copy&paste mistake from the US definitions? |
| 16:33.21 | *** part/#asterisk TobSnyder (~schneider@146-52-43-241-dynip.superkabel.de) |
| 16:35.20 | [TK]D-Fender | jaflong: Have never tried, but I suspect it is possible if you have matching files in all the right codecs... |
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| 16:38.00 | jaflong | [TK]D-Fender: I am not aware that this feature is builtin like music on hold |
| 16:39.44 | jaflong | would like require manual streaming, and is this a good idea |
| 16:41.24 | [TK]D-Fender | Well WHAT you use would be important. |
| 16:41.47 | [TK]D-Fender | In "files" mode I could imagine it pulling both as it does for Voicemail, etc |
| 16:41.58 | [TK]D-Fender | Streaming... I wouldn't bet on... |
| 16:42.18 | WIMPy | Oh, VM will do video? |
| 16:42.44 | [TK]D-Fender | yup |
| 16:43.51 | WIMPy | BTW: Does any one know exactely if VM caches any information about a mailbox? I.e. is there any danger in symlinking the INBOX etc directories of multiple mailboxes? |
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| 16:52.20 | jaflong | so, is this correct, I should look into playback of 'video ' voicemail through the confbridge? |
| 16:54.49 | [TK]D-Fender | you should be using FILES mode for MoH, and have BOTH the video & audio streams |
| 16:55.09 | [TK]D-Fender | this has nothing to do with VoiceMail. They were just a sample of one place that records all the streams |
| 16:58.35 | jaflong | thanks for making it clearer |
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| 17:00.00 | gmalsack | morin y'all!!! yik yik... |
| 17:02.24 | gmalsack | planning on upgrading from 1.8.24 to 11.7. I've done it on a development system, but can't really test it 100%. However if I maintain the original system source, can I simply run 'make install' in the 1.8.24 source and restart asterisk to go back if something in production blows up? I've done it on the dev server and it reports that the version is rolled back, however I would like a second opinion on if that's proper practice... |
| 17:02.59 | Qwell | gmalsack: make uninstall in the old (new, 11.x) source first |
| 17:03.27 | gmalsack | Qwell: oh ok. thanks!!! +1 |
| 17:05.07 | gmalsack | ruh row shaggy.... -> http://pastebin.com/tW1sh2tv |
| 17:06.25 | WIMPy | I have never tried make uninstall. But if you don't have exactely the same modules, you will get a warning about modules from another version on make install. |
| 17:06.42 | WIMPy | You can just go and wipe the contents of the modules directory before make install. |
| 17:07.09 | gmalsack | ok. |
| 17:07.17 | gmalsack | thanks WIMPy! |
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| 17:44.33 | julgr | Hey, folks. I'm trying to setup a loop of intermittent silent RTP injected into a muted audio channel. I'm just trying to figure out how to implement the loop. |
| 17:45.23 | julgr | this is what I am thinking will do it. |
| 17:45.29 | julgr | same => n(playsilence),Playback(silence/10) |
| 17:45.30 | julgr | same => n,Wait(600) |
| 17:45.30 | julgr | same => n,Goto(playsilence) |
| 17:46.02 | julgr | is there some simpler way of looping playback on a call? |
| 17:46.25 | [TK]D-Fender | that deos it |
| 17:46.33 | [TK]D-Fender | oes* |
| 17:46.48 | julgr | phew! |
| 17:46.55 | julgr | Thanks! |
| 17:47.10 | julgr | just a sanity check |
| 17:52.12 | pabelanger | well, you would get any audio for 600 seconds |
| 17:52.26 | pabelanger | so, why are you even playing the silence/10? |
| 17:53.06 | [TK]D-Fender | Technically... if you're looping like that, audio is already established, so the "wait" IS audio everybit as much as the PLayback is |
| 17:53.48 | pabelanger | well, asterisk will not generate audio when you are in Wait() |
| 17:54.11 | pabelanger | I doubt there will be any RTP activity |
| 17:54.16 | pabelanger | unless I am missing something |
| 18:03.45 | paulc | Isn't there a config setting somewhere where it DOES generate (silent) RTP somewhere? Like when recording or waiting for DTMF or something.. it caught me out once before with an SBC that said "no RTP = dead call.. <click!>" |
| 18:04.21 | file | yes, asterisk.conf |
| 18:04.43 | file | but I don't know if that works for Wait |
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| 18:27.50 | Katty | CABINENT DOORS |
| 18:31.10 | gmalsack | I've got a dev system up and running right now. just added an extension with wait. I'll let you know on the rtp in a sec. what version are you running? |
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| 18:35.16 | gmalsack | as |
| 18:41.12 | gmalsack | ast 11.6 during wait, ast is not sending rtp to the phone, however the phone is sending rtp to ast... even when muted. this was tested with grandstream and polycom phones. however if during the wait, the phone places the call on hold there is no rtp stream. |
| 18:44.27 | [TK]D-Fender | There is no such thing as "mute" to Asterisk. |
| 18:44.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.230) |
| 18:44.44 | [TK]D-Fender | The phone can choose not to send audio, but it knows nothing about a state |
| 18:44.51 | *** join/#asterisk biox (~jesseolso@63.228.60.235) |
| 18:45.38 | gmalsack | Clarification, pressing the mute button on either of those 2 manufacturers phones does nothing to change the flow of RTP from the phone to ast |
| 18:46.12 | [TK]D-Fender | it may send RTP... doesn't mean there is SOUND in it |
| 18:46.37 | pabelanger | gmalsack, it is phone specific. I know of phones that don't send audio when your mute the local microphone |
| 18:47.45 | gmalsack | oh right. totally true. I'm sure the mute button does nothing other than electronically disconnect the mic on the handset/speakerphone. Though like pabelanger said, this could very easily be manufacturer specific... |
| 18:48.15 | gmalsack | which is why I listed which manufactures I tested. Oh, additionally, I tested a digium phone as well. |
| 18:48.49 | WIMPy | Yes, big issue with chan_sip, that you know nothing about hold state. |
| 18:49.17 | [TK]D-Fender | Um, last I checked * knows if a channel is on "hold" |
| 18:49.32 | gmalsack | that's what I was going to say. |
| 18:49.43 | [TK]D-Fender | Which is why we see it trigger MoH... |
| 18:49.44 | WIMPy | But the peer doesnt know. |
| 18:49.49 | [TK]D-Fender | What peer? |
| 18:50.11 | WIMPy | Watever we put on hold. |
| 18:50.15 | [TK]D-Fender | I call into the dialplan and hit voicemail. I hit "Hold". My channel is listed as "hold" |
| 18:50.31 | [TK]D-Fender | If I'm in a bridged call, * gives them Moh... |
| 18:50.39 | [TK]D-Fender | that sounds like "indication" to me. |
| 18:50.52 | [TK]D-Fender | It's just that * has a ROLE to do. |
| 18:51.05 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 18:51.11 | WIMPy | Something that doesn't exist. |
| 18:51.20 | [TK]D-Fender | ? |
| 18:51.23 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.129) |
| 18:51.28 | gmalsack | ???? |
| 18:51.42 | [TK]D-Fender | You're missing some coherence there... try again... |
| 18:51.44 | WIMPy | The indication _from_ Asterisk. |
| 18:51.53 | [TK]D-Fender | Well you see it as MoH. |
| 18:52.13 | WIMPy | Define "you". |
| 18:52.15 | [TK]D-Fender | It isn't just reporting a state to the phone and leaving it up to the phone to know to reinvited a MoH stream, etc |
| 18:52.19 | gmalsack | additionally, the phones hold button that placed the call on hold lights up. |
| 18:52.27 | WIMPy | TNo peer will know when it was placed on hold. |
| 18:52.31 | [TK]D-Fender | When you do a channel dump you can see that the channel is on hold. |
| 18:52.43 | [TK]D-Fender | It isn't a SIP MESSAGE if that's what you mean |
| 18:52.51 | WIMPy | Yes, Asterisk knows, buy noone else. |
| 18:52.54 | [TK]D-Fender | because * isn't passing on the responsibility. |
| 18:53.14 | [TK]D-Fender | * is playing a good B2B UA as it should |
| 18:53.19 | [TK]D-Fender | it isn't a proxy for this... |
| 18:53.20 | WIMPy | Yes, and that's bad. |
| 18:53.24 | gmalsack | oh are you talking about indication via BLF for a hold status? |
| 18:53.44 | WIMPy | No. The other end of the call needs to know it's on hold. |
| 18:53.59 | [TK]D-Fender | WIMPy: I don't recall how any other system aadvertises it honestly.... |
| 18:54.08 | WIMPy | Have you ever been in a conference call and received another call? |
| 18:54.15 | [TK]D-Fender | yes |
| 18:54.27 | [TK]D-Fender | the Holdee in my cases GETS MoH. that's the "clue" |
| 18:54.53 | [TK]D-Fender | And the fact I say "hold on a minute" and then they magically stop hearing me talk to them. |
| 18:55.01 | WIMPy | Great. If you're using Asterisk you might have less freinds after that unless you're carefull about the situation. (and aware of the issue). |
| 18:55.35 | WIMPy | Yes and you play music to a conference, making it unusable for the other participants. |
| 18:55.40 | [TK]D-Fender | WIMPy: I've never in my life met the hypersensitive people you're referring to. |
| 18:55.48 | [TK]D-Fender | WIMPy: Do you live next to a trauma center? |
| 18:56.08 | WIMPy | You call that hypersensitive? |
| 18:56.09 | [TK]D-Fender | WIMPy: Perhaps it's just a culteral thing about accepting being put on hold. |
| 18:56.37 | WIMPy | Did you read what I wrote? |
| 18:56.49 | [TK]D-Fender | Putting people on hold without telling them would be rude here in general, so they know it's coming, and when they hear MoH they know it's NOW |
| 18:56.59 | WIMPy | I was talking about being in a conference. |
| 18:57.06 | [TK]D-Fender | [13:55]WIMPyYes and you play music to a conference, making it unusable for the other participants. <- well THIS.. that's another matter. |
| 18:57.19 | [TK]D-Fender | So what do you propse for haveing a hold with music for SOME and not others? |
| 18:57.30 | WIMPy | I don't. |
| 18:57.37 | [TK]D-Fender | You could PARK the confernce to a lot WITHOUT MoH with audio |
| 18:57.52 | [TK]D-Fender | There's one solution |
| 18:57.54 | WIMPy | But if the peer knw it was on hold, it wouldn't forward the MOH to the conference. |
| 18:58.06 | [TK]D-Fender | You could NOT use MoH and everyone just gets silence. |
| 18:58.08 | [TK]D-Fender | There another. |
| 18:58.15 | [TK]D-Fender | You could just MUTE the call |
| 18:58.30 | WIMPy | It's just a case of missing basic functionality. |
| 18:58.32 | [TK]D-Fender | Someone's splitting hairs at this point... |
| 18:59.14 | gmalsack | how do you consider that missing basic functionality? Do you know of another system that sends that information??? |
| 18:59.30 | WIMPy | I'm not the only one who has experienced that issue. |
| 18:59.42 | [TK]D-Fender | WIMPy: How would you enact this difference on another susyem? Does SIP specifically implement it? |
| 19:00.11 | WIMPy | gmalsack: Sure. The PSTN or any usual PBX. |
| 19:01.07 | gmalsack | wrong.... None of the hybrid systems or digital systems that I installed would send a signal through the pstn to the other side telling them that I had placed them on hold. |
| 19:01.25 | [TK]D-Fender | PRI sure knows nothing of "hold" |
| 19:01.30 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 19:01.35 | [TK]D-Fender | ANALOG infinitely less |
| 19:01.38 | gmalsack | Let alone tell their conference app to ignore the audio I'm sending them |
| 19:01.39 | WIMPy | SIP in its original state is pretty useless. And I certainly do not have any motivation of finding out what additions are available for all the missing functionality that is a bit of a hit or miss on |
| 19:01.44 | WIMPy | the interoperability side anyway. |
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| 19:02.28 | gmalsack | ooooooo kay |
| 19:02.32 | WIMPy | Sorry, but you should go and read a bit. Off course will PRIs forward that information. |
| 19:02.46 | gmalsack | next |
| 19:02.50 | [TK]D-Fender | WIMPy: What system has this hybrid treatment you're referring to? Is it ONLY "internal conference protection"? |
| 19:03.20 | [TK]D-Fender | WIMPy: Every PBX & key system I've used has had internal MoH. |
| 19:03.35 | [TK]D-Fender | Nothing was ever signalled back |
| 19:03.52 | *** join/#asterisk powerunits (b6b982ef@gateway/web/freenode/ip.182.185.130.239) |
| 19:04.13 | WIMPy | I don;t know what you guys are using, but that's the kind of functionality I have been taking for granted for many years. |
| 19:04.35 | [TK]D-Fender | So you never used MoH? |
| 19:04.49 | [TK]D-Fender | Just expec som kind of "signal" back? What does that sound like? |
| 19:05.06 | *** join/#asterisk nicknam1232 (5c15e126@gateway/web/freenode/ip.92.21.225.38) |
| 19:05.07 | powerunits | hello team i need some suggestion.. i want to install asterisk on a hosted server could you recommend how this server AMD Opteron 8Core 8x 2.3 GHz.. |
| 19:05.18 | powerunits | will preform if we use asterisk on it |
| 19:05.21 | gmalsack | and what does the receiving system do with the signal it gets back? |
| 19:05.22 | [TK]D-Fender | WIMPy: So far your experience runs counter to .... lets just say pretty much all of North America |
| 19:05.37 | WIMPy | Again: There's nothing wrong with sending MOH. It's the missing signalling. |
| 19:06.30 | WIMPy | [TK]D-Fender: Well, I guess that's why everyone know that you souldn't get hardware or consultancy from there. |
| 19:07.37 | gmalsack | So what you're saying is that with systems you've installed. When you program a system to host a conferencing bridge, that bridge looks for a hold signal to come through and basically mutes the channel that sent the hold signal so the conference doesn't hear the moh? |
| 19:08.06 | powerunits | any suggestion guys |
| 19:08.07 | powerunits | ? |
| 19:08.08 | WIMPy | gmalsack: Yes. |
| 19:08.37 | WIMPy | gmalsack: Or the bad solution: Hold doesn't work in that situation. |
| 19:09.18 | WIMPy | Although I think the "bad" one is the more usual one. |
| 19:09.37 | WIMPy | But that's at different points obviousely. |
| 19:10.00 | gmalsack | powerunits: I guess it would depend on what you're going to expect asterisk to do... |
| 19:10.10 | [TK]D-Fender | [14:06]WIMPy[TK]D-Fender: Well, I guess that's why everyone know that you souldn't get hardware or consultancy from there. <- Or perhaps it's a more limited view of a different percentage of the population. |
| 19:10.50 | powerunits | gmalsack: simple calls to outbound trunk using g729 codec or ulaw |
| 19:11.36 | [TK]D-Fender | [14:07]gmalsackSo what you're saying is that with systems you've installed. When you program a system to host a conferencing bridge, that bridge looks for a hold signal to come through and basically mutes the channel that sent the hold signal so the conference doesn't hear the moh? <- this would imply an expectation that no system would ever inject MoH. |
| 19:11.39 | gmalsack | [14:06] WIMPy [TK]D-Fender: Well, I guess that's why everyone know that you souldn't get hardware or consultancy from there. <- Or perhaps it's a more limited view of a different percentage of the population. <- which would make me wonder why I consult on phone systems in central america, austraila, and germany.... |
| 19:12.10 | [TK]D-Fender | gmalsack: I've consulted all over the globe as well... |
| 19:12.57 | gmalsack | [TK]D-Fender: and did you every run into what WIMPy is talking about???? Obviously which would be why we are both totally puzzled by this conversation... lol |
| 19:13.06 | [TK]D-Fender | nope |
| 19:13.25 | WIMPy | <[TK]D-Fender> basically mutes the channel that sent the hold signal so the conference doesn't hear the moh? <- this would imply an expectation that no system would ever inject MoH. <- You're sill completely missing the point. |
| 19:13.58 | [TK]D-Fender | His experience is quite unique in terms of technological and cultural outlook. I don't know where the applicability boundaries are, but I don't doubt his experience, it's just that I'm not sure what the applicable zone is all the time... |
| 19:14.42 | WIMPy | Oh. and BTW: Did you know that when using DAHDI, you might be listening to your Asterisks MOH instead of wheat the other end sends as MOH? |
| 19:15.09 | [TK]D-Fender | WIMPy: Actually I think I just caught a part of it. The signal doesn't have to imply that there isn't MoH, it just offers the other side an chance to "opt-out" until it is restored. |
| 19:15.30 | WIMPy | [TK]D-Fender: Correct. |
| 19:15.39 | WIMPy | And to display the information to the user, off course. |
| 19:15.57 | [TK]D-Fender | WIMPy: I can imagine this at least better now... |
| 19:16.15 | [TK]D-Fender | WIMPy: it wasn't really a stretch.... just the first level threw me from seeing the possibiilty as fast as I could have |
| 19:16.28 | *** join/#asterisk u0m3 (~u0m3@92.80.105.67) |
| 19:16.38 | gmalsack | powerunits: If you're planning on doing a bunch of g729 trancoding it could get harry.... I have 2 quad core xeon 3.4 ghz cpus. I was doing software transcoding and was peaking out the cpu when I would get to about 120 simultaneous calls. |
| 19:16.47 | WIMPy | At least we've got that sorted. |
| 19:17.36 | [TK]D-Fender | WIMPy: From what I've seend of every system... nothing in north America seems to do anything about this. |
| 19:17.42 | WIMPy | And in the good old days you would also be notified if the other party would put you in to a conference. |
| 19:17.46 | [TK]D-Fender | WIMPy: I put you on hold, you get MY music. |
| 19:18.11 | [TK]D-Fender | WIMPy: I'd be seriously doubting that DAHDI transmits this for PRI. |
| 19:18.21 | [TK]D-Fender | or receives and processes it at any rate |
| 19:18.23 | WIMPy | [TK]D-Fender: Unless I use Asterisk with the "right" configuration. |
| 19:18.40 | [TK]D-Fender | WIMPy: I could see this a very bad thing for conferencing services. |
| 19:18.44 | WIMPy | [TK]D-Fender: I not sure it transmits it, but it definitely receives and uses it. |
| 19:18.48 | gmalsack | powerunits: however if you can get the hosting provider to install a transcoder card(s) for you, the system should easily be able to double that load if not more. Digiums transcoder cards will handle 120 g729 trancodings, offloading that overhead from the CPU. And you can install multiple transcoder cards in 1 system. |
| 19:19.06 | [TK]D-Fender | WIMPy: And if that's a primary aspect of your business and the expectation is that you'll NEED this.. I could see why * looks "poor" to you |
| 19:19.43 | [TK]D-Fender | WIMPy: I've never perceived any action on *'s behalf for this. If you had a sample I'd be interested in seeing it.. |
| 19:19.54 | WIMPy | [TK]D-Fender: No, it's just one of many details that are just plain annoying. |
| 19:20.20 | WIMPy | [TK]D-Fender: I think it's even configurable. |
| 19:20.53 | [TK]D-Fender | WIMPy: You've got me curious on it, but I don't recall having scanned over an option in any of the sample configs that sounded like that |
| 19:21.13 | gmalsack | [TK]D-Fender: but even so. it's a feature that is 100% dependant on the far side system. I just can't see it.... Merlyn legends, didn't send it. Lucent systems didn't, panasonic systems didnt, I don't believe mitel system did either.... |
| 19:21.20 | WIMPy | [TK]D-Fender: chan_dahdi.conf: discardremoteholdretrieval |
| 19:22.15 | [TK]D-Fender | WIMPy: Do you have an idea how far back that got in? |
| 19:22.30 | WIMPy | No |
| 19:23.01 | WIMPy | I'm not fiddling with chan_dahdi that much. |
| 19:23.25 | powerunits | gmalsack: Digiums transcoder cards how much this card cost? |
| 19:24.12 | drmessano | $1100 |
| 19:24.16 | drmessano | Google |
| 19:24.31 | powerunits | thanks |
| 19:24.46 | drmessano | Thank google |
| 19:27.15 | pabelanger | CPU are cheaper :D |
| 19:27.34 | drmessano | I didnt see where he was expecting all that transcoding though |
| 19:28.02 | gmalsack | powerunits: I can get you those cards for 996.00 |
| 19:28.08 | WIMPy | [TK]D-Fender, gmalsack: The conclusion is that you were both incorrect. You both know a system that is able to ignore MOH from a remote party in a conference: Asterisk, as long as the call come in via chan_dahdi. |
| 19:28.16 | powerunits | rite |
| 19:28.22 | powerunits | i have one question.. |
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| 19:28.57 | powerunits | what do you think.. about codec at http://asterisk.hosting.lv/ |
| 19:29.03 | powerunits | and this card g729 codec |
| 19:29.19 | gmalsack | drmessano: <powerunits> gmalsack: simple calls to outbound trunk using g729 codec or ulaw |
| 19:29.38 | powerunits | do you think this card will also |
| 19:29.39 | drmessano | If your endpoints are G729 and your provider is g729 capable, there is no transcoding |
| 19:29.48 | drmessano | So where is the card needed or even the software codec? |
| 19:29.56 | WIMPy | And BTW: Asterisk will also make use of hold states received via SIP, it just doesn't send them. |
| 19:30.47 | drmessano | Sounds like someone is trying to sell a card |
| 19:30.59 | gmalsack | drmessano: right, IF the provider is doing it. You don't always know what the capabilities are of the system you are connecting to for RTP... |
| 19:31.03 | powerunits | the card you have is on warranty? |
| 19:31.23 | gmalsack | powerunits: new from digium. I'm a reseller... |
| 19:31.33 | drmessano | lol |
| 19:31.41 | drmessano | This is nuts |
| 19:31.59 | drmessano | Most providers support G729 |
| 19:32.00 | powerunits | please can i have ur email |
| 19:32.01 | gmalsack | no not trying to sell. |
| 19:32.15 | gmalsack | just trying to help. |
| 19:32.24 | powerunits | oooooh |
| 19:32.25 | powerunits | ok |
| 19:32.34 | powerunits | sorry i thought you are selling |
| 19:32.35 | powerunits | :) |
| 19:33.08 | drmessano | I can't think of a provider that doesnt support G729, and an endpoint that doesnt. So this whole transcoding card discussion makes little sense |
| 19:33.09 | file | raises eyebrow |
| 19:33.39 | pabelanger | powerunits, buy g729 licenses from digium |
| 19:33.41 | pabelanger | support the project |
| 19:34.44 | gmalsack | pabelanger: again g729 licenses are software transcoding which puts strain on the cpu, the transcoder card removes that. |
| 19:35.09 | gmalsack | pabelanger: plus the digium transcoder card is less than purchasing 120 licenses at 10.00 each. |
| 19:35.12 | pabelanger | well, when you buy the transcoder, you are buying the g729 license |
| 19:35.14 | drmessano | What if he's never transcoding? |
| 19:35.18 | drmessano | or rarely? |
| 19:35.30 | pabelanger | s/when/then |
| 19:35.32 | gmalsack | drmessano: what if he needs to record the audio... |
| 19:35.49 | drmessano | He never specified any of that |
| 19:36.25 | pabelanger | There are pros and cons for the card. I'm not opposed to it, however, theses days, a CPU will do just as well |
| 19:36.28 | drmessano | He asked about specs and was told he needed a transcoding card when he mentioned G729 |
| 19:36.32 | drmessano | Thats a bit off |
| 19:36.56 | gmalsack | drmessano: no but didn't want to assume anything. I run a call center doing about 1000 calls per hour. connecting with tons of affiliates throughout the us. we use a ton of transcoder channels... |
| 19:37.03 | drmessano | I run all G729 on a couple of boxes and a few licenses do just fine for the little bit that's not native g729 |
| 19:37.42 | [TK]D-Fender | [14:28]WIMPy[TK]D-Fender, gmalsack: The conclusion is that you were both incorrect. You both know a system that is able to ignore MOH from a remote party in a conference: Asterisk, as long as the call come in via chan_dahdi. <- I have no problem accepting that. Took just a little bit to trigger a response that got me on the path to understanding how the converse scenario could function,... |
| 19:37.44 | [TK]D-Fender | ...so thanks :) |
| 19:38.54 | WIMPy | [TK]D-Fender: Great. Now we just need the missing bits in Asterisk :-) |
| 19:39.03 | [TK]D-Fender | WIMPy: Deal! |
| 19:39.18 | gmalsack | I gave up and started helping powerunits which led into another lengthy discussion... lol apparently today is controversy day!!! |
| 19:40.01 | [TK]D-Fender | Nah, this was "typically transparent thing you could apparently go this long without encountering" |
| 19:40.02 | drmessano | No, but talking someone into a hardware or software purchase when they're in the early stages of planning and havent even made a case for the need is kinda silly |
| 19:40.25 | WIMPy | gmalsack: I just added that parts of that functionality are already available in Asterisk. There are just some bits missing to make the whole thing work. |
| 19:40.28 | drmessano | Especially for something like a 120 channel transcoder card that many obviously live without] |
| 19:40.52 | drmessano | Positioning it as a necessity and all |
| 19:41.06 | gmalsack | drmessano: I wasn't trying to talking him into anything, I was sharing my experience when no one else was... |
| 19:41.40 | [TK]D-Fender | I see the point... the key things that would need it would be conference services that need to stifler "filler". Can't think of anything els e specific offhand as a necessity, but I suppose an indication that you are in-fact on hold... could be considered "nice" Mind you it's always seemed evident in my calling experience |
| 19:42.00 | [TK]D-Fender | drmessano: For a conferencing system it would be and I can see the point for it. |
| 19:42.17 | [TK]D-Fender | drmessano: You want to protect your attendee's from idiots forcing MoH on you |
| 19:42.28 | [TK]D-Fender | drmessano: And it's part of the spec... |
| 19:42.55 | [TK]D-Fender | I find that all argument against things fails when it is part of the actual spec. |
| 19:43.03 | WIMPy | [TK]D-Fender: Well, the indication is received and used in both chan_dahdi and chan_sip. Chan_dahdi also offers you to replace MOH, so you just have to enable that and set a silent MOH class beofre |
| 19:43.08 | WIMPy | entering ConfBridge. |
| 19:43.23 | [TK]D-Fender | Why not? Its the spec. It SHOULD do it, and there are cases that full warrant it |
| 19:43.55 | gmalsack | [TK]D-Fender, WIMPy: So if * is the broken piece, does that mean that everyone else in the SIP industry does support a hold signal? i.e. my phones will send a SIP message when the call is placed on hold? |
| 19:44.06 | WIMPy | chan_sip doesn;t offer you to replace MOH and (th IMHO worse part) it doesn;t send the information. |
| 19:44.13 | [TK]D-Fender | WIMPy: that is a viable idea |
| 19:44.47 | WIMPy | Generalisations in the SIP world are impossible. Everyone implemets it their own way, wich is part of the problem. |
| 19:45.16 | [TK]D-Fender | RFC = No F*&$ing Comment ;) |
| 19:45.20 | gmalsack | +100 for that!!! |
| 19:45.32 | [TK]D-Fender | Also frighteningly close to KFC. Yup.. you be FRIED |
| 19:45.42 | WIMPy | But when I have a call between my standard ISDN phone and a sip phone, and the sip end places the call on hold, my ISDN phone will display the information. |
| 19:45.43 | gmalsack | lol |
| 19:46.23 | WIMPy | (both connected to Asterisk, off course) |
| 19:46.26 | gmalsack | WIMPy: really. I've never in all my 40 years saw a phone tell me I was placed on hold.... |
| 19:46.59 | WIMPy | gmalsack: Well, again somethign I have taken for granted for many years. |
| 19:47.14 | [TK]D-Fender | WIMPy: I guess the larger importance of that forwarding of info is when you cross "trunks". Internal MoH would be expected, but you wouldn't want to infringe on an outside network. Then the fact that SIP is SIP is SIP in * leads you to the issue |
| 19:47.19 | gmalsack | WIMPy: cool feature! |
| 19:47.39 | [TK]D-Fender | WIMPy: Probably moreso with the fact that BRI is a reality there. |
| 19:47.56 | WIMPy | gmalsack: The phone would also display the fact when the remote end puts me in to a conference. |
| 19:48.04 | [TK]D-Fender | WIMPy: A lot of irony as to how tech rolled out world-wide |
| 19:48.35 | WIMPy | [TK]D-Fender: True. These are things that work perfectely even without any PBX, just with a bog standard phone directely connected to the line. |
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| 21:45.57 | Katty | looks in |
| 21:46.57 | drmessano | hides |
| 21:47.46 | Katty | am i supposed to count to 10 now? |
| 21:51.51 | Chainsaw | Definitely. |
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| 22:00.12 | Katty | drmessano: 10 seconds are up! |
| 22:00.16 | Katty | goes hunting for drmessano |
| 22:11.36 | eppigy | hello |
| 22:11.49 | Katty | hi |
| 22:12.46 | eppigy | hi Katty |
| 22:12.49 | eppigy | i am dave |
| 22:13.14 | Katty | are you now |
| 22:13.15 | pabelanger | hello yes, this is dog |
| 22:16.50 | eppigy | hello dog |
| 22:16.51 | eppigy | i am dave |
| 22:20.58 | Kobaz | yo dog i heard you like dogs |
| 22:24.21 | drmessano | Sorry, got busy |
| 22:27.29 | Katty | a likely excuse lol |
| 22:32.43 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 22:34.06 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:36.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:50.56 | *** join/#asterisk LinoSP (~LinoSP@190.42.18.19) |
| 22:55.12 | Dovid | hi all. |
| 22:55.30 | Dovid | has anyone ever tried to have Asterisk record sound files directly to mySQL ? |
| 22:59.55 | paulc | Like.. store audio files as BLOBs in a MySQL database? |
| 23:03.41 | [TK]D-Fender | Dovid: Where do you see this option? |
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| 23:52.45 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
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| 23:58.38 | *** join/#asterisk Psil0Cybin (~psil0cybi@unaffiliated/psil0cybin) |
| 23:58.52 | Psil0Cybin | hey guys I just have a stupid question if I am using Asterisks, where do i get my VoIP service from? |