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| 00:15.45 | *** join/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 00:17.08 | root52 | OK So it seems that using UnixODBC and FreeTDS with querys in func_odbc.conf I can only run one query per connection. Is there a way to tell asterisk to close and make a new connection everytime it "goes" for a query in func_odbc.conf. |
| 00:17.19 | root52 | Oh and hello all! |
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| 01:14.04 | stranf | hi all, for a project i needed to take the audio stream from a conference call and introduce it into a gstreamer pipeline |
| 01:14.29 | stranf | can you please suggest a good way to do it? |
| 01:15.21 | root52 | Again I fail.... res_odbc.conf share_connections => no ;-( |
| 01:23.57 | *** join/#asterisk stope (~nobody@sud-cable-cmts3-69-60-242-63.vianet.ca) |
| 01:33.25 | stope | Does anyone see why the call gets dropped upon parking (sometimes) http://pastebin.com/3tiE7sKr |
| 01:33.58 | stope | Aastra 53i with asterisk 1.6.2.15 |
| 01:34.22 | stope | using the default call parking - lot 701, 702 ... etc. |
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| 03:05.18 | carrar | You forgot a period |
| 03:05.20 | carrar | thats why |
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| 03:18.58 | jplank | Is there any video support in meetme? |
| 03:22.10 | carrar | http://www.gnugk.org/index.html |
| 03:22.59 | kaldemar | jplank: no |
| 03:24.08 | jplank | app_conference is the only way? |
| 03:24.56 | jplank | carrar: h323 :( |
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| 03:26.15 | carrar | Then might I suggest a nice pretty shiny red Geiger counter instead of h323 |
| 03:26.15 | carrar | http://www.blackcatsystems.com/GM/GeigerCounters.html |
| 03:28.00 | jplank | sorry, I get get the joke/relation. |
| 03:28.08 | jplank | s/get/don't/ |
| 03:28.59 | jplank | I don't know h323 well enough, and the project I'm working on isn't really enough incentive. |
| 03:32.31 | pabelanger | jplank: keep an eye on trunk, there is alot of work going on right now with codec negotiation, maybe even video |
| 03:33.34 | therawr | video? |
| 03:33.36 | therawr | orly pabelanger |
| 03:34.15 | pabelanger | therawr: not 100%, but dvossel is doing alot of work on codecs ATM |
| 03:34.20 | therawr | nice! |
| 03:35.45 | pabelanger | therawr: he is working on CELT support ATM |
| 03:35.47 | therawr | oh _REALLY_ |
| 03:37.16 | pabelanger | ya, I suggest keeping an eye on http://svn.digium.com/view/asterisk/team/dvossel/ |
| 03:37.52 | jplank | pabelanger: hd_confbridge? |
| 03:38.34 | pabelanger | jplank: indeed, we've been using it a digium for about 1 month now |
| 03:39.19 | jplank | I know this is a probably a pipedream, but is it going to support multiple video streams at once? |
| 03:41.13 | pabelanger | unknown, video support has not been added, I don't see why it couldn't be. |
| 03:41.21 | pabelanger | check out https://wiki.asterisk.org/wiki/display/AST/Media+Overhaul for more information |
| 03:41.25 | *** join/#asterisk cesurasean (~Sean_Brad@server.simplewebhosting.us) |
| 03:41.31 | Kobaz | jplank: if you want video conferencing, check out woomera by dialogic |
| 03:41.34 | pabelanger | sorry, https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal |
| 03:42.18 | pabelanger | jplank: specifically Handling Multiple Media Streams |
| 03:44.22 | *** join/#asterisk kaushal (~kaushal@182.72.14.170) |
| 03:47.39 | kaushal | hi |
| 03:47.40 | kaushal | Does Asterisk supports VXML ? |
| 03:47.48 | kaushal | is there a way to make it work by using some sort of plugin or API's ? |
| 03:48.02 | jplank | Kobaz: http://www.projectdiastar.org/ ? |
| 03:48.09 | pabelanger | kaushal: no native support |
| 03:49.17 | jplank | pabelanger: so basically, look else where? ;) |
| 03:50.16 | Kobaz | jplank: yeap that's it |
| 03:51.10 | kaushal | pabelanger: so any API's ? |
| 03:52.17 | pabelanger | kaldemar: no, not in Asterisk. I believe there are 3rd party apps, but I don't know of any. |
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| 04:02.19 | kaushal | pabelanger: ok |
| 04:09.42 | stranf | hi, I needed to take audio from the conference and pipe it to a gsteamer pipeline, how can i do that? |
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| 04:30.26 | stranf | is writing the stream to a named-pipe and then reading from that in gstreamer a good idea? |
| 04:30.32 | stranf | any suggestions? |
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| 05:01.25 | Mango | ~itsplist-ca |
| 05:01.25 | infobot | Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca , http://www.voip.ms |
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| 06:07.18 | schmidts | TGIF |
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| 06:13.20 | wdoekes2 | morning |
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| 06:13.48 | schmidts | hello walter ;) |
| 06:13.58 | schmidts | how are you? |
| 06:14.05 | wdoekes2 | hi stefan, just fine |
| 06:15.19 | schmidts | atleast its friday best working day of the week ;) |
| 06:15.45 | WiretapSeven | friday will be over for me in 6 hours :p |
| 06:15.49 | WiretapSeven | well, 5h45 |
| 06:15.55 | kleszcz | morning |
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| 06:16.27 | schmidts | :( stillt 8h45 for me |
| 06:17.38 | kleszcz | yeah me too schmidts .. |
| 06:20.01 | WiretapSeven | schmidts, I mean till its saturday :P |
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| 06:27.53 | schmidts | WiretapSeven ok then its fine :D |
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| 06:41.38 | Iiiak | plop |
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| 07:06.13 | jacc0 | good morning all |
| 07:07.17 | jacc0 | i'm experiencing instabilitie with multiple asterisk 1.8.3 reatime setups |
| 07:07.35 | jacc0 | and I was reading this : |
| 07:07.36 | jacc0 | http://pastebin.com/iKGX4tTU |
| 07:07.53 | jacc0 | "set wait_timeout internal to the MySQL server to 864000 or higher" |
| 07:08.00 | jacc0 | where do I set this timeout |
| 07:08.02 | jacc0 | ? |
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| 07:15.17 | wdoekes2 | jacc0: mysql> show variables like '%wait%'; |
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| 07:35.55 | jacc0 | ty |
| 07:39.53 | jacc0 | I added wait_timeout=86400 |
| 07:40.01 | jacc0 | under [mysqld] in my.cnf |
| 07:40.24 | jacc0 | + an extre 0 |
| 07:40.28 | jacc0 | *exrta |
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| 07:51.02 | stranf | hi all, sorry for the repeated identical messages, but I havent got any replies |
| 07:51.08 | stranf | <PROTECTED> |
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| 07:52.19 | stranf | <PROTECTED> |
| 07:54.26 | kaldemar | stranf: someone else was asking about getting a stream to gstreamer, and iirc he did it with a named pipe. |
| 07:56.04 | stranf | is there a direct way that i could do this? |
| 07:56.09 | jacc0 | @stranf: I'm not sure what you want to do with the stream; but i've been using ices and icecast and running vlc in backgroud to broadcast audio |
| 07:56.36 | stranf | i want to mix it with a video, realtime |
| 07:56.37 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 07:58.08 | jacc0 | @stranf: there is an gstreamer plugin for icecast |
| 07:59.20 | kaldemar | stranf: you could try originating a call to the conference and connecting it to app Record that writes to the pipe. |
| 08:00.48 | stranf | kaldemar, so write it to a named pipe and use it as a source for gstreamer? |
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| 08:01.45 | kaldemar | stranf: something like that. |
| 08:03.11 | stranf | well any idea how ices is implemented, because what i am looking for is similar, only instead of stream to a Icecast server I am doing it locally |
| 08:04.59 | stranf | anyway thanks kaldemar , jacc0 |
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| 10:39.55 | *** join/#asterisk casix (~casix@155.134.17.95.dynamic.jazztel.es) |
| 10:39.56 | casix | hello |
| 10:43.08 | casix | I have a problem asgning a valor with quotes to a var. I want to assign "value" with the quotes but the quotes disapear. I have tired to keep them with \ set(var=\"value\") but they disapear neither. How can I do that? |
| 10:43.17 | *** join/#asterisk Invade (~M@173.192.195.187-static.reverse.softlayer.com) |
| 10:44.02 | Invade | Hi, just wondering if anyone knows how i can set a global variable for all sip extensions |
| 10:44.17 | Invade | basically i have a folder in /var/lib/asterisk/sounds called audio |
| 10:44.50 | Invade | i want to be able to use audio files from ../audio/ for all extensions, without adding the option "language=audio" to each sip user |
| 10:45.38 | casix | you can create a template and create each user based on this template |
| 10:45.56 | casix | if you want to change this you only have to change the template |
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| 10:46.22 | Dryanta | the michigantelephone blog for gvoice trunks |
| 10:46.51 | Dryanta | how do you do outside routes off same dial pattern? |
| 10:47.01 | Invade | casix: im guessing that would be the best way to go about it then right? |
| 10:48.04 | Invade | just thinking, thats no different to me adding it manually that's all |
| 10:51.36 | casix | yes |
| 10:52.02 | casix | you have to add it too. But if you want to modify it later is easiest |
| 10:52.40 | casix | I don't know how to do it without having to add something to each extension... |
| 10:53.41 | Invade | casix: ok cool, thanks. |
| 10:53.50 | Invade | will go down that route heh |
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| 10:57.30 | casix | I have a problem asgning a valor with quotes to a var. I want to assign "value" with the quotes but the quotes disapear. I have tired to keep them with \ set(var=\"value\") but they disapear too. How can I do that? |
| 11:03.39 | kaldemar | how are you verifying that they disappear? |
| 11:04.27 | casix | in a noop ${var} they didn't appear |
| 11:04.49 | casix | and in a regexp it matches without quotes |
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| 11:06.11 | kaldemar | works for me in 1.8.3.2. what version are you using? |
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| 11:06.32 | casix | 1.4.26.2 |
| 11:12.35 | casix | but I can't upgrade to 1.8 :( |
| 11:14.22 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 11:16.36 | kaldemar | casix: looks like 1.4 stips the quotes. |
| 11:17.13 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 11:17.20 | kaldemar | strips that is. |
| 11:18.43 | kaldemar | wait, no. with var=\\"value\\" it does not. |
| 11:20.48 | kaldemar | Set behaves differently in newer versions. in 1.8, backslashes are included in the variable value. |
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| 11:22.04 | casix | kaldemar: ok, i will try \\ thanks :) |
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| 11:45.11 | WiretapSeven | I've googled and I've googled, but I can't find a list of differences between ast1.6 and ast1.8 other than version number? |
| 11:47.10 | *** join/#asterisk coppice (~chatzilla@m121-203-193-55.smartone-vodafone.com) |
| 11:50.59 | WiretapSeven | found it |
| 11:51.01 | *** join/#asterisk cbf (~c1@unaffiliated/hrhrhr) |
| 11:51.14 | cbf | howdy, does anyone understand this: PRI span 1/0: Provisioned, In Alarm, Down, Active |
| 11:51.22 | cbf | down AND active? |
| 11:51.30 | cbf | it's a new BT install |
| 11:51.42 | cbf | old asterisk 1.4 box doesn't seem to want to work |
| 11:53.50 | *** join/#asterisk SupaYoshi (~SupaYoshi@ip54502567.adsl-surfen.hetnet.nl) |
| 11:53.53 | SupaYoshi | Hello |
| 11:54.15 | SupaYoshi | Im using FreePBX with an SIP trunk with voipbuster, and an inbound route, but my extensions won't ring |
| 11:54.26 | SupaYoshi | IT gives some weird message on an incoming call in the CLI log |
| 11:55.30 | fauxalliance | redirects SupaYoshi to > #freepbx please have your pastebin ready |
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| 12:14.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:18.02 | cbf | anyone using BT PRI in london atm? |
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| 12:31.58 | c4rg | is it a correct AGI 'statement' in 1.6?: SET VARIABLE MONITOR_FILENAME "${STRFTIME(${EPOCH},CET,%Y-%m-%d-%H-%M-%S)}" |
| 12:33.54 | *** part/#asterisk cbf (~c1@unaffiliated/hrhrhr) |
| 12:34.28 | kaldemar | c4rg: i don't think you can use a dialplan function or a variable directly like that in AGI. |
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| 12:35.56 | kaldemar | c4rg: you probably need to use GET VARIABLE to get the value of ${STRFTIME(${EPOCH},CET,%Y-%m-%d-%H-%M-%S)} first. |
| 12:36.28 | c4rg | the trick is that strftime() is expanded when the call is answered |
| 12:36.44 | c4rg | not when this statement is sent using AGI |
| 12:36.48 | c4rg | and it worked |
| 12:36.56 | c4rg | now it works partially |
| 12:38.13 | c4rg | I either get strftime() with the default format ('%c') or with proper format, but then the time zone is set wrong |
| 12:38.17 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 12:39.45 | eject_ck | Hi all, in sip show peers result why I see 345/345 as I understand one value is defaultuser value and what about 2nd ? Can I see something like callerid info in that fields ? |
| 12:40.37 | leifmadsen | first value is [345] and other is defaultuser=345 |
| 12:40.59 | leifmadsen | they will be different if you do something like: [asterisk-toronto] |
| 12:41.05 | leifmadsen | defaultuser=asterisk-london |
| 12:41.06 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:41.35 | leifmadsen | eject_ck: you can't change what displays in the field -- you can only do a 'sip show peer 345' to see the extra data. |
| 12:41.38 | c4rg | kaldemar: I don't know if the commas shouldn't be escaped with '\' |
| 12:42.35 | leifmadsen | eject_ck: you'd probably be interested in https://issues.asterisk.org/view.php?id=19076 |
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| 13:23.24 | devon_hillard | are there major voip providers that use IAX2 instead of SIP? |
| 13:24.00 | devon_hillard | simply giving a customer a phone would work much better if you don't have to mess about in their router settings |
| 13:26.50 | psilikon | devon_hillard, voip.ms, vitelity. Just shop around |
| 13:27.51 | _Corey_ | depends on your definition of "major" I guess... :) |
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| 13:57.33 | mocker | hehe, oreilly keeps pushing back the publication date of the ebook |
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| 14:16.48 | Andrew__M | Hello All! |
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| 14:17.09 | stmaher | hi guys, |
| 14:17.20 | Becca_R | Hello |
| 14:17.25 | stmaher | I just setup a new e1 card.. and im trying to figure out groups and just make a call |
| 14:17.35 | stmaher | I did my dahdi_genconfig |
| 14:17.41 | stmaher | and the line has no alarms |
| 14:18.03 | stmaher | just trying to find out what group it was setup with and drop an outbound call to test |
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| 14:20.11 | Andrew__M | Q: How do I call an extension that I specify in sip.conf with zero-out = 2345. I made this line up, but load without an error. I want to set up what extension each phone zeroes out to, if possible right in sip.conf for simplicity. Then I want to call that extension in the dialplan. |
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| 14:21.27 | Becca_R | The zip.conf is for device registration. You'll need entries in your extensions.conf to make the extensions ring devices. |
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| 14:23.23 | Andrew__M | Becca_R: Yes, but you tell the system what extension to use for mailbox in sip.conf. So why not also tell what zero-out to use? |
| 14:24.17 | stmaher | Hi guys.. If you have dahdi configured how do i find out what trunk group to use.. |
| 14:24.41 | stmaher | just wanted to test an outbound call with dahdi/g0/2039820398 |
| 14:24.53 | Becca_R | check your /etc/asterisk/dahdi-channels.conf |
| 14:24.58 | Andrew__M | stmaher: Do you have chan_dahdi.conf configured? |
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| 14:28.36 | stmaher | Hi guys |
| 14:28.42 | stmaher | I have chan_dahdi.conf configured |
| 14:28.52 | stmaher | trunkgroup => 1,24 |
| 14:28.57 | stmaher | and spanmap =< 1,1,1 |
| 14:29.04 | Becca_R | Andrew here is a forum about zero-out of voicemail that may be helpful http://forums.digium.com/viewtopic.php?t=74324 |
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| 14:30.27 | Becca_R | stmaher you'll need to dialout group 1 or 24 then and not 0 |
| 14:30.56 | stmaher | Becca_R thanks.. I could be able to do g1 then? |
| 14:31.07 | Becca_R | you should be able to |
| 14:33.31 | stmaher | <PROTECTED> |
| 14:33.31 | stmaher | [Apr 8 15:32:07] NOTICE[2915]: channel.c:5167 __ast_request_and_dial: Unable to request channel Dahdi/1/32809328409238 |
| 14:33.33 | stmaher | [Apr 8 15:32:07] NOTICE[2915]: pbx_spool.c:349 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 14:34.18 | stmaher | my show status is fine |
| 14:34.19 | stmaher | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 14:34.19 | stmaher | Wildcard TE120P Card 0 OK 1 0 0 CAS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) |
| 14:34.23 | Andrew__M | stmaher: You should be able to: Dial(DAHDI/g1/area-code-number) |
| 14:34.55 | stmaher | Im dropping outbound calls via /var/spool/asterisk/outgoing |
| 14:35.19 | Andrew__M | Becca_R: Thanks, I will read through it. |
| 14:35.55 | Andrew__M | stmaher: Any clues in Asterisk CLI when you make those calls? |
| 14:36.07 | stmaher | I have verbose set to 10 |
| 14:36.13 | stmaher | but thats all the output i get.. |
| 14:36.15 | stmaher | call failure |
| 14:36.25 | stmaher | Unable to request channel |
| 14:37.02 | Andrew__M | What line in extensions.conf dials? |
| 14:37.34 | Andrew__M | Can you paste it here? |
| 14:38.11 | stmaher | I dont have anything in extensions.conf yet.. It should just dial out the dahdi interface right? |
| 14:38.33 | Andrew__M | How do you make the call? |
| 14:38.53 | stmaher | drop an outgoing file into /var/spool/asterisk/outgoing |
| 14:39.04 | stmaher | I just need to make sure the E1 is fine.. |
| 14:39.13 | stmaher | so a quick drop call will do :-D |
| 14:39.52 | Andrew__M | I don't know that method. I dial from a set, and use the extension.conf to tell Asterisk how to dial. |
| 14:40.19 | stmaher | I understand.. but it should be the same if I was to use dial(dahdi/g1/98398393) |
| 14:40.39 | Andrew__M | If you say so. |
| 14:42.22 | Becca_R | in your drop file are you building a call to a phone and dialing out or just trying to dial out with only one leg of the call built? |
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| 14:46.46 | stmaher | Becca_R Im just trying to drop a call out to prove a circut works.. Not too worried about content just yet |
| 14:46.59 | stmaher | so the outgoing file will just be a drop call |
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| 14:48.53 | themadchatter | Hey all, quick question. Is there a way to restrict the amount of incoming calls in FreePBX? The boss doesn't want to use the queues I have set up. Thanks in advance to anyone that can lend a hand! |
| 14:49.21 | Andrew__M | 336469 |
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| 14:50.21 | Andrew__M | themadchatter: Do you want to give the callers busy signals? |
| 14:51.26 | themadchatter | Andrew: thats what they want |
| 14:51.26 | Dryanta | or a recording of boss lol |
| 14:52.04 | themadchatter | Hey, we're to busy to take your call right now...lol |
| 14:52.39 | Andrew__M | themadchatter: Just turn off voice mail and restrict the number of calls a line can get on the phone. |
| 14:54.53 | themadchatter | I thought about that, but they want the voice mail active. The problem is, we aren't doing Shared Lines, and multiple people have access to pick up a ringing line. |
| 14:54.58 | Andrew__M | themadchatter: I mean comment out voicemail in extensions.conf. Or you can build a GotoIfTime logic for voice mail after hours. |
| 14:55.35 | themadchatter | I was just hoping there was someway to just restrict incoming channels, similar to the outgoing for the trunk setup... |
| 14:56.42 | Becca_R | stmaher: I'm sorry but I'm not real experienced with the drop files. I know there is a minimum that you have to have and a syntax but it should work. |
| 14:57.02 | Becca_R | I believe the drop files still look to the context in extensions.conf though for routing. |
| 14:58.31 | stmaher | I think i may have found something new.. |
| 14:58.40 | stmaher | I got a million of these in my log file |
| 14:58.41 | stmaher | chan_dahdi.c:2982 my_handle_dchan_exception: PRI got event: Event 32721 (32721) on D-channel of span 1 |
| 15:01.46 | Andrew__M | themadchatter: What connection do you have to the outside world? |
| 15:02.10 | Andrew__M | stmaher: Does dahdi_tool show the e1 as OK? |
| 15:02.49 | Dovid | hello all |
| 15:05.01 | psilikon | <PROTECTED> |
| 15:05.11 | themadchatter | I'm using sip trunks over a cable connection |
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| 15:06.48 | Andrew__M | themadchatter: Then you can either reduce the number of channels over the SIP trunk and the provider will give the callers busy signals, or you can receive them, and give them busy signals from your switch. |
| 15:07.00 | Becca_R | you could always rate limit your SIP in your firewall which would prevent a potential DOS attack as well. |
| 15:08.21 | themadchatter | Makes sense. |
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| 15:10.44 | stmaher | Andrew__M I dont have dahdi_tool |
| 15:10.47 | themadchatter | I wonder if my provider can restrict just one way though. The bosses...plural...will want to have 10 people dial out and only 4-5 dial in at once. The problem is, each of the bosses has a different opinion on how it should work. |
| 15:10.52 | stmaher | Andrew__M everything else but _ tool |
| 15:10.53 | themadchatter | Thanks guys |
| 15:12.19 | stmaher | Andrew__M but dahdi show status shows the trunk is up with no alarms |
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| 15:18.51 | BeeBuu | i found a bug in cdr_mysql addon,what can i do? |
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| 15:21.06 | Becca_R | http://www.asterisk.org/developers/bug-guidelines |
| 15:22.03 | Becca_R | I would suggest searching for the bug you are experiencing in the issue tracker to make sure it hasn't already been identified and worked. |
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| 16:56.24 | m_tadeu | hi...when a call is finished, shouldn't the channel be removed from "sip show channels"? |
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| 16:56.52 | pabelanger | m_tadeu: depends, did you Hangup() the call? |
| 16:57.58 | m_tadeu | pabelanger: the call goes into a queue and after the queue command I have the hangup() command |
| 16:58.39 | pabelanger | m_tadeu: PB you output from 'sip show channels' |
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| 17:00.24 | m_tadeu | pabelanger: here it goes http://pastebin.com/KLEqC48h |
| 17:01.21 | m_tadeu | pabelanger: there's a missing line at the end...."7 active SIP dialogs" |
| 17:01.54 | pabelanger | m_tadeu: what version of Asterisk? |
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| 17:02.24 | m_tadeu | pabelanger: version 1.8.3.2 |
| 17:03.29 | pabelanger | m_tadeu: who is 89.180.229.4? |
| 17:04.13 | m_tadeu | pabelanger: the ip for my testing internet connection |
| 17:05.54 | millsu2 | Is it possible to pass arguments to a macro or gosub through the Queue command? |
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| 17:07.51 | pabelanger | m_tadeu: It looks like some device is sending SIP PUBLISH messages to your asterisk box. If I am reading this properly, it looks like the DEFAULT_PUBLISH_EXPIRES = 3600. |
| 17:07.57 | pabelanger | And I don't see a way to configure it |
| 17:08.32 | pabelanger | If you do another 'sip show peer' in 1 hour, I wonder what happens. If they expire on their own |
| 17:08.57 | kuku | For the first time in 6 years working with asterisk, a new client has it running on a VM machine. The Vmware tools are installed. I moved it to a host with lots of free ram and fast scsi disks. The problem is the voice is still choppy. The choppines is even there when calling from extension to extension ( same location/switch ) so its not the sip TRUNK ( cbeyond ). Any suggestions ? |
| 17:10.09 | m_tadeu | pabelanger: but, that's what's supposed to happen, right? isn0t that the keep alive for the client? |
| 17:11.00 | m_tadeu | kaldemar: you probably don't remember, but thanx for the tip...the router I was using was the problem |
| 17:11.54 | pabelanger | m_tadeu: I believe so. I'be have to read up on what SIP PUBLISH provides. If you enable a sip debugging, do you see active PUBLISH messages? If so, you can likely ignore the channels, because they are actually being used |
| 17:17.23 | m_tadeu | pabelanger: I'll check it in an hour then |
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| 17:57.51 | ariel_ | I have a small issue which I think has a solution, Asterisk VoiceMail notification is sent when a message is left, but if the phone is not on, it does not get it. Some phones are passive and wait for the notification. How can I have asterisk resend the notifications? |
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| 18:13.50 | gruvfunk | Need help with an IVR: What's the best method for keeping an inbound caller on hold within an IVR? (Queue? Park? ConfBridge? other?) The scenario is the following: a person calls in to record their message (say it's: "I need apples, who has apples?",) the PBX records that message and generates outbound calls (to say: Fruit providers) via .call files and puts them into their own context where they are played the original messa |
| 18:13.50 | gruvfunk | ge ("I need apples") and the provider has the opportunity to repond via DTMF (say: 1 for yes I can sell you apples, and 2 for no apples here), and additionally another DTMF data point (say: numerical price per pound, like 33 cents), then I need to present this DTMF data to the original caller who is "waiting" and the next step is to allow the caller to make a selection on vendor (based on say: price). Can anyone point me in the rig |
| 18:13.51 | gruvfunk | ht direction? |
| 18:17.15 | gruvfunk | I currently have the caller going into 1 context, and looking to hold him there waiting to be next steps (I also have the outbound calls to "providers" going into their own context) |
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| 18:20.09 | gruvfunk | I welcome private messages |
| 18:20.31 | ariel_ | gruvfunk: you put them into a parking lot but issue is how to wait long enough for the reponse |
| 18:21.23 | gruvfunk | @ariel: right, it needs to be dynamic where when data is ready to be fed we can pull the caller out of park/queue/conf and feed it |
| 18:22.33 | gruvfunk | @ariel: there could be more than 1 provider answering back "I can sell you apples", and we'd want to tell the caller "you have these options" as they come in |
| 18:24.06 | ariel_ | your going to have to write up a script that does the back end work while it's on hold, and you can send to it the parking lot you put it in. you can have many of them. |
| 18:26.08 | gruvfunk | ariel: so, put the caller in parkinglot, and then process data and play it into that same parkinglot? - can the caller still make DTMF choices while inside the parkinglot? |
| 18:28.08 | ariel_ | kinda transfer the call to the person you want |
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| 18:28.45 | ariel_ | like dynamic queues |
| 18:30.23 | gruvfunk | so when the caller hears a good price on apples, they press 1 and it would transfer or bridge the caller with the apple provider |
| 18:40.23 | gruvfunk | @ariel_ what would you suggest for the back end script for a newbie? (currently have bash shell scripts to generate outbound call files) |
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| 18:42.29 | mikemking | is there any way using realtime sip peers, to make a change to sip_buddies table and have that take effect without pruning the peer? |
| 18:42.43 | mikemking | specifically, changing the accountcode |
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| 18:44.07 | ariel_ | gruvfunk: you need to find someone that is willing to work with you either with a perl script, php or agi |
| 18:44.20 | gruvfunk | good start, thx |
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| 22:02.06 | gruvfunk | How do I force Park to use a specific extension, say CALLERID(num) ? |
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| 23:02.30 | *** join/#asterisk brainiac (~brainiac@208.86.215.38) |
| 23:03.29 | brainiac | Does anyone know how I can set up international calling to work with both 11 & 12 digits? It works with 11 digits, but not with 12. |
| 23:05.11 | gruvfunk | brainiac: you need to adjust your dial plan accordingly |
| 23:05.38 | brainiac | yes, but what adj to make? |
| 23:05.49 | gruvfunk | so, in this case if using Asterisk, likely extensions.conf (unless you have your configurations elsewhere) |
| 23:06.17 | gruvfunk | likely adding a . at the end of your current pattern, but dunno - please share your dial pattern so we can see |
| 23:06.35 | brainiac | ok |
| 23:07.13 | brainiac | all I can do is as follows: |
| 23:07.30 | brainiac | [ Context 'INTERNATIONAL' created by 'pbx_config' ] |
| 23:07.31 | brainiac | <PROTECTED> |
| 23:07.31 | brainiac | <PROTECTED> |
| 23:07.31 | brainiac | <PROTECTED> |
| 23:07.31 | brainiac | <PROTECTED> |
| 23:07.33 | brainiac | <PROTECTED> |
| 23:07.35 | brainiac | <PROTECTED> |
| 23:07.37 | brainiac | <PROTECTED> |
| 23:07.40 | brainiac | <PROTECTED> |
| 23:07.43 | brainiac | <PROTECTED> |
| 23:07.45 | brainiac | <PROTECTED> |
| 23:07.48 | brainiac | <PROTECTED> |
| 23:07.50 | brainiac | <PROTECTED> |
| 23:07.53 | brainiac | <PROTECTED> |
| 23:08.40 | brainiac | I think the offender is line 2 |
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| 23:13.53 | gruvfunk | brainiac: yeah _011X. is what I was looking for.. this is over my head - not sure what is happening in lines 2-6 |
| 23:14.07 | gruvfunk | but I think you're on the right track.. |
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| 23:20.29 | *** join/#asterisk Jouva (Jouva@fluffy.moufette.com) |
| 23:22.45 | Jouva | I'm aware that this might be something that I need to do on my VoIP provider's side, but I'm wondering about how to set proper outbound caller ID info. I see the phone number, but I'm also looking to make the name work. I tried Set(CALLERID(name)=Name here) in the dialplan but that didn't seem to work |
| 23:25.44 | gruvfunk | Jouva: try "quotes" around "Name Here" |
| 23:27.10 | gruvfunk | Set(CALLERID(all)="Joe Smith" <2095551212>) |
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| 23:27.29 | gruvfunk | but.. all dependent on your provider allowing you to set your own values |
| 23:28.13 | Jouva | Well according to the settings on their site I can atleast set the number |
| 23:28.46 | gruvfunk | Set(CALLERID(num)=2095551214) |
| 23:28.50 | Jouva | right |
| 23:29.12 | gruvfunk | otherwise use (all) with shown parameters |
| 23:29.24 | gruvfunk | or Set(CALLERID(name)="Name Here") |
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| 23:30.23 | Jouva | yeah |
| 23:30.35 | Jouva | I just tried (all) to no avail |
| 23:30.48 | Jouva | I originally did (num) and (name) but just added quotes |
| 23:30.54 | Jouva | And same results. "Unknown name" |
| 23:34.03 | Jouva | I know there's also ANI stuff so I don't know if that also needs to be set or not |
| 23:34.51 | Jouva | Cause I never used them but I know the callerid function has those as potential items to set now |
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| 23:37.38 | gruvfunk | "Unknown Name" to me means the provider is not allowing you to set it, or a CID looking is taking place somewhere along the line and pulling that data |
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| 23:40.49 | Jouva | I'm talking to their support now |
| 23:42.49 | Jouva | Ahhh yeah they have to set this up themselves |
| 23:45.21 | gruvfunk | :) |
| 23:49.26 | gruvfunk | Anyone tell me how to set Park to a specific value (calleridnum) so I can ParkedCalls(that value) later? |
| 23:49.38 | gruvfunk | I've disabled the parkinglot in features.conf |
| 23:50.46 | gruvfunk | ? Set(CHANNEL(parkinglot)="value") ? |
| 23:51.36 | gruvfunk | ? Set(CHANNEL(parkinglot)=${CALLERID(num)}) ? |
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