| 00:00.31 | tristero | I'm trying to connect a Panasonic PBX extension to an Asterisk FXO channel, but when it rings Asterisk doesn't "pick up"; it doesn't execute the corresponding extension in my dialplan. Any suggestions? |
| 00:03.17 | ChannelZ | Do you mean FXS? IE the Panasonic system thinks asterisk is a phone, not a phone line? |
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| 00:11.19 | iulhk | is there any possibility to override gsm outbound lines caller id? |
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| 00:23.55 | tristero | ChannelZ: It's an FXO channel (i.e. on a 4-port FXO card) with FXS signalling, since it should see the Panasonic extension as an "incoming call". |
| 00:24.12 | iulhk | do we have any functoinality where asterisk set caller id for outbound pri lines? |
| 00:27.51 | ChannelZ | tristero: ok well assuming the Panasonic side is really using the right signalling, "the extension" asterisk sees will only be 's' |
| 00:30.35 | tristero | ChannelZ: yes, but it's not getting that far. all I see (when debug is set) is ANALOG_EVENT_RINGBEGIN and ANALOG_EVENT_RINGOFFHOOK from sig_analog.c; no "Executing [s@doorphone]" |
| 00:31.35 | tristero | I can use a plain analog phone on that extension and it rings fine, so I thought Asterisk would detect it |
| 00:32.18 | ChannelZ | What do you mean? You plug a phone into that port? |
| 00:32.50 | tristero | Only t test that the Panasonic PBX is ringing that extension. |
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| 00:32.54 | tristero | s/t/to/ |
| 00:34.04 | ChannelZ | So you need an FXS port on the asterisk side, not FXO. |
| 00:34.16 | ChannelZ | Of ypi |
| 00:34.19 | ChannelZ | oops |
| 00:34.38 | ChannelZ | if you're saying you're plugging an analog phone into a port on your Panasonic that you are otherwise trying to connect to Asterisk |
| 00:35.22 | ChannelZ | actually I'm completely confused about your physical setup |
| 00:35.41 | m0sphere | with agi and php scripts, i have a function in my php script, and the call proceeds, as soon as i call the function 'functionName();' the call drops right away, is there a trick to calling functions or do i have an error somewhere? |
| 00:36.14 | ChannelZ | m0sphere: your script is probably broken and terminating completely |
| 00:38.04 | ChannelZ | tristero: re-reading what you've said you are doing I think your ports are right so ignore what I said.. |
| 00:38.28 | tristero | ChannelZ: I'm pretty sure I need an FXO port. Asterisk has to "answer" the extension just like it would answer a call from the CO. And using a handset on the Panasonic port "proves" that it's signalling correctly, at least as far as making it ring. |
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| 00:40.15 | ChannelZ | tristero: yes I just misunderstood your direction and am distracted doing other things at the same time |
| 00:41.14 | ChannelZ | start by pb-ing your chan_dahdi.conf |
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| 00:44.37 | ChannelZ | and are you running with some console verbose on? it should be saying *something* assuming the channel is configured in some fashion |
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| 00:51.37 | m0sphere | http://pastebin.com/pr2Yefgk it doesn't seem to call functionTest, or it does, but there's no output to the cli |
| 00:55.54 | ChannelZ | Seeing the asterisk side won't help much, it's your PHP that is failing |
| 00:56.35 | ChannelZ | Do you have PHP logging on? try looking at /var/log/php it's probably giving you a syntax error of some sort |
| 00:58.50 | ChannelZ | oh nm totally missed you included the script (I'm delerious today) |
| 01:00.59 | m0sphere | I tried $agi2 = new AGI(); inside the function, and then the phone call just hangs |
| 01:01.08 | m0sphere | and by hangs i mean doesn't hang up |
| 01:01.13 | m0sphere | and no output to cli |
| 01:01.30 | ChannelZ | $agi is not global |
| 01:01.39 | ChannelZ | so inside your function $agi is uninitialized |
| 01:02.00 | ChannelZ | you need to say "global $agi;" first to bring it into the function's scope. Or pass it in. |
| 01:03.32 | ChannelZ | (inside your function that is) |
| 01:04.09 | m0sphere | ahhhh |
| 01:04.11 | m0sphere | ty |
| 01:04.12 | ChannelZ | http://pastebin.com/R1p7nTFu |
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| 02:03.02 | tristero | ChannelZ: chan_dahdi.conf is here: http://paste.debian.net/67436/ |
| 02:03.53 | tristero | and debug output is here: http://paste.debian.net/67437/ |
| 02:04.29 | ChannelZ | Do you have multiple TDM cards? |
| 02:04.59 | tristero | yes, one 8-port FXS and one 4-port FXO |
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| 02:08.19 | ChannelZ | Hmm I've never used signalling=auto, assuming it's doing the right thing - what does 'dahdi show channel 12' say for Signalling Type? |
| 02:09.07 | tristero | FXS Kewlstart |
| 02:09.39 | tristero | I use that same card to answer my incoming PSTN calls, and it handles those fine |
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| 02:10.20 | ChannelZ | I can only wonder if the panasonic side is doing something odd. |
| 02:11.34 | tristero | Probably; but googling seemed to show others handling incoming calls from Panasonic PBX, so I think I'm just doing something wrong |
| 02:11.57 | ChannelZ | assuming of course your 'doorphone' context of extensions.conf exists and should otherwise do something logical and it's not just silently barfing. I asked before, do you have some console verbosity on? |
| 02:12.13 | tristero | yes, both verbose and error = 3 |
| 02:12.22 | tristero | s/error/debug/ |
| 02:15.58 | ChannelZ | well it's seeing the ring but I don't know why it's not going any further. You might try stripping back channel 12, turn callerid off for it to make sure something there isn't bungling things up.. show your extensions.conf |
| 02:19.05 | tristero | yes, I'll try that. there's also hwec on that card, I need to try disabling that too |
| 02:20.52 | tristero | extension is just this: http://paste.debian.net/67438/ |
| 02:21.35 | tristero | but none of it appears in the log, so I'm pretty sure it isn't getting there (other parts of the dialplan show up when incoming calls are received, for example) |
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| 06:10.00 | AcidSlide | hello.. |
| 06:10.05 | AcidSlide | i'm new to asterisk.. |
| 06:10.32 | AcidSlide | i'm trying to setup a demo asterisk.. what do you recommend? using SIP or IAX for softphones? |
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| 06:19.15 | [TK]D-Fender | SIP |
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| 11:05.22 | m0sphere | is there any way to stop these? Executing [9011972597562753@from-pstn:1] Set("SIP/1.2.3.4-00000c7a", "__FROM_DID=9011972597562753") in new stack |
| 11:06.03 | m0sphere | this shit floods my server for hours at a time |
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| 11:32.56 | kaldemar | m0sphere: don't allow unauthenticated calls to your box. allowguest=no. |
| 11:35.09 | kaldemar | alwaysauthreject=yes too if you don't have it. many people use for example fail2ban to block attacker ip addresses. |
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| 12:32.24 | danfromuk | Hi, I'm trying to switch to cdr_adaptive_odbc, but the calldate field is always '0/0/0 00:00'. Any idea what I'm doing wrong? Is the field name wrong? |
| 12:36.19 | danfromuk | Is it possible to output all the CDR() variables to the cli so I can check? |
| 12:38.02 | danfromuk | !book |
| 12:39.37 | roswell | danfromuk, possibly you sql server cannot recognize timestamp your asterisk feeds it with. what's the server you use? |
| 12:39.59 | danfromuk | mysql |
| 12:40.07 | danfromuk | calldate is of type DATETIME |
| 12:40.21 | danfromuk | Works with cdr_odbc |
| 12:40.34 | danfromuk | Does the type change for adaptive? |
| 12:41.08 | roswell | not actually. make sure there's no any alias in cdr_adaptive_odbc.conf for that column |
| 12:41.49 | danfromuk | I'll set up a timestamp field and alias that and see what happens |
| 12:42.35 | roswell | well, the datetime type is just fine |
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| 12:49.53 | danfromuk | Didnt seem to help |
| 12:51.51 | roswell | strange. well try setting mysql to log every query and see what comes from asterisk |
| 12:53.41 | danfromuk | Good idea. But i'm sure its asterisk not adding the information. Its also not adding the custom column i made. |
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| 12:54.56 | AlphaPiJR | hi |
| 12:55.07 | AlphaPiJR | what about cisco spa-3000 clone on aliexpress? |
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| 12:55.19 | AlphaPiJR | is working well? |
| 12:57.26 | WIMPy | AlphaPiJR: You are aware that the analog stuff had the possibility for different qualities? |
| 12:58.05 | danfromuk | I dont think i would ever buy a clone. |
| 12:58.42 | roswell | yet to say, as with any clone, chance of success is 50%. it's either works or it doesn't. it's all up to a customer |
| 12:59.43 | WIMPy | Analog doesn't just work or not it can work in all qualities. |
| 13:00.12 | WIMPy | Which is what I wanted to point out. |
| 13:00.20 | roswell | btw danfromuk have you reloaded the odbc module after applying modifications to its configuration? |
| 13:01.41 | AlphaPiJR | it's for home/experiment pourpose |
| 13:01.48 | AlphaPiJR | it's not for professional use |
| 13:02.06 | AlphaPiJR | however nobody have personal experience with such device? |
| 13:02.24 | WIMPy | Bad quality gives more space for experimentation. |
| 13:03.31 | danfromuk | roswell: yes |
| 13:04.13 | roswell | just asking to make sure |
| 13:05.48 | AlphaPiJR | WIMPy: did you use only A grade stuff? |
| 13:06.26 | WIMPy | AlphaPiJR: No, I don't use analog. |
| 13:08.23 | mic_ | hello, dahdi made by openvox contains their patches? |
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| 13:56.20 | zemmali-voip | please guys!i need help to create web service: -Insert , Edit , Detele compte SIP and update file from asterisk with JAVA! |
| 13:56.20 | zemmali-voip | best regard ! |
| 14:10.47 | Gugge | zemmali-voip: just use realtime, and edit the db, or generate the conf files from your software. |
| 14:18.34 | Gugge | zemmali-voip: if you have any asterisk related questions, ask here |
| 14:19.08 | zemmali-voip | thanks Gugge |
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| 14:27.22 | ziz212 | hi |
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| 15:15.01 | ziz212 | Guys, What cause Asterisk TLS not work in internet ?... I have tried TLS and SRTP in LAN and it work perfectly. But I have tried it with internet it doent work. Still not getting any request in Asterisk Debug ? pls help |
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| 15:26.19 | ghost75 | is someone using rtcp channel variables? |
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| 17:06.15 | redotis | chan_sip.c:23475 handle_response_peerpoke: Peer 'vilhelm' is now Lagged. |
| 17:07.12 | redotis | Is there a way to change the timeouts for when a sip connection lags? I'm having a problem with folks using satellite connections staying "connected" and in a conference? |
| 17:07.43 | redotis | The delay is not an issue but they seem to drop out because I think sip thinks they are gone when it's just the delay. |
| 17:09.16 | [TK]D-Fender | qualify <- |
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| 17:23.14 | redotis | What's the deal with qualify |
| 17:23.31 | redotis | Oh...take it out and it doesn't check or kick them off the connection? |
| 17:23.37 | [TK]D-Fender | That is what is timing out |
| 17:23.50 | [TK]D-Fender | lagged = not responding within the timefrime |
| 17:23.56 | redotis | ok cool |
| 17:24.10 | redotis | do you think if I take that out they will stay in the conference? |
| 17:24.40 | file | qualify doesn't impact active SIP sessions |
| 17:25.00 | redotis | ok so in my case what does that mean |
| 17:25.18 | redotis | If I remove qualify, will they likely stay in the conference? |
| 17:25.29 | file | no, while it may indicate a problem that in and of itself is not the problem |
| 17:25.53 | redotis | so they won't stay in the conference? |
| 17:26.08 | file | what? |
| 17:26.14 | redotis | jesus |
| 17:26.28 | redotis | i'll experiment |
| 17:26.35 | redotis | thanks TKD |
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| 19:43.06 | danfromuk | <PROTECTED> |
| 19:43.16 | danfromuk | As you can see, the calldate field is not being used. |
| 19:46.08 | pabelanger | danfromuk, does calldate exist in your database? |
| 19:46.48 | danfromuk | Yes, and it works perfectly with cdr_odbc |
| 19:47.11 | cusco | isn't calldate a alias for 'start' ? |
| 19:47.42 | cusco | create the column 'start', 'answer' and 'end' |
| 19:47.46 | cusco | reload the module |
| 19:47.50 | cusco | and see what happens |
| 19:48.33 | pabelanger | danfromuk, *CLI> core show function CDR |
| 19:48.50 | pabelanger | those are the fields |
| 19:48.58 | pabelanger | if you want them named something else, setup a mapping |
| 19:49.39 | danfromuk | ok, so to switch from cdr_odbc to cdr_adaptive, the field called calldate changes to 'start', or I need to alias it back to calldate? |
| 19:50.12 | cusco | just use start, answer and end.. great for whatever comes along lol |
| 19:50.50 | danfromuk | I know, but the field 'calldate' is already integrated into our billing platform |
| 19:51.11 | cusco | you can alias it in the config, I guess |
| 19:51.18 | pabelanger | danfromuk, then setup a mapping |
| 19:51.25 | pabelanger | cdr_custom.conf |
| 19:52.04 | pabelanger | maybe not |
| 19:52.08 | pabelanger | I think that is just for csv |
| 19:52.42 | pabelanger | cdr_adaptive_odbc.conf |
| 19:52.51 | pabelanger | alias start => calldate |
| 19:52.57 | pabelanger | reload |
| 19:54.14 | danfromuk | Bingo. Thanks for the help |
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| 20:15.22 | Jinxed- | <Jinxed-> I was hoping to do some research on E1 based pbx's any other |
| 20:15.35 | Jinxed- | <PROTECTED> |
| 20:18.18 | cusco | trasition ? |
| 20:18.31 | cusco | I didn't quite understand what you're looking for |
| 20:18.37 | cusco | care to develop on that? |
| 20:20.11 | ghost75 | could somebody check please whats in the channel(rtpqos.audio,all) variable in h ext ? |
| 20:20.50 | ghost75 | the only thing i get is number of packets |
| 20:21.30 | ghost75 | in sip show channelstats everything is displayed |
| 20:22.14 | cusco | I'm not quite sure what it is that you're asking, but is it possible that in the h extension, variables are alredy gone, as is the call? |
| 20:23.31 | ghost75 | then number of packets would be also empty |
| 20:23.56 | cusco | I'm not sure, as some stuff stays |
| 20:24.06 | cusco | and some other goes, in h exten |
| 20:24.24 | cusco | cdr.conf even has a option to <something>beforehexten: yes/no |
| 20:24.43 | cusco | I guess someone that would understand should answer, however |
| 20:25.05 | ghost75 | can the variables be displayed during call? |
| 20:26.49 | danfromuk | ghost75: make a macro that calls DumpChan and run it using a feature code |
| 20:26.53 | danfromuk | *CLI> core show application DumpChan |
| 20:28.25 | ghost75 | press key on phone to do gosub? |
| 20:29.04 | danfromuk | make a macro in your extensions.conf that calls DumpChan |
| 20:29.29 | ghost75 | macro is deprecated |
| 20:29.29 | danfromuk | Then look at features.conf to see how to create a *code to run the macro during the call |
| 20:30.25 | danfromuk | ok, i didnt know that. but it still works in asterisk 11. |
| 20:32.59 | ghost75 | i remember something feature codes dont work on sip phones |
| 20:33.40 | danfromuk | Never heard that, and I use them all the time. So not sure where you got that from |
| 20:34.05 | cusco | we use features.conf with sip, no problems |
| 20:34.06 | danfromuk | I have a macro that starts/stops call recording when the user dials *1 during the call. |
| 20:35.33 | ghost75 | testfeature => *9,caller,Macro,test |
| 20:35.36 | ghost75 | not sure if correct |
| 20:37.06 | danfromuk | I think you also have to do Set(__DYNAMIC_FEATURES=testfeature) in your dial plan, before the Dial command. |
| 20:40.51 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:47.31 | ghost75 | No such context 'macro-test' for macro 'test' |
| 20:47.46 | ghost75 | i didnt put macro-test anywhere |
| 20:47.48 | danfromuk | Pastebin your macro, including the [contextname] |
| 20:48.29 | ghost75 | [test] |
| 20:48.36 | ghost75 | exten => s,1,Dumpchan(0) |
| 20:48.55 | danfromuk | change it to [macro-test] |
| 20:49.09 | danfromuk | And i think its just Dumpchan() |
| 20:49.11 | danfromuk | no 0 |
| 20:49.20 | danfromuk | actually. i'm wrong |
| 20:49.20 | ghost75 | 0 is the level |
| 20:49.28 | danfromuk | yes |
| 20:49.29 | danfromuk | correct |
| 20:49.48 | ghost75 | but why is going macro-test, maybe because i did testfeature => *9,caller,Macro,test |
| 20:50.45 | ghost75 | or all macros need the name macro- in context? |
| 20:51.04 | danfromuk | all macros are prefixed with macro- |
| 20:51.08 | danfromuk | otherwise its just a normal context |
| 20:52.38 | ghost75 | HAHA |
| 20:52.44 | ghost75 | no rtcp listed |
| 20:52.56 | danfromuk | what do you mean? |
| 20:53.16 | ghost75 | it does not really list all variables, just a few |
| 20:53.25 | ghost75 | no channel variables at all |
| 20:53.53 | ghost75 | no rtcp channel variables ^^ |
| 20:53.55 | danfromuk | I use DumpChan(30) |
| 20:55.14 | ghost75 | now 30 doesnt display anything at all |
| 20:55.59 | ghost75 | DumpChan([<min_verbose_level>]) |
| 20:56.04 | ghost75 | Displays information on channel and listing of all channel variables. If min_verbose_level is specified, output is only displayed when the verbose level is currently set to that number or greater. |
| 20:56.16 | danfromuk | I've used Dumpchan(30) and it outputs all the channel variables, without any problems. |
| 20:56.45 | danfromuk | What are you looking for exactly? |
| 20:56.49 | ghost75 | so do you see any rtpqos variables? |
| 20:57.49 | danfromuk | This is my output |
| 20:57.50 | danfromuk | http://pastebin.com/xAvqmqxa |
| 20:58.21 | ghost75 | i see same |
| 20:59.22 | ghost75 | this is what i see in h: http://pastebin.com/ay423SM6 |
| 20:59.43 | danfromuk | ok. in that case, in the same macro, try something like Verbose(${channel(rtpqos)}) |
| 20:59.50 | danfromuk | i'm guessing by the way |
| 21:00.23 | danfromuk | You might need to change that a bit |
| 21:00.23 | danfromuk | The option rtpqos takes two additional arguments |
| 21:00.27 | danfromuk | http://www.voip-info.org/wiki/view/Asterisk+func+channel |
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| 21:00.54 | ghost75 | this is in dialplan |
| 21:01.14 | danfromuk | And do you get the result you want? |
| 21:01.29 | ghost75 | i try now verbose |
| 21:01.34 | danfromuk | It might be that you can only get the information you want once the call is bridged. So use the macro to do that. |
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| 21:03.45 | ghost75 | waits for the jitter |
| 21:06.58 | danfromuk | What jitter are you getting? Have you tried a different codec? |
| 21:07.59 | danfromuk | What type of connection are you using? Have you tested the connection? Try this and see what results you get http://myspeed.visualware.com/indexvoip.php |
| 21:08.00 | ghost75 | my android always jitters a bit |
| 21:08.19 | danfromuk | Choose a node thats labelled Quality Testing Available |
| 21:09.31 | ghost75 | nice page |
| 21:10.40 | ghost75 | test didnt work |
| 21:10.45 | ghost75 | unable to measure |
| 21:11.01 | danfromuk | Ive never had a problem. Try a different node |
| 21:13.19 | ghost75 | http://pastebin.com/V8HqU0R3 so maybe cannot expect more to display |
| 21:13.59 | danfromuk | yep. i think so |
| 21:14.27 | file | the RTP variables are only set at hangup, if you want to grab the current values you need to use the CHANNEL dialplan function |
| 21:15.03 | ghost75 | after hangup is fine |
| 21:17.29 | ghost75 | but it displays only of one phone, sip show channelstats displays both |
| 21:18.18 | file | correct... sip show channelstats is a command which specifically shows all stats for all SIP channels |
| 21:18.28 | file | the variables on the channel are about the RTP for that channel |
| 21:20.34 | ghost75 | so how i can get data from the remote channel |
| 21:20.54 | file | you can't there |
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| 21:21.14 | ghost75 | :< |
| 21:21.52 | ghost75 | just needed output from channelstats in cdr |
| 21:22.14 | file | no, you want the RTP statistics for the bridged channel |
| 21:23.49 | ghost75 | yes, the sip show channelstats |
| 21:33.56 | ghost75 | danfromuk: that page rocks |
| 21:35.38 | ghost75 | 57ms jitter with torrent enabled :> |
| 21:36.51 | danfromuk | Yes, its a very useful page. |
| 21:37.56 | ghost75 | only having 384kbit upload lol |
| 21:40.56 | danfromuk | You should try out g729. It doesnt cost that much, and you should get some good results. |
| 21:41.05 | danfromuk | its only 9kb for each side of the call |
| 21:41.21 | ghost75 | my isp doesnt support it |
| 21:41.58 | ghost75 | or lets say different |
| 21:42.16 | ghost75 | mobile calls come in with g729 but normal calls are g711 |
| 21:42.17 | danfromuk | ?? isp's have nothing to do with g729 |
| 21:42.44 | ghost75 | sip account from isp |
| 21:43.14 | danfromuk | Where is your asterisk server? |
| 21:43.28 | danfromuk | The same network as your android phone? |
| 21:43.32 | ghost75 | local of course |
| 21:43.59 | danfromuk | Why 'of course'? Ours is in a data centre with 1Gb upload/download |
| 21:44.16 | ghost75 | its only home server |
| 21:44.33 | danfromuk | ok. so maybe see what codecs they have available. |
| 21:45.00 | ghost75 | once i tried only g729 in asterisk and i couldnt call all persons |
| 21:45.41 | danfromuk | You probably need to buy a license |
| 21:46.00 | ghost75 | g729 itself was working fine |
| 21:46.21 | danfromuk | If you cant call all people, then its probably a licensing issue. |
| 21:46.37 | ghost75 | i could call some people |
| 21:46.46 | ghost75 | that was confusing me |
| 21:47.13 | [TK]D-Fender | [16:40]danfromukits only 9kb for each side of the call <- quite a bit more in the real world... |
| 21:47.31 | danfromuk | [TK]D-Fender: ok, but its better than most other codecs |
| 21:47.37 | [TK]D-Fender | true |
| 21:47.58 | ghost75 | disallow=all and allow=g729 in sip.conf |
| 21:48.35 | danfromuk | ghost75: if you havent bought any licenses, then its a fight to get asterisk to work with g729 |
| 21:49.09 | danfromuk | I tried for a few days, then decided that i'm wasting my time and bought licenses. |
| 21:49.10 | ghost75 | they are not bought but i have some :) |
| 21:49.25 | danfromuk | but you have to make sure that your android phone supports g729. usually its an addon |
| 21:49.34 | danfromuk | What app are you using? |
| 21:49.37 | [TK]D-Fender | not a question of "fight" * doesn't come with transcoding support natively. You have to pay for that |
| 21:49.52 | ghost75 | about android i dont care, important is my cisco phone |
| 21:50.17 | danfromuk | [TK]D-Fender: i tried to avoid transcoding, but my asterisk still thought there was some going on. Couldnt pin point where. |
| 21:50.33 | [TK]D-Fender | there is no "think". |
| 21:50.37 | ghost75 | to any mobile numbers the g729 was working without issue |
| 21:50.46 | [TK]D-Fender | It was expected to by an action that was taking place |
| 21:51.03 | [TK]D-Fender | next "mobile numbers" know nothing of codecs |
| 21:51.19 | [TK]D-Fender | your PEERS and the services on the other side of that is another matter |
| 21:51.32 | [TK]D-Fender | Stop talking like a phone number cares or knows about "codec" |
| 21:51.50 | ghost75 | maybe my isp allows g729 only for mobile no idea |
| 21:52.20 | [TK]D-Fender | I can tell about "no idea" .. because I don't see you LOOKING. |
| 21:53.22 | ghost75 | prolly did the mistake to enable *only* g729 during that time |
| 21:54.02 | ghost75 | if the remote end is also ip based and has no g729 then boom |
| 21:54.39 | [TK]D-Fender | "ip based"? |
| 21:54.49 | [TK]D-Fender | Where do you keep coming up with this terms? |
| 21:54.57 | [TK]D-Fender | these* |
| 21:55.05 | ghost75 | voip |
| 21:55.13 | ghost75 | instead analog or isdn |
| 21:55.30 | [TK]D-Fender | There is no "remote end" when you call, it goes to your provider. That is ALL |
| 21:55.36 | [TK]D-Fender | There is no "remote end" |
| 21:55.43 | [TK]D-Fender | the provider sends it via the ACTUAL PSTN |
| 21:55.59 | [TK]D-Fender | And this has nothng to do with you or your codecs to your provide |
| 21:57.21 | ghost75 | so he takes care of codec translation? |
| 21:57.40 | [TK]D-Fender | there is no more codec at that point |
| 21:57.50 | [TK]D-Fender | You are inventing things that don't exist |
| 21:58.10 | [TK]D-Fender | Why are you not looking at the actual calls right now? |
| 21:58.39 | danfromuk | ghost75: all you need to do is ensure the provider can support g729 calls. They will usually accept it for all calls. They won't change it based on the number dialled. |
| 21:59.34 | ghost75 | i thought that isp just forwards all ip and sip stuff |
| 21:59.54 | danfromuk | Ignore what they do. They'll sort themselves out. |
| 22:00.44 | danfromuk | If they offer g729, then they'll deal with the issues regarding forwarding the call to the target. |
| 22:01.05 | [TK]D-Fender | [16:59]ghost75i thought that isp just forwards all ip and sip stuff <-No. |
| 22:01.17 | ghost75 | they might offer g729 only inofficially because their routers come always with g711 |
| 22:01.29 | danfromuk | routers? |
| 22:01.37 | [TK]D-Fender | ghost75: the world is not voip <- |
| 22:01.47 | [TK]D-Fender | ghost75: Get this nonsense out of your head |
| 22:02.34 | [TK]D-Fender | ghost75: The real PSTN is G.711 over TDM and your rpovider talks that out to the rest of the world. IT takes whatever you give it (that they accept) and translates |
| 22:02.54 | WIMPy | Unlikely. |
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| 22:09.01 | ghost75 | danfromuk: most of the customers here are using routers where voip is configured by isp with g711 |
| 22:09.36 | danfromuk | Have you asked the ISP whether they support g729? |
| 22:09.51 | ghost75 | they will not answer such questions |
| 22:10.01 | [TK]D-Fender | And you're wasting time |
| 22:10.03 | danfromuk | Have you tried? |
| 22:10.22 | [TK]D-Fender | ghost75: All of this guessing is worthless. |
| 22:10.26 | ghost75 | yes, as written above |
| 22:10.34 | WIMPy | I'm not even sure if we have a word for "support". |
| 22:10.35 | [TK]D-Fender | ghost75: The call TELLS you what they ar offering. |
| 22:10.43 | [TK]D-Fender | ghost75: Wake up. |
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| 22:14.22 | danfromuk | ghost75: what [TK]D-Fender is saying, is you can look at the sip debug to see what they support |
| 22:14.33 | ghost75 | yes i know |
| 22:14.56 | [TK]D-Fender | danfromuk: He is sitting here inventing terms assuming the "remote caller" is a factor and not looking at his own calls. |
| 22:15.32 | ghost75 | but sure that *every isp* is handling *all* sip traffic over g.711/TDM even if the call stays at one isp? |
| 22:16.00 | [TK]D-Fender | ghost75: ISP = Internet Service Provider. |
| 22:16.01 | [TK]D-Fender | not voice |
| 22:16.03 | WIMPy | That would be stupid. |
| 22:16.05 | [TK]D-Fender | ~itsp |
| 22:16.06 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 22:16.14 | [TK]D-Fender | get your terms right |
| 22:16.20 | [TK]D-Fender | And the other side is NOT YOUR PROBLEM |
| 22:16.23 | ghost75 | its all from one |
| 22:16.32 | [TK]D-Fender | Doesn't matter |
| 22:16.37 | [TK]D-Fender | LOOK AT YOUR CALL |
| 22:17.44 | [TK]D-Fender | They either offer a codec or they don'ty |
| 22:17.48 | [TK]D-Fender | This is not a guessing game |
| 22:17.52 | [TK]D-Fender | it's RIGHT THERE in the call |
| 22:18.33 | danfromuk | [TK]D-Fender: as I said......Ignore what they do. They'll sort themselves out. |
| 22:18.57 | [TK]D-Fender | danfromuk: Nothing to ignore because there is nothing to see. |
| 22:19.35 | [TK]D-Fender | danfromuk: He's making up nonsense and doing everthing except looking at his call. 4 stupid words in * CLI. That's what he is refusing to do. |
| 22:19.36 | ghost75 | well i only remember that i had problems on specific calls but not on all when i only had enabled g729 |
| 22:20.03 | [TK]D-Fender | ghost75: These stories are a waste of time. |
| 22:21.35 | carrar | ghost stories? |
| 22:22.27 | ghost75 | if they dont officially support, no one guarantees a working system |
| 22:22.44 | carrar | http://www.youtube.com/watch?v=9A0jRVd7TZQ&feature=related |
| 22:23.02 | [TK]D-Fender | ghost75:Why are you still talking about "if"? |
| 22:23.38 | carrar | obakechan! |
| 22:23.39 | ghost75 | do you know the difference about inofficial and official? |
| 22:24.01 | [TK]D-Fender | There is no such thing |
| 22:24.10 | [TK]D-Fender | Their call offers what it offers |
| 22:24.18 | [TK]D-Fender | You are inventing terms and making excuses for not looking |
| 22:24.22 | WIMPy | And no such things as guarantees anyway. |
| 22:24.24 | [TK]D-Fender | This is a waste of all of our time |
| 22:24.36 | ghost75 | lets stop it |
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