| 00:04.04 | newtonr | ChannelZ: yeah Drupal. What 'W' ? |
| 00:04.25 | newtonr | wtf, thats new |
| 00:04.29 | newtonr | I'll see what changed |
| 00:04.40 | ChannelZ | :) |
| 00:04.48 | newtonr | Webdev team must have ran an update of some kind :D |
| 00:04.53 | ChannelZ | Maybe it's supposed to be a symbol or something from a different font |
| 00:05.04 | ChannelZ | I don't really remember what it looked like before |
| 00:05.13 | newtonr | https://www.asterisk.org/downloads/asterisk/all-asterisk-versions |
| 00:05.18 | newtonr | yeah those still have the little download icon |
| 00:05.31 | newtonr | but on this page http://www.asterisk.org/downloads they have the W |
| 00:05.32 | ChannelZ | Ahh. |
| 00:05.57 | newtonr | if you ever see something goofy like that, you can always file it at http://www.asterisk.org/contact , but i'm filing it right now internally |
| 00:06.41 | ChannelZ | Ok thanks. |
| 00:10.44 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 00:12.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 00:12.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 00:23.59 | rue_work | anyone good with mgcp.conf? |
| 00:25.43 | *** kick/#asterisk [smkelly!~file@asterisk/developer-and-muffin-lover/file] by file (smkelly) |
| 00:25.53 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 00:25.57 | smkelly | I saw that coming. |
| 00:26.06 | smkelly | I was even over on this tab waiting for it. |
| 00:26.23 | file | I bet. |
| 00:28.15 | smkelly | <3 |
| 00:30.58 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 00:31.39 | rue_work | I have two mgcp gateways configured, but nomatter what I do with the config, only one works |
| 00:33.47 | sawgood | If the Asterisk CLI shows -- Started music on hold, class 'default', on SIP/Asterisk-Services1-00000027 (but nothing is heard on the phone put on hold) What is a good place to start troubleshooting? |
| 00:34.02 | sawgood | Asterisk 1.8.20.1 under CentOS 5.5 or 6.3 |
| 00:34.25 | sawgood | normal calls have two way audio |
| 00:37.06 | ChannelZ | networking |
| 00:37.36 | ChannelZ | or possibly the phone |
| 00:37.41 | ChannelZ | or possibly your config |
| 00:37.56 | ChannelZ | Sunspots even. |
| 00:38.12 | sawgood | I'll go with sunspots (but I'll double check musiconhold.conf) |
| 00:38.14 | ChannelZ | If you make an extension that just calls MusicOnHold() do you get musical stylings? |
| 00:38.28 | ChannelZ | (you might have to Answer first that I think of it..) |
| 00:38.40 | sawgood | I put those two lines in extensions.conf |
| 00:38.47 | dpilon | which 2 ? |
| 00:39.02 | sawgood | answer() and then play MOH |
| 00:39.07 | ChannelZ | yes |
| 00:39.13 | ChannelZ | in proper syntax of course |
| 00:39.21 | sawgood | :> |
| 00:39.29 | [TK]D-Fender | Yeah, somewhere FreePBX can get to them .... |
| 00:39.42 | ChannelZ | farts |
| 00:39.54 | sawgood | no more FreePBX |
| 00:40.22 | sawgood | If FreePBX was just a 'GUI' for Asterisk, but those scripts I'm done dealing wiht |
| 00:40.26 | sawgood | no more I give up |
| 00:40.43 | ChannelZ | Yay! |
| 00:41.00 | dpilon | he is going to the dark side |
| 00:42.06 | dpilon | sawgood: why don't you put up the MOH config so people can look at them |
| 00:42.26 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 00:42.58 | sawgood | well, it is the default file with not a single change |
| 00:43.03 | sawgood | from samples |
| 00:43.28 | dpilon | even the extensions.conf |
| 00:43.47 | sawgood | no I have an exensions.conf working |
| 00:43.54 | sawgood | but a good question |
| 00:45.32 | [TK]D-Fender | sawgood: If you didn't even configure your own MoH or look at it... don't ask why it's not working. |
| 00:45.52 | [TK]D-Fender | sawgood: Go prove it's configured and that the things it looks at are even there. |
| 00:46.00 | ChannelZ | It's Magical. Like the iPad. |
| 00:46.15 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 00:46.23 | dpilon | the mini is magical |
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| 00:49.53 | phix | All Apple products are shit, the end |
| 00:52.43 | ChannelZ | But they are magical shit. |
| 01:00.19 | sawgood | If I dial 6000 now, I hear MOH (thank you for the 101 .conf file reminders) ... so that is working now |
| 01:01.04 | ChannelZ | ok so at least that works, assuming it's the same MOH context (default) |
| 01:06.19 | sawgood | When I hit the hold button on the phone (the called party does not hear MOH) ... If I dial x6000 I hear MOH (so, how do I make it so hold = 6000) |
| 01:07.18 | WIMPy | Is the call really being put on hold? |
| 01:07.20 | sawgood | http://pastebin.com/3WLtufq7 = extensions.conf |
| 01:07.54 | [TK]D-Fender | sawgood: Where are the 2 calls to compare? |
| 01:08.01 | sawgood | http://pastebin.com/7W7J2Nq9 = musiconhold.conf |
| 01:09.01 | sawgood | If I make an outbound call, and hit the HOLD button (normally by default) MOH plays (with scripting) ... I have a bare-bones system with one phone and Asterisk 1.8.20.1 nearly factory defaulted |
| 01:09.07 | [TK]D-Fender | sawgood: Where are the calls? |
| 01:09.19 | sawgood | coming in 10 seconds |
| 01:10.16 | *** part/#asterisk janmate (~janmate@89.173.160.127) |
| 01:11.24 | ChannelZ | if the device has an MOH server configured specifically it might be going off on its own to possibly nowhere. |
| 01:11.34 | sawgood | http://pastebin.com/F26sUtaN = Asterisk CLI with outbound call from x404 to a cell phone |
| 01:12.20 | sawgood | after the cell phone answered, I pressed the hold button on the phone (waited 5 seconds) and could hear nothing on the cell phone (and then) I pressed the hold button again on x404 to speak on the phone |
| 01:12.40 | [TK]D-Fender | sawgood: I said BOTH. |
| 01:12.42 | [TK]D-Fender | sawTWICE |
| 01:12.56 | sawgood | which other call |
| 01:13.12 | [TK]D-Fender | one that works and one that doesn't |
| 01:13.20 | sawgood | Oh.. I can call x6000 and hear the MOH (coming in 10 seconds) |
| 01:14.01 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:14.13 | sawgood | http://pastebin.com/Bcf3AzBy = me calling from x404 to x6000 and hearing MOH play |
| 01:14.44 | [TK]D-Fender | sawgood: enable SIP debug. Try the call that failed again the same way |
| 01:14.50 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 01:18.17 | sawgood | http://pastebin.com/Xkk6Ti6m = sip set debug on (call out from x404 over a SIP trunk to a cell phone) answered on the cell phone (then) hold button pressed on x404 (nothing heard on cell phone) (long pastebin) |
| 01:21.47 | *** join/#asterisk dozment (~dozment@173-165-163-145-atlanta.hfc.comcastbusiness.net) |
| 01:22.05 | [TK]D-Fender | sawgood: wait a little longer on another test. So far it seems to check out... |
| 01:22.59 | sawgood | at one time, the above process (pushing HOLD) on the x404 phone (did play) MOH on the cell phone (it stopped working) after I removed the exten => 6000 lines in extensions.conf |
| 01:23.31 | sawgood | I put them back in extensions.conf (did a reboot of the machine) took a break, and it stopped working now |
| 01:24.25 | [TK]D-Fender | check your networking, WAN IP's, etc |
| 01:24.32 | [TK]D-Fender | Things should not jsut break like that |
| 01:24.41 | [TK]D-Fender | And nothing in extensions.conf should have that kind of impact |
| 01:25.03 | sawgood | I hit hold, hear nothing on the cell phone, took it off hold from x404 (spoke again to the cell phone), and then hit HOLD again (this 2nd time I hear MOH) |
| 01:26.33 | sawgood | hold does not seem to work until it is done twice (2nd time forward) MOH works (funny( |
| 01:26.53 | [TK]D-Fender | Sounds like a reinvite / networking issue |
| 01:27.19 | sawgood | ha! |
| 01:27.23 | sawgood | cool .. thanks! |
| 01:27.35 | sawgood | that was fun! |
| 01:28.30 | sawgood | what is a phone does not have a hold button (how does one put something in extensions.conf to work like the HOLD button = for example 444 = hold 445 = unhold |
| 01:29.01 | [TK]D-Fender | sawgood: PARKING <- |
| 01:29.15 | sawgood | on it ... reading now ... thank you |
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| 01:39.24 | sawgood | So, parking is working (if you press the transfer key and then dial 700) (parkedcall), but what if the phone has neither a transfer or hold button (only a key pad) |
| 01:39.39 | sawgood | I tried using ##700 or ##700# |
| 01:39.55 | [TK]D-Fender | What phone? |
| 01:40.12 | sawgood | in this case, a cell phone acting as a remote extension for example |
| 01:40.38 | [TK]D-Fender | Now rephrase that in terms of something incredible specific that is actually talking to asterisk |
| 01:41.27 | sawgood | Ok .. using DISA from a cell phone, how do I place a caller on hold from the cell phone (after the DISA call is establihsed) and then take the call back from hold (using a standard cell phone) |
| 01:43.53 | [TK]D-Fender | sawgood: again far from a complete picture |
| 01:45.04 | sawgood | The description or the solution |
| 01:45.51 | [TK]D-Fender | You don't HAVE the solution.... you are the one asking the question... |
| 01:47.11 | sawgood | Well, here is my goal = using a cell phone (pass-through with DISA and Asterisk) ... one the call is established to a another phone (analog line), how does the 'cell phone' user put the call on hold (so Asterisk plays MOH) |
| 01:47.40 | sawgood | what can the cell phone user press on the cell phone (to make Asterisk) know HOLD on is requested? |
| 01:49.59 | [TK]D-Fender | "Cell-phone" doesn't magically reach DISA. You don't seem to understand the definition of the word "complete" |
| 01:50.18 | [TK]D-Fender | Now draw a SOLID LINE of this call |
| 01:50.24 | sawgood | ok |
| 01:52.35 | igcewieling | sawgood: see features.conf |
| 02:05.54 | *** join/#asterisk mjordan (~mjordan@216.186.152.216) |
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| 02:20.17 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:21.27 | sawgood | In FreePBX (there is the ablity to put in a T,t for dial commands) allowing one to transfer a call ... any idea where this T,t goes in extensions.conf |
| 02:21.36 | sawgood | I have a need to allow transfers from any phone ... |
| 02:24.48 | *** part/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 02:25.16 | sawgood | found it ... sorry! |
| 02:35.45 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
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| 02:54.08 | volga629 | Why asterisk one transmit nat and no nat https://fpaste.networklab.ca/ngu8/ ? |
| 02:54.20 | volga629 | What possible cause |
| 03:05.24 | [TK]D-Fender | volga629: Show complete SIP debug. |
| 03:27.53 | volga629 | https://fpaste.networklab.ca/SEOX/ |
| 03:42.49 | [TK]D-Fender | volga629: that is 2 * boxes QUALIFY ing to one another. |
| 03:43.18 | [TK]D-Fender | volga629: The inbound is not matched against a peer, it is simply responded to so your [general] settings are what apply to the communication |
| 03:43.39 | [TK]D-Fender | volga629: Then when that box Qualify's to the other it is SENDING and therefor is using the settings of the peer |
| 03:47.38 | volga629 | hmmm I se, yes this local trunk between two asterisk boxes in different geo locations |
| 03:51.57 | volga629 | so that general NAT setting both ends ? (I am trying find source previous issue) |
| 03:52.36 | volga629 | is on thunk allowed set nat=no ? |
| 03:56.09 | volga629 | I corrected and I see NoNat both end which I see the rigth way |
| 03:56.22 | volga629 | think |
| 04:01.25 | volga629 | [TK]D-Fender Thank you for you help, I will test tomorrow might I will closer to track down my original issue |
| 04:12.30 | *** join/#asterisk darkdrgn2k (~DarkPhoni@69-165-131-20.dsl.teksavvy.com) |
| 04:12.31 | darkdrgn2k | Hey guys |
| 04:12.38 | darkdrgn2k | is there a way to map a phone number to a CLEC? |
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| 05:02.49 | Minotaur01 | o/ hello all dose anyone know if there is a irc channel for digium phones? |
| 05:03.51 | *** part/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
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| 05:10.09 | [TK]D-Fender | Minotaur01: There isn't. This is the closest there is. |
| 05:16.58 | Minotaur01 | Okay, dose anyone know how to make the D70 sort the contact list |
| 05:17.18 | Minotaur01 | The contacts seem to stay in the order there put in |
| 05:18.43 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 05:26.00 | [TK]D-Fender | Why should it sort it? You choose your won order. |
| 05:26.06 | [TK]D-Fender | That is removeing a choice from you... |
| 05:34.35 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 05:42.35 | ChannelZ | I think the choice he wants is to alphabetize it. |
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| 06:26.51 | fling | hello! |
| 06:27.05 | fling | I'm connecting two asterisk servers over iax2 |
| 06:27.13 | fling | don't I need to use register or switch? |
| 06:27.35 | fling | I use static ip addresses |
| 06:29.20 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
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| 06:35.06 | rue_house | can anyone help me with a mgcp problem? my second gateway wont work... |
| 06:36.28 | *** join/#asterisk bandroidx (~bandroidx@2605:6400:2:fed5:4:0:414:11) |
| 06:41.01 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
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| 06:44.18 | kaldemar | fling: only if you want to. |
| 06:44.48 | fling | kaldemar: ok |
| 06:45.56 | fling | I can't connect to two asterisk servers over sip with ekiga |
| 06:46.07 | fling | auth failed on second one all the time |
| 06:46.47 | kaldemar | so ekiga->asterisk->asterisk and auth failure on the second asterisk? |
| 06:47.36 | kaldemar | what failure are you seeing and what does your configuration look like? |
| 06:47.38 | fling | kaldemar: no, [asterisk] <-sip- [ekiga] -sip-> [asterisk] |
| 06:48.45 | kaldemar | is that one or two calls? |
| 06:49.09 | fling | kaldemar: single client box with ekiga |
| 06:49.21 | fling | kaldemar: two separate asterisk servers |
| 06:49.30 | fling | I'm just registering with ekiga |
| 06:49.43 | fling | so on second server I always have this > [Jan 30 13:48:17] NOTICE[22155]: chan_sip.c:27726 handle_request_register: Registration from '<sip:000@pbx>' failed for '10.0.2.2:5060' - Wrong password |
| 06:49.58 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 06:50.35 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
| 06:50.45 | fling | kaldemar: if I connect this one first, I have same error on another one :p |
| 06:51.39 | *** join/#asterisk razorgank (~quassel@203.160.180.2) |
| 06:51.46 | kaldemar | oh? wouldn't be surprised if ekiga was to blame. |
| 06:52.13 | fling | kaldemar: right, I think something is wrong in ekiga |
| 06:52.42 | kaldemar | but look at sip debugs for each case, you'll see at least the user ekiga is trying to authenticate as. |
| 06:59.41 | fling | kaldemar: do I need '^Authorization: ' line? |
| 06:59.44 | *** join/#asterisk iztech (~iztech@76-246-226-131.uvs.irvnca.sbcglobal.net) |
| 07:00.55 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:01.06 | kaldemar | fling: that's the header that is used for authentication. |
| 07:02.36 | fling | kaldemar: Authorization: … algorithm=MD5, response="xxx" |
| 07:02.42 | fling | kaldemar: so the hash do not match |
| 07:02.54 | ChannelZ | still doesn't even get what you're doing |
| 07:03.23 | fling | ChannelZ: hey :p just registering on two separate asterisk servers with ekiga |
| 07:04.02 | fling | ChannelZ: and ekiga always sends wrong pass to the second server |
| 07:04.07 | ChannelZ | ekiga the client? |
| 07:04.13 | fling | yes |
| 07:04.38 | ChannelZ | "sends the wrong password"? Or perhaps your second server is misconfigured and not matching the peer you think it's supposed to be? |
| 07:05.27 | fling | ChannelZ: if I connect server A first, I have auth fail on server B, If I connect server B first, I have auth fail on A :P |
| 07:06.40 | ChannelZ | Interesting. Busted-ass client. |
| 07:06.57 | fling | or maybe version 3.2.7 is too old |
| 07:09.08 | kaldemar | fling: you might have the wrong peer matching. if you want someone to really help you with that, pastebin a sip debug for the whole failed registration. |
| 07:14.14 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:15.51 | *** join/#asterisk din3sh (~din3sh@41.136.87.190) |
| 07:16.09 | din3sh | hello all |
| 07:16.23 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 07:16.25 | fling | din3sh: hey |
| 07:17.48 | ChannelZ | pants-off dance-off! |
| 07:18.00 | fling | \o\ |
| 07:18.07 | fling | /o/ |
| 07:18.14 | fling | \o/ |
| 07:18.40 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:18.52 | ChannelZ | <('.'<) |
| 07:19.03 | ChannelZ | (>'o')> |
| 07:19.14 | fling | [Jan 30 14:18:45] NOTICE[11311]: chan_sip.c:23248 handle_response_peerpoke: Peer '740' is now Reachable. (24638ms / 50000ms) |
| 07:21.29 | ChannelZ | That's a lousy network. |
| 07:22.11 | fling | ChannelZ: 1.2kbps 3g, can't call there because of rtp overhead :\ |
| 07:22.31 | ChannelZ | ~25 seconds. Faster to talk to the moon. |
| 07:24.30 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 07:24.47 | fling | how to reload iax? |
| 07:25.11 | ChannelZ | take a wild guess |
| 07:25.46 | gavimobile | when I get one of these in my log "chan_sip.c: Sending fake auth rejection for device 101<sip:101@my_servers_ip>;tag=074d250c..." is there any way I can find out the ip that tried to make the connection? |
| 07:26.35 | ChannelZ | Hack the source. I did this for a couple of messages to trap in fail2ban. I keep forgetting to submit a patch. |
| 07:27.24 | gavimobile | fling: in the definitive guide I do not see anything like "iax reload". it does however use this method chan_sip.c: Sending fake auth rejection for device 101<sip:101@81.218.196.30>;tag=074d250c |
| 07:27.26 | gavimobile | oops |
| 07:27.31 | gavimobile | module reload chan_sip.so |
| 07:27.37 | gavimobile | module reload chan_iax2.so |
| 07:27.45 | fling | gavimobile: thanks |
| 07:27.46 | ChannelZ | iax2 reload |
| 07:27.58 | ChannelZ | rocket science it is |
| 07:27.59 | fling | ChannelZ: no such command :p |
| 07:28.11 | fling | ChannelZ: I probably do not have the module loaded |
| 07:28.14 | gavimobile | ChannelZ: I get the same thing, but I don't have an iax.conf |
| 07:28.27 | gavimobile | fling: did you add the 2 like ChannelZ said? |
| 07:28.31 | ChannelZ | then "reload" is a relative term I guess. |
| 07:28.56 | fling | ChannelZ: no |
| 07:28.58 | gavimobile | ChannelZ: can you send me a link on how to hack the source? |
| 07:29.12 | kaldemar | all the iax2 commands come from the chan_iax2 module. if it is not loaded, the commands are not there. |
| 07:29.15 | fling | module needs to be loaded even before `module reload chan_iax2.so` |
| 07:29.20 | fling | it works now, thanks |
| 07:30.45 | ChannelZ | gavimobile: here's my own patch: http://burner.com/asterisk-fakerej.patch |
| 07:32.15 | ChannelZ | cd your-asterisk-source-dir; patch -p0 < /path/to/asterisk-fakerej.patch |
| 07:32.37 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
| 07:33.14 | gavimobile | ChannelZ: thaks |
| 07:33.16 | gavimobile | thanks |
| 07:34.22 | ChannelZ | sure |
| 07:36.07 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:57.40 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
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| 08:03.18 | gavimobile | ChannelZ: is my source directory the root for the directory which contains the asterisk make files? |
| 08:03.41 | gavimobile | in my case /usr/src/downloads/asterisk/1.8/ |
| 08:04.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:06.19 | ChannelZ | yes |
| 08:09.01 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 08:15.34 | gavimobile | sweet, I ended up using the patch from this ticket instead https://issues.asterisk.org/jira/browse/ASTERISK-19348?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel |
| 08:20.07 | ChannelZ | whatever turns you on |
| 08:20.31 | *** join/#asterisk PbxMan (c335d968@gateway/web/freenode/ip.195.53.217.104) |
| 08:20.51 | PbxMan | good morning |
| 08:25.15 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:8064:c60a:b7ad:204f) |
| 08:26.32 | ChannelZ | aloha |
| 08:33.39 | fling | PbxMan: hey |
| 08:34.10 | fling | ChannelZ: what 'peercontext=foo' does? |
| 08:37.58 | ChannelZ | where? |
| 08:43.44 | kaldemar | fling: it allows you to request the used dialplan context on the other side. |
| 08:45.39 | fling | hmm hmm |
| 08:45.50 | fling | kaldemar: but why? umm I need to read about it |
| 08:48.33 | din3sh | i havent been able to sort out the stupid dtmf problem |
| 08:49.25 | din3sh | dtmf not recognized on calls from south africa on E1 line.... |
| 08:50.28 | din3sh | <ChannelZ>:http://burner.com/asterisk-fakerej.patch----->which * version? |
| 08:51.12 | ChannelZ | din3sh: should be pretty agnostic. I think I originally made it under 10.x but it still applies to 11.x |
| 08:51.50 | din3sh | would work on 1.8? |
| 08:52.02 | ChannelZ | dunno. let me see if I have a source archive lying around already |
| 08:54.14 | ChannelZ | looks like no, chan_sip.c changed a bit. |
| 08:54.58 | ChannelZ | But you can look at the patch and figure it out and apply your own changes to chan_sip.c - it's not rocket science, just adding an extra parameter in the ast_log lines |
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| 08:57.57 | din3sh | oki |
| 08:58.24 | din3sh | thnks |
| 08:58.38 | din3sh | you've ever used CallCompletionRequest? |
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| 09:02.16 | ChannelZ | no |
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| 09:17.15 | din3sh | how to implement callback in *? |
| 09:17.42 | ChannelZ | Upgrade? :) |
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| 09:29.00 | TazzNZ | din3sh: still having issues with dtmf ? |
| 09:30.13 | TazzNZ | reads more history - guess so |
| 09:37.23 | din3sh | TazzNZ: yes :s |
| 09:38.59 | ChannelZ | Have you recorded a call and listened to the DTMF? there's not a lot you can do about it if it's coming to you already chewed up |
| 09:40.16 | din3sh | dahdi monitor? |
| 09:40.35 | din3sh | havnt tried that one yet |
| 09:40.44 | ChannelZ | yes |
| 09:46.07 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 09:46.48 | danfromuk | Hi. When using static sip.conf files, does 'sip reload' remove all registrations like it does when realtime is used? |
| 09:46.51 | din3sh | will that enable me to confirm if the recieved audio/dtmf is of poor quality? |
| 09:50.38 | kaldemar | danfromuk: no. |
| 09:51.10 | kaldemar | danfromuk: wait, do you mean inbound or outbound? |
| 09:51.11 | gavimobile | ChannelZ: was your patch for 1.8? |
| 09:51.41 | din3sh | gavimobile: i guess not, he said it was for 10 , 11 |
| 09:51.49 | gavimobile | din3sh: didn't see that |
| 09:52.21 | gavimobile | where did he say that? |
| 09:53.31 | gavimobile | got it |
| 09:54.17 | danfromuk | kaldemar: Inbound. Peers that are registered on the asterisk server. Not outgoing registrations that. |
| 09:54.31 | danfromuk | that i would make calls through |
| 09:54.37 | kaldemar | danfromuk: they remain. |
| 09:55.04 | kaldemar | outbound registrations are not for making calls. |
| 09:56.13 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 09:57.51 | danfromuk | kaldemar: i'm a little confused what you mean by outbound and inbound registrations. By outbound, do you mean a statement like register => ? |
| 10:00.34 | kaldemar | danfromuk: yes. when asterisk registers somewhere, it is outbound. when some device registers to asterisk, it is inbound. |
| 10:01.52 | danfromuk | Thats what i thought you meant. |
| 10:01.53 | danfromuk | Ok thanks |
| 10:01.56 | kaldemar | danfromuk: you're probably confused by what registration is. an outbound registration is not like a login that is used for outbound calls. instead, it is a way to let the other end know where you are. |
| 10:02.55 | *** join/#asterisk TazzNZ (~pieter@nz.insync.za.net) |
| 10:02.58 | danfromuk | Yes, i realised that when you said o.b. is not for making calls |
| 10:04.56 | danfromuk | I'm switching from RT to statis sip.conf files. We have two asterisk servers using one realtime DB, and it seems for some reason, the 2nd server is picking up registrations from the main server (i think through the DB) and sending out keepalive packets. In some cases, the endpoints are actually being tricked into switching to the 2nd server. |
| 10:05.10 | danfromuk | static* |
| 10:07.54 | danfromuk | incoming registrations |
| 10:37.20 | *** join/#asterisk smirker (Smirker@203-206-242-36.dyn.iinet.net.au) |
| 10:43.55 | fling | kaldemar: Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT |
| 10:44.04 | fling | CAUSE : No authority found \n CAUSE CODE : 50 |
| 10:44.11 | fling | kaldemar: my peers are not authorized? |
| 10:45.33 | fling | http://bpaste.net/show/73832/ http://bpaste.net/show/73833/ |
| 10:53.30 | fling | what am I doing wrong? |
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| 11:04.24 | kaldemar | fling: where are you seeing that and when? |
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| 11:25.43 | npx | What's the best bet for tight integration between Asterisk and a J2EE application? JAsterisk? By "tight" I mean that when I add a user I want them to have an extension, when they make a call I want to record it. etc etc |
| 11:28.28 | kaldemar | npx: what do you want the java application to do? handle adding users? handle triggering recording on a channel? |
| 11:28.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:30.25 | npx | kaldemar: both, and possibly (forgive me if this is silly, I'm still exploring this) route calls to open extensions |
| 11:32.41 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:35.06 | kaldemar | npx: then you need something that can modify configurations and use AMI. jasterisk seems quite dead, see asterisk-java. |
| 11:39.06 | fling | kaldemar: in iax2 debug on the second box when I dial it from the first box |
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| 11:55.21 | cusco | hi folks |
| 11:55.37 | cusco | when using features.conf att xfer |
| 11:55.45 | cusco | I get: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x4 (ulaw)(4)/0x8 (alaw)(8) |
| 11:56.07 | cusco | how can I fix this? |
| 12:04.34 | cusco | channel is: NativeFormats: 0x8 (alaw) |
| 12:04.34 | cusco | <PROTECTED> |
| 12:04.34 | cusco | <PROTECTED> |
| 12:06.46 | kaldemar | fling: which one is first and which one is second? how are you dialing in your dialplan? |
| 12:07.13 | cusco | I thought I had everyone and everything using alaw |
| 12:07.22 | cusco | I don't understand where the ulaw comes from |
| 12:07.45 | cusco | ok, call comes from sip peer where only allowed format is alaw |
| 12:08.08 | cusco | plays back some audiofile swith Playback() those files are alaw, (file.al) .. |
| 12:08.14 | cusco | and goes into Queue |
| 12:08.31 | cusco | member that picks up the call has both formats allowed, but alaw in first place |
| 12:09.03 | cusco | then he uses features.conf to transfer the call, and I'm not sure yet, but I'm guessing that this call (while atended) is in ulaw ?! |
| 12:09.09 | cusco | im going to check this now |
| 12:11.13 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 12:13.10 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 12:34.35 | *** join/#asterisk Diffen (~diffen@c-7476e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 12:36.45 | Diffen | Evning. Im trying to connect two asterisk via sip. I wont one that acts like a pstn operator and the other one is normal. I have the trunk registered from the second to the first one but when im trying to call out i get "SIP/2.0 407 Proxy Authentication Required" from the first one on the invite. Where in hell can i read about how to setup my second asterisk (1.8) so it does send the correct authentication information |
| 12:36.46 | Diffen | <PROTECTED> |
| 12:43.19 | gavimobile | Diffen: I believe this is discussed in the definitive guide |
| 12:43.31 | kaldemar | Diffen: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291281 |
| 12:44.13 | gavimobile | see page 148 |
| 12:44.27 | gavimobile | or use the link that kaldemar gave ya |
| 12:44.43 | gavimobile | but that does sound like a fun project to do :-) |
| 12:45.50 | Diffen | ok brb then :) thanks for the links :) |
| 12:47.54 | *** join/#asterisk davlefouAMD (~david@197.15.73.137) |
| 12:48.24 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 12:48.36 | bombev | hii guys :) |
| 12:50.53 | *** join/#asterisk racho (~racho@46.40.123.204) |
| 12:53.25 | racho | i have an agi script in the beginning of an extension. it checks some numbers and if they please it, it makes 'Goto', 'context,extension,priority'. However the rest of the step in the extension are still executed. is there a way to prevent that? |
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| 12:56.13 | Diffen | MOTHER OF ALL… haha CRAP |
| 12:56.14 | Diffen | :) |
| 12:56.28 | Diffen | I have been using type=friend on the "pstn" asterisk hahaha |
| 12:56.41 | Diffen | not good! well thanks gavimobile and kaldemar :) |
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| 13:14.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 13:28.08 | *** join/#asterisk nacho2k (~Thunderbi@r190-64-14-98.ir-static.anteldata.net.uy) |
| 13:29.30 | nacho2k | hi guys, I am looking for add ground lines to my asterisk, I was looking that Cisco SPA400 has 4 FXO ports. Is this compatible for using with asterisk? |
| 13:38.31 | *** join/#asterisk piffey (~pfe@gateway/tor-sasl/piffey) |
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| 13:47.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:54.01 | igcewieling | nacho2k: Do you mean ground start signaled lines? |
| 13:55.25 | *** join/#asterisk serafie (~erin@nat/digium/x-mzawzeyvpnlziycm) |
| 13:55.36 | igcewieling | racho: The goto should work, but I never do anything like that from an AGI. I normally use my AGIs to set dialplan variables and then use the diaplan to do the goto or Dial |
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| 14:05.26 | racho | igcewieling, the problem is I want to evade the mix monitor for certain caller ids. however the same caller ids are quite a lot. so i dumped a fast script that works and it skips n steps to where i want it to go. but then asterisk "rewinds" itself up the extension and starts mix monitoring and all the other stuff |
| 14:06.15 | *** join/#asterisk James87 (~wiv@87.213.232.93) |
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| 14:23.53 | piffey | We have a series of ring groups we run through before hitting our own call group -- all mobile numbers with confirm calls. All users except for two are able to rpess 1 to confirm calls with no problem. The two with issues both use BlackBerrys and intermittently (1 in 50 calls) pressing 1 actually works. This followed us from 1.6 to 10. Since this seems like a BB issue, was thinking it had to be something |
| 14:23.56 | piffey | with DTMF tones, maybe a threshold I can lower for how long the tone has to be or something? Only my 3rd day using asterisk/doing voip so thinking I must just be searching wrong since I'm finding no one else with this problem. Anyone heard of it before or can point me in a proper direction to start looking for a solution? Verbose debugging shows no anomalies between the BBs and other phones on the Asterisk |
| 14:23.57 | piffey | end, also making me think it's BB specific. |
| 14:24.31 | piffey | Sorry for the long text, been working on this the last three days so wanted to lay it all out since I really have gone through the gambit on this one. :) |
| 14:31.51 | [TK]D-Fender | piffey, You haven't described how you are calling the BB's exactly |
| 14:33.51 | piffey | Calling into the phone server, it rings through the groups, then hits the on call group if no one in support is available and the server calls from there. |
| 14:34.15 | [TK]D-Fender | piffey, Lets try this again. |
| 14:34.18 | piffey | Or do you mean types of lines? We have analog lines and a TDM400P, the BBs are on vzw. |
| 14:34.38 | [TK]D-Fender | piffey, your BB's are plastic an metal phones. How precisely is Asterisk calling them? |
| 14:35.31 | [TK]D-Fender | piffey, you have 1 option : relaxdtmf=yes |
| 14:35.37 | [TK]D-Fender | piffey, this is for your chan_dahdi.conf |
| 14:36.06 | [TK]D-Fender | piffey, it will loosen up the constraints on detection. It might help and is just about the only thing you can do short of a change on the BB side |
| 14:37.46 | piffey | Ahhhh okay. I'll try adding that, I don't see it in there now. As for the how precisely it's calling, I'm not sure I'm following you there. I'd assume Asterisk uses one of the three other lines to place a call to the BlackBerrys, then bridges those two lines on confirmation? |
| 14:38.33 | [TK]D-Fender | piffey, You're calling out your TDM400P's analog lines. That is the answer I was looking for. |
| 14:38.58 | [TK]D-Fender | piffey, For all I know * was using a SIP ITSP and DTMF is OOB to them and we have no control. Or you were calling a SIP client directly on the phone |
| 14:39.14 | [TK]D-Fender | piffey, Or you had a GSM card in your server. |
| 14:39.27 | [TK]D-Fender | piffey, See your "connection" could have meant almost anything |
| 14:39.46 | [TK]D-Fender | piffey, and not knowing doesn't give us options to tell you |
| 14:40.01 | piffey | Ooooh, got it. Sorry, new to the whole phone end of things. I can /definitely/ see how that would make a big difference. |
| 14:40.17 | [TK]D-Fender | piffey, Only night & day really.... |
| 14:41.02 | [TK]D-Fender | piffey, so make your change, restart * completely and test |
| 14:41.52 | piffey | And does that just go under the [channels] section in chan_dahdi.conf? |
| 14:43.55 | Katty | hi lads. |
| 14:44.34 | [TK]D-Fender | piffey, yes |
| 14:44.57 | bulkorok | asterisk segfaulting at receivefax... damn |
| 14:46.53 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:46.53 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:47.44 | Katty | Qwell: ping |
| 14:49.18 | Kobaz | P-Asserted-Identity: "Foo Bar" <sip:invalid.unknown.domain |
| 14:49.22 | nacho2k | igcewieling: yes, I mean ground start signaled lines.. |
| 14:49.29 | Kobaz | so I'm geting that from an avaya v6 |
| 14:49.37 | Kobaz | any idea how to get the avaya to send callerid number? |
| 14:50.11 | igcewieling | nacho2k: You would have to contact Cisco/Linksys to find out of the SPA supports groundstart lines. I doubt they do. |
| 14:50.44 | igcewieling | groundstart lines are not all that common. |
| 14:53.02 | *** join/#asterisk orn (~orn@2a01:8280:101:0:4cc7:b020:2256:80de) |
| 14:55.55 | rue_house | http://www.vikingelectronics.com/products/view_product.php?pid=578 |
| 14:56.58 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 14:57.43 | igcewieling | that device costs around $200, seems like it would be better to use a device which supports GS natively. |
| 14:57.53 | *** join/#asterisk iztech (~iztech@76-246-226-131.uvs.irvnca.sbcglobal.net) |
| 14:58.49 | rue_house | think you will find that $200 is less than the difference of hte groundstart equiped devices |
| 15:01.05 | orn | when using users.conf, can you somehow specify (force) which port is used when contacting the peer? |
| 15:02.18 | drmessano | nacho2k: You could also try a Mediatrix 1204 |
| 15:04.08 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 15:05.09 | igcewieling | or one of many many Adtran boxes |
| 15:07.45 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 15:10.25 | nacho2k | great Guys! I will find out,Mediatrix 1204 or Viking... really appreciate your help!! |
| 15:13.27 | Kobaz | it just seems insanity that avaya cannot send callerid number on a sip link |
| 15:15.37 | mjordan | orn: the standard nat settings should be parsed out of users.conf. You should be able to set nat=yes (or nat=auto_force_rport,auto_comedia) and it will use rport |
| 15:16.50 | orn | mjordan: Thanks |
| 15:18.12 | igcewieling | Kobaz: have you considered asking Avaya? |
| 15:19.30 | Kobaz | doing that in a bit |
| 15:19.47 | Kobaz | i don't do avaya, but the avaya guy is going to make some calls |
| 15:20.18 | Kobaz | the avaya guy is used to linking avayas to other avayas, and apparently you need to build some sort of 'domain trust' for it to pass callerid number |
| 15:20.21 | Kobaz | which is really weird |
| 15:20.56 | Kobaz | like it passes the name just fine, but the p-asserted header comes in as invalud.unknown.domain |
| 15:21.39 | igcewieling | keep in mind Asterisk does not by default use PAID for CallerID, it uses the From: header |
| 15:21.51 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 15:22.15 | igcewieling | you can enable PAID to CallerID using the trust_rpid=pai (I think, check the sip.conf.sample) in 1.8 |
| 15:22.31 | Kobaz | i have trustpid |
| 15:22.34 | Kobaz | yeah my end is set up fine |
| 15:22.59 | igcewieling | Does the From: header also have non-useful info? |
| 15:23.32 | Kobaz | correct |
| 15:23.40 | Kobaz | the from: is anonymous.anonymous.invald |
| 15:24.20 | Kobaz | the INVITE is totally useless in terms of proviing caller number |
| 15:24.22 | Kobaz | it sends the name though |
| 15:24.36 | Kobaz | avaya sounds weird and annoying to work with in terms of sip |
| 15:24.46 | Kobaz | i never liked configuring isdn or analog on it either |
| 15:25.00 | Kobaz | 15 pages of setup |
| 15:25.15 | bombev | What does useragent stands for? |
| 15:25.37 | Kobaz | stand for? it doesn't stand for anything |
| 15:25.40 | Kobaz | useragent is useragent |
| 15:25.41 | *** join/#asterisk l2trace99 (~Thunderbi@rrcs-71-43-25-66.se.biz.rr.com) |
| 15:25.56 | Kobaz | it's the device or application that's at the endpoint |
| 15:27.33 | igcewieling | Kobaz: oh well. was worth checking |
| 15:27.39 | Kobaz | yeah |
| 15:27.59 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 15:28.08 | Kobaz | it's like, the avaya has some propriatary thing with domains it seems |
| 15:28.21 | Kobaz | like it needs to 'sync' to another sip endpoint via some sort of domain config |
| 15:28.28 | Kobaz | and then it will 'trust' the endpoint and send it callerid number |
| 15:28.36 | as001 | Hi is it possible to find out which side hang up sip call in asterisk cdr ? |
| 15:28.50 | Kobaz | as001: cdr, nope |
| 15:29.09 | as001 | is there any other way ? |
| 15:31.07 | *** join/#asterisk WinstonNEWS (~WinstonNE@92-67-190-109.dsl.ovh.fr) |
| 15:31.10 | WinstonNEWS | hi |
| 15:31.27 | WinstonNEWS | i have a problem with router thomson tg784n |
| 15:31.41 | igcewieling | as001: yes. |
| 15:31.56 | *** join/#asterisk Tim_Toady (~fuzzy@194.50.55.245) |
| 15:32.04 | WinstonNEWS | {Administrator}=>:nat mapadd intf=Internet outside_addr=109.190.67.92 inside_ad |
| 15:32.04 | WinstonNEWS | dr=192.168.1.68 |
| 15:32.13 | WinstonNEWS | <PROTECTED> |
| 15:33.42 | igcewieling | as001: if the dialplan hits extension "h" then the source hung up. If the call continues to the next priority in the dialplan and you are using the "g" option to dial, then the destination hung up |
| 15:34.54 | as001 | ok thanks igcewieling: |
| 15:35.30 | igcewieling | ast001: This applies to 1.4 - 1.8, older and newer versions may be different |
| 15:35.48 | igcewieling | WinstonNEWS: sounds like a NAT issue. |
| 15:35.54 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 15:37.24 | WinstonNEWS | igcewieling, well yes I think so but how to solve ? |
| 15:38.04 | igcewieling | Fix your NAT problems. If your asterisk server is behind NAT then you need to set your externip, localnet, and disable directmedia |
| 15:38.21 | WinstonNEWS | in the router i have adding : :nat mapadd intf=Internet outside_addr=109.190.67.92 inside_addr=192.168.1.68 for DMZ |
| 15:38.30 | igcewieling | WinstonNEWS: stop doing that. |
| 15:39.30 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 15:39.53 | igcewieling | if you have external phones registering to Asterisk then you need to forward port 5060/udp to the internal asterisk box, but that is about it. Also disable SIP ALG on your router. ALG stands for Aplh Lehhg Glong, which is Elvish for "break SIP" |
| 15:40.47 | WinstonNEWS | igcewieling, i use other port bindport=7777 |
| 15:40.57 | igcewieling | WinstonNEWS: stop doing that too. |
| 15:41.18 | WinstonNEWS | bindport=7777 |
| 15:41.18 | WinstonNEWS | <PROTECTED> |
| 15:41.27 | WinstonNEWS | externip=109.190.67.92 |
| 15:41.36 | *** join/#asterisk PbxMan (c335d91f@gateway/web/freenode/ip.195.53.217.31) |
| 15:41.39 | igcewieling | remove both binds. You are preventing Asterisk from doing the Right Thing |
| 15:41.53 | igcewieling | what is your localnet set to? |
| 15:42.12 | PbxMan | good afternoon |
| 15:42.21 | WinstonNEWS | localnet = 192.168.1.0/255.255.255.0 |
| 15:42.53 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:43.34 | igcewieling | good. do the stuff I suggested and it may start working |
| 15:44.17 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 15:44.32 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:44.42 | WinstonNEWS | igcewieling, disable SIP ALG |
| 15:44.45 | WinstonNEWS | oups |
| 15:44.52 | WinstonNEWS | igcewieling, how to disable SIP ALG? |
| 15:45.45 | igcewieling | WinstonNEWS: contact your router mfgr. |
| 15:46.10 | WinstonNEWS | SIP udp 5060 snooping: enabled translate-predict: enabled |
| 15:46.17 | WinstonNEWS | {admin}[connection]=>bindlist |
| 15:47.05 | *** join/#asterisk Tarcert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 15:48.42 | din3sh | how to play *.raw files recorded with dahdi_monitor |
| 15:50.30 | rue_house | having fun with polycom audio levels? |
| 15:50.53 | rue_house | ASTERISK REALLY NEEDS A REALTIME VU MONITOR TOOL |
| 15:51.36 | rue_house | not just for dahdi channels either |
| 15:51.42 | Qwell | rue_house: So write one. |
| 15:52.17 | rue_house | I tried, couldn't work out enough of the asterisk framework |
| 15:53.12 | rue_house | as I recall the dahdi monitor code completly revolves around the fact it operates with dahdi |
| 15:53.37 | din3sh | tried opening the *.raw file in audacity, recording is garbled |
| 15:54.21 | rue_house | somewhere I have a ulaw to dbm converter function I made for it |
| 15:54.47 | rue_house | din3sh, your channels/samplesize/samplerate are wrong |
| 16:02.20 | WinstonNEWS | connection bindblacklist delete ip=109.190.67.92 proto=udp port=5060 |
| 16:02.27 | WinstonNEWS | Failed to delete bind blacklist entry. |
| 16:05.20 | WinstonNEWS | port 5061 good :p thank igcewieling |
| 16:07.50 | *** join/#asterisk _zoom_ (~zoom@196.1.219.122) |
| 16:08.49 | _zoom_ | I configued my sangoma a102 pri, when I try to call out it gets me "Channel 0/1, span 1 got hangup request, cause 3" any idea |
| 16:08.52 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 16:08.55 | saint_ | hi all |
| 16:09.01 | _zoom_ | cause 3 => to route to destination |
| 16:09.07 | _zoom_ | *no route |
| 16:09.08 | _zoom_ | :p |
| 16:09.24 | saint_ | Can someone tell me if its possible to change somehow the caller ID that a Digium phone receives ? |
| 16:09.49 | Kobaz | receives? Set(CALLERID(num)=1234); |
| 16:10.11 | saint_ | My issue is that I am connected to google, so in the missed calls log I see the "trunk" name. When I hit "DIAL" on the keys, instead of calling the number, Digium tries to call the trunk name and of course tells me that it can't because it's not in the LocalSets. |
| 16:10.31 | saint_ | So any way to get ride of the trunk name and just display the incoming number, so when I hit dial, the phone can send the correct sequence ? |
| 16:11.59 | Katty | infobot: crittercam |
| 16:12.00 | infobot | well, crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 16:12.09 | Katty | so. there's two squirrels sitting on the feeder. right next to each other. |
| 16:12.10 | navaismo | Set(CALLERID(all)=XXXX) |
| 16:12.14 | Katty | this is unheard of. |
| 16:12.26 | Katty | they usual squabble and fight when the other gets too close. |
| 16:12.31 | Katty | i'm thinking they're related, mayhaps. |
| 16:14.48 | jmetro | theyre like REALLY close |
| 16:15.51 | Katty | yes. extremely. |
| 16:15.59 | Katty | maybe they're expecting baby squirrels. |
| 16:16.17 | saint_ | navaismo this is an answer to my question ? |
| 16:18.12 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 16:24.42 | jmetro | Katty: Youre making all the squirrels in your area so fat. I bet that entirecontainer is food. |
| 16:25.00 | Katty | entire...container? |
| 16:25.11 | Katty | magpie market? |
| 16:25.14 | jmetro | From the cam, it looks like a big wooden box filled with seeds |
| 16:25.21 | Katty | well. it's peanuts actually |
| 16:25.23 | jmetro | that they are standing on. |
| 16:25.25 | Katty | but yes, i filled it up this morning ;) |
| 16:25.37 | Katty | oh the tray? yes. it was full of seeds this morning. |
| 16:27.12 | SuperNull | Anyone have SPAs restart midcall on them ? (some techs are saying during registration period) |
| 16:29.48 | ChannelZ | Haven't had that. PoE? Maybe something is freaking out there |
| 16:30.07 | igcewieling | SuperNull: first thing to do is make sure you have the latest firmware |
| 16:33.15 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 16:33.22 | SuperNull | yeahhh |
| 16:33.48 | SuperNull | uhg. cisco ruined the SPA stuffs. We just started provisioning SPA122s .. and they take near 2 minutes each. the old SPA2102s 30 seconds. |
| 16:34.16 | din3sh | configured 2 SPA122s last week |
| 16:35.07 | din3sh | if its supposed to be better than spa2102 then it fails miserably |
| 16:36.08 | din3sh | also deployed Cisco SMB switches which are Linksys one basically |
| 16:37.30 | *** part/#asterisk _zoom_ (~zoom@196.1.219.122) |
| 16:38.01 | igcewieling | SuperNull: Good news then, cisco sold Linksys to Belkin. LOL! |
| 16:38.51 | SuperNull | yeahhh |
| 16:39.02 | SuperNull | i dunno if thats better honestly. i think cisco wanted to make linksys as dirt cheap shit as it could. |
| 16:39.12 | SuperNull | at least Linksys when it was Linksys it was legitimate. |
| 16:39.22 | igcewieling | I was (mostly) joking. Belkin does not have a stellar record |
| 16:39.42 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 16:39.49 | jmetro | avoid netgear for home equipment. |
| 16:40.19 | jmetro | i use D-link nowadays. Before linksys sold i would always use linksys. |
| 16:40.23 | igcewieling | SuperNull: Cisco did not want Linksys products to cut into their enterprise products so they had virtually no incentive to really improve the Linksys line. |
| 16:41.02 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 16:41.44 | SuperNull | yeah. |
| 16:41.53 | SuperNull | we use 'readynet' routers.. opposed to linksys. |
| 16:42.09 | SuperNull | they are basically .. linksys wrt54gl |
| 16:43.03 | SuperNull | only upgraded radios cpus memory.. so not wrt54gl |
| 16:44.24 | jmetro | that sqirrel is OBESE |
| 16:44.38 | igcewieling | We use Adtran NetVantas and TotalAccess boxes, but we do commercial service, not residential. |
| 16:45.07 | igcewieling | jmetro: killing them slowly with kindness. |
| 16:46.25 | saint_ | In Digium Phones call logs, there are 2 columns. The first one seems to be the missed call number. What's the 2nd column for ? If I press DIAL , will Digium phone try to dial the variable in the 1st column or 2nd column ? |
| 16:49.54 | Qwell | That would be Caller ID. |
| 16:49.58 | SuperNull | cisco for business stuff.. not cisco linksys bs. real cisco. like 1800+ series routers. |
| 16:50.26 | Katty | Qwell: YOU |
| 16:50.33 | Qwell | ME |
| 16:50.38 | Katty | Qwell: if it was valentines day |
| 16:50.42 | Qwell | O.o |
| 16:50.52 | Katty | Qwell: and your lady friend was getting you...either a refurbed ipad, or a chromebook |
| 16:50.55 | Katty | Qwell: which would you prefer? |
| 16:51.06 | Qwell | I refuse to use Apple products. |
| 16:51.10 | Qwell | So there's that. |
| 16:51.14 | jmetro | the chromebook! |
| 16:51.23 | Qwell | (does he have a laptop?) |
| 16:51.28 | Katty | Qwell: hrmm. you're not a good test subject then. |
| 16:51.30 | Katty | yes, he has a macbook. |
| 16:51.33 | jmetro | upvotes Qwell for refusing the ipad. |
| 16:51.38 | Qwell | then I doubt he wants a Chromebook. |
| 16:51.44 | Katty | he wants both |
| 16:51.48 | Qwell | well then |
| 16:51.49 | Katty | he's not a mac fanboy or anything |
| 16:51.58 | Qwell | I just meant that it would be redundant. |
| 16:52.01 | Katty | there's also a corsair case he's been eyeballing. |
| 16:52.08 | Katty | well he has an ipad too, 1st gen tho |
| 16:52.13 | Katty | so theoretically, they'd both be redundant |
| 16:52.23 | Qwell | one is an upgrade though |
| 16:52.35 | Katty | he won't be upgrading anything |
| 16:52.38 | Qwell | the other is like... I don't know when I'd ever use a netbook, if I had a laptop (and I do) |
| 16:52.42 | slav3_kitten | so many people buying pads an books :O |
| 16:52.43 | Katty | it'll just be in addition to all the other stuff |
| 16:53.05 | Katty | we have computers and laptops and netbooks and tablets coming out the ears. |
| 16:53.11 | jmetro | kindle > ipad for home entertainment |
| 16:53.19 | Katty | we have one of those too |
| 16:53.32 | Katty | well he does. i'm not much of a tablet reader. |
| 16:54.36 | Katty | any other alternative suggestions to refurbed ipad, chromebook, or corsair case? |
| 16:55.08 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:55.12 | jmetro | a raspberri pi, or a good external hdd |
| 16:55.18 | Katty | he has a raspberry pi |
| 16:55.20 | Katty | and a nas |
| 16:55.32 | Katty | i can't imagine he'd want an external hdd for anything |
| 16:55.54 | jmetro | well, yeah, the nas negates that. |
| 16:56.07 | Katty | a ups for upstairs he might like tho |
| 16:56.17 | Katty | considering the power outage on the storms last night. |
| 16:56.25 | Katty | we lost an 8 port switch in all the ruckus |
| 16:57.37 | navaismo | facepalms |
| 16:58.52 | igcewieling | get him or her a stuffed squirrel |
| 16:59.23 | Katty | i'd be all for a stuffed squirrel! but that's not really his thing. :/ |
| 16:59.25 | saint_ | can someone tell me what does this line return: ${CALLERID(name):2:10} I cant find anything in the doc about the :X:Y stuff in a variable |
| 17:00.26 | SuperNull | saint_ lookup string manipulation |
| 17:00.33 | jmetro | http://www.voip-info.org/wiki/view/Asterisk+func+callerid |
| 17:00.43 | igcewieling | saint_: see the asterisk book, section titled "Using the ${EXTEN} channel variable" |
| 17:00.45 | SuperNull | :X is starting position :Y is length. |
| 17:00.45 | jmetro | "resolution" |
| 17:01.03 | saint_ | igcewieling: just found out.. i was slightly above in the variable section .. thanks |
| 17:01.40 | saint_ | igcewieling: so i m reading something about using asterisk and google.. |
| 17:01.51 | saint_ | because my issue is with the CALL BACK function that does not work. |
| 17:01.57 | igcewieling | saint_: I'm sorry to hear that. |
| 17:01.59 | saint_ | The article is recomending to use this: same => n,Set(CALLERID(number)=${CALLERID(name):2:10}) |
| 17:02.22 | igcewieling | saint_: do you understand what that does? |
| 17:02.33 | saint_ | yeah, it takes the digits from position 2 to 10 |
| 17:02.38 | saint_ | basically strips the 1 |
| 17:02.57 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:03.01 | igcewieling | this assumes the callerid name contains the number. It also assumes you are using an older version of asterisk. |
| 17:03.06 | saint_ | now, if a local number calls a local number, using 7 digits only. would the DID show up as 7 digits, or would the did show the area code first ? |
| 17:03.11 | saint_ | that's something I can't test here.. |
| 17:03.17 | igcewieling | I think CALLERID(number) was deprecated a while ago |
| 17:03.28 | saint_ | i saw that, i m using num |
| 17:03.39 | igcewieling | Be ware of using docs which are incorrect. |
| 17:03.56 | jmetro | seems like the asterisk wiki has LOTS of outdated information. |
| 17:04.02 | saint_ | indeed |
| 17:04.13 | igcewieling | it does. The asterisk book(s) are pretty good though. |
| 17:04.34 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-160-65.red.bezeqint.net) |
| 17:04.46 | saint_ | so i have 5 numbers I can use here, but they are all in different area code: 855, 954, 609, 908, and 201 |
| 17:05.10 | saint_ | can someone tell me something about a number calling another number in the same area: would the DID show 10 or 7 digits ? |
| 17:05.34 | igcewieling | I wish Asterisk allowed setting HANGUPCAUSE and DIALSTATUS using Set() |
| 17:06.36 | file | you can pass cause codes to Hangup() |
| 17:06.54 | file | that's about it for control, alas |
| 17:06.59 | Qwell | What good would setting DIALSTATUS be? |
| 17:07.23 | igcewieling | Qwell: to overide the dialstatus. |
| 17:08.15 | igcewieling | I workaround the issue by setting my own dialstatus and hangupcause variables (named something else) after every dial then the rest of my dialplan uses my variables. |
| 17:08.51 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1003) |
| 17:09.18 | igcewieling | example: http://pastebin.ca/2308964 |
| 17:09.45 | igcewieling | when calling an asterisk box there the dial on the destination box times out, I don't get back a NOANSWER, for example. |
| 17:10.28 | igcewieling | Trivial to work around, but still would be nice. |
| 17:11.41 | igcewieling | When calling an adtran where the dest does not existin on the adtran I get back chanunavail and hangupcause 1, as another example/. |
| 17:13.09 | *** join/#asterisk mathi (~Matthew@109.133.200.219) |
| 17:13.10 | mathi | hi |
| 17:13.15 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 17:13.16 | mathi | [TK]D-Fender, hello |
| 17:14.02 | mathi | [TK]D-Fender, are you there? I have some news about the bad SIP quality I had |
| 17:16.29 | saint_ | BRB |
| 17:28.40 | *** join/#asterisk jsjc (~Adium@186.153.78.188.dynamic.jazztel.es) |
| 17:43.21 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 18:05.43 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:05.55 | *** join/#asterisk Notre1 (~Notre1@208.72.133.110) |
| 18:26.21 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:26.24 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 18:26.44 | Mon|A|rch | is there a free, or reasonably inexpensive TTS system i can use in place of festival? |
| 18:27.02 | Mon|A|rch | festival just isn't cutting it |
| 18:29.00 | SuperNull | you could record every word there is and attempt that. |
| 18:29.50 | igcewieling | Mon|A|rch: no. Cepstral is about as close as you'll get |
| 18:30.49 | navaismo | ceptrals really sounds bad. |
| 18:30.50 | SuperNull | Microsoft Sam ? |
| 18:31.09 | [TK]D-Fender | SuperNull, Dr. Sbaitso :p |
| 18:31.35 | SuperNull | theres always mac os edit tts. |
| 18:31.43 | SuperNull | might require some interesting integration |
| 18:32.14 | [TK]D-Fender | Cepstral is largely decent |
| 18:32.28 | SuperNull | nothing says good like 'largely decent' ;) |
| 18:32.32 | [TK]D-Fender | Good value |
| 18:32.53 | Mon|A|rch | how much does it cost? |
| 18:33.18 | navaismo | maybe for english, for spanih it really sucks |
| 18:33.47 | igcewieling | all TTS sucks, Cepstral seems to suck less than most. |
| 18:34.08 | navaismo | nope, google tts sounds better than cepstral |
| 18:34.09 | Mon|A|rch | sounds good, so how much is it going to run me? |
| 18:34.21 | navaismo | lumenvox sounds better |
| 18:35.40 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:36.15 | [TK]D-Fender | Mon|A|rch, https://www.cepstral.com/store/telephony-server-linux |
| 18:36.35 | [TK]D-Fender | Mon|A|rch, http://www.cepstral.com/demos/ ,--- demo it |
| 18:36.38 | [TK]D-Fender | ^ |
| 18:37.35 | Mon|A|rch | dang, that's expensive |
| 18:38.14 | Mon|A|rch | i guess I'll pitch it, see if they want to buy it |
| 18:39.02 | din3sh | even the licensed version sucks |
| 18:39.36 | [TK]D-Fender | Mon|A|rch, Save-to-file = unlimited concurrency |
| 18:40.39 | Mon|A|rch | what does the concurrency mean exactly? |
| 18:41.00 | Mon|A|rch | how many active tts instances can be active? |
| 18:41.26 | Qwell | There's a link right next to that that explains it... |
| 18:42.55 | Mon|A|rch | thanks |
| 18:42.59 | Mon|A|rch | wasn't paying attention |
| 18:45.49 | Mon|A|rch | [TK]D-Fender, any experience with the google tts? |
| 18:46.30 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:d4c4:d849:49d2:1028) |
| 18:46.47 | [TK]D-Fender | nope, but I did hear it was pretty good and there are guides for setting it up with * |
| 18:46.54 | [TK]D-Fender | Get ... Googling ;) |
| 18:46.58 | [TK]D-Fender | heads off for a bit |
| 18:47.19 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 18:47.20 | saint_ | back |
| 18:47.31 | saint_ | anyone using OpenSIPS or Kamailio with Asterisk ? |
| 18:51.30 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 18:52.33 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 18:56.16 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
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| 19:02.13 | pabelanger | saint_, am now |
| 19:02.16 | pabelanger | kamailio |
| 19:06.57 | saint_ | pabelanger: what for if you don t mind ? |
| 19:13.43 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 19:16.08 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:17.49 | Mon|A|rch | so, having just installed the google TTS |
| 19:17.52 | Mon|A|rch | i can say that it's magical |
| 19:17.57 | Mon|A|rch | in comparison to festival |
| 19:18.27 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 19:18.57 | navaismo | yes but the lenght of the message is limited |
| 19:19.17 | pabelanger | saint_, Will become the core of my SIP network, asterisk will be edge servers for conference, IVR, voicemail, etc |
| 19:19.35 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:19.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:19.51 | saint_ | pabelanger: ok. was it a pain in the butt to setup ? |
| 19:19.56 | igcewieling | pabelanger: in our setup all calls end up being sent to asterisk for processing anyway so we decided at this time to not use Kamailio |
| 19:20.15 | igcewieling | saint_: Asterisk is easy to set up compared to Kamailio or other SIP proxies. |
| 19:20.39 | saint_ | pabelanger: any reasons why you chose Kamailio instead of OpenSIPS ? |
| 19:20.56 | pabelanger | saint_, Not really, maybe 4 hours for the initial setup. But, now comes the fun of learning how kamailio routes things. So, that will take weeks |
| 19:20.57 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:21.06 | saint_ | lol yeah |
| 19:21.21 | carrar | opensips offers a online webex class |
| 19:21.30 | carrar | starting soon |
| 19:21.53 | pabelanger | saint_, no specific reason, kamailio is package in debian, so that helps |
| 19:22.08 | carrar | http://www.opensips.org/Training/EBootcamp |
| 19:22.30 | pabelanger | igcewieling, Yup. We are planning to do more failover stuff, etc |
| 19:23.28 | jmetro | i would definitely like to hear why to choose opensips or kamaillio over the other |
| 19:24.16 | carrar | opensips seems to be reaching out more to help people learn it |
| 19:24.39 | pabelanger | Ya, both do the same, just different people supporting them |
| 19:24.41 | Mon|A|rch | navaismo, not a huge issue |
| 19:24.45 | saint_ | jmetro: probably because they are open source, and they are the top (reverse) proxy available around ? |
| 19:25.02 | jmetro | saint_ which one is |
| 19:25.11 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 19:25.16 | saint_ | jmetro: both OpenSIPS and Kamailio |
| 19:25.28 | jmetro | My question was why hcoose one over the other. |
| 19:25.56 | carrar | I use opensips |
| 19:26.35 | saint_ | Oh.. that's probably a preference of features and GUI. They both came from OpenSER.. |
| 19:26.46 | saint_ | I don t remember which one, but NETFLIX uses one of the 2. |
| 19:26.51 | saint_ | Or Hulu. |
| 19:27.07 | carrar | They both will do what you need, opensips seemed to have better support |
| 19:27.31 | carrar | my personal view point |
| 19:27.56 | carrar | But they are both nice |
| 19:29.34 | igcewieling | you basically have to have a very good understanding of SIP of you want to use a SIP proxy |
| 19:29.40 | carrar | yes |
| 19:29.49 | carrar | read 3261 |
| 19:29.53 | saint_ | lol |
| 19:29.54 | carrar | pretty much a requirement |
| 19:31.21 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:32.31 | Mon|A|rch | navaismo, does googletts only have one voice? |
| 19:32.37 | Mon|A|rch | I'm guessing there aren't any male voices |
| 19:33.44 | navaismo | i guess in my test always a female speak |
| 19:33.47 | ghost75 | using asterisk for receiving fax, whats the maximum wait time the remote sender will accept? |
| 19:34.11 | artyx | You mean how long till it gives up trying to start a handshake? |
| 19:34.37 | ghost75 | i wanted to use wait cmd before receivefax |
| 19:35.37 | *** join/#asterisk Simon-- (~sim@2606:6a00:0:28:5604:a6ff:fe02:702b) |
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| 19:38.14 | Simon-- | what's the reason that hardware DTMF detection is disabled by default with the VPM450? is it broken? since we swapped out a TE410P with one with the echo canceller (3rd gen TE410P), echo went away :), but dtmf detection became lossy |
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| 20:18.18 | Mon|A|rch | so guys, there's this huge delay that happens after i play an mp3 file with mp3player |
| 20:18.30 | Mon|A|rch | some of it is the delay associated with a file not being terminated quickly enough |
| 20:18.35 | Mon|A|rch | but it also tends to take a while |
| 20:18.50 | Mon|A|rch | and I get app_mp3.c:133 Poll timed out |
| 20:18.53 | Mon|A|rch | often |
| 20:19.09 | Mon|A|rch | I haven't the foggiest idea of what that means |
| 20:19.27 | Mon|A|rch | obviously line 133 of the app_mp3 file churns out that error |
| 20:33.08 | *** join/#asterisk F|ReSTaRT (~dlyh@unaffiliated/firestart) |
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| 20:54.48 | *** join/#asterisk ankurgupta0001 (~ankurgupt@216-75-234-234.static.wiline.com) |
| 20:55.37 | ankurgupta0001 | One way audio after changing switching device from wifi to 3g |
| 20:55.47 | *** join/#asterisk paweu (~root@li502-253.members.linode.com) |
| 20:55.58 | paweu | hi |
| 20:55.59 | igcewieling | sounds like a NAT issue |
| 20:56.38 | ankurgupta0001 | yes,everything works fine initially |
| 20:57.22 | ankurgupta0001 | but after switching between wifi and 3g I can only hear the other person |
| 20:57.26 | ankurgupta0001 | he cannot hear me |
| 20:57.36 | ankurgupta0001 | I am using asterisk 11.2 |
| 20:57.39 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:57.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:57.45 | igcewieling | is your asterisk sdrver on the 3g or is the phone on the 3g? |
| 20:57.58 | ankurgupta0001 | phone is on 3g |
| 20:58.12 | igcewieling | try restarting the phone. What codec? |
| 20:58.20 | ankurgupta0001 | u law |
| 20:58.21 | kaldemar | ankurgupta0001: and phone does not hear the other end? |
| 20:58.33 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 20:58.56 | igcewieling | try gsm or (if your asterisk server has a license) g729. I can never get ulaw to work on my 3g, not enough bandwidth on outgoing |
| 20:59.47 | sawgood | $50 or $100 instant PayPal.com bounty offered for help with this (description at) http://pastebin.com/v59WTrwd (drawing at) http://imagebin.org/244835 |
| 20:59.51 | ankurgupta0001 | will try that,but everything till I try to switch between networks on cellphone |
| 21:00.57 | *** join/#asterisk alsuren_ (~dlaban@188-221-138-117.zone12.bethere.co.uk) |
| 21:01.24 | ankurgupta0001 | will try that,but everything works fine till I try to switch between networks on cellphone |
| 21:02.25 | igcewieling | ankurgupta0001: when you switch networks do you get a new IP address? |
| 21:02.38 | ankurgupta0001 | yes |
| 21:02.48 | igcewieling | does your phone know this? Does Asterisk know this? |
| 21:03.20 | igcewieling | sip show peer XXX where XXX is the peer of your phone, verify asterisk sees the new IP. I don't know how to verify this on your phone. |
| 21:03.32 | zerohalo | sawgood: you should be able to do this in about 10 minutes - check out features.conf |
| 21:03.57 | sawgood | zerohalo: looking now |
| 21:04.09 | igcewieling | Didn't I suggest that yesterday, sawgood? |
| 21:04.17 | sawgood | today is a new day! |
| 21:04.19 | zerohalo | I think you did. :) |
| 21:04.58 | navaismo | so cute |
| 21:05.08 | sawgood | oh ... features.conf is neat! |
| 21:05.13 | zerohalo | indeed |
| 21:05.55 | navaismo | loses 50 bucks LOL |
| 21:06.04 | zerohalo | oops - sorry. ha |
| 21:06.08 | ankurgupta0001 | yes asterisk knows it because I can still hear but somehow the other person stops hearing what I am speaking |
| 21:06.43 | navaismo | ankurgupta0001, your provider allow sip over 3g? |
| 21:06.54 | ankurgupta0001 | yes they do |
| 21:07.44 | ankurgupta0001 | somehow the rtp media is causing issues after the ip change |
| 21:07.54 | igcewieling | ankurgupta0001: do NOT assume just because you have one-way audio that asterisk knows the correct up. |
| 21:08.00 | igcewieling | s/up/ip. |
| 21:08.17 | ankurgupta0001 | let me check it again |
| 21:08.30 | ankurgupta0001 | one more thing I am using sip over tcp |
| 21:09.04 | navaismo | sip debug |
| 21:09.09 | igcewieling | unless the phone re-registers when the endpoint IP changes asterisk will not know the correct IP. This is tr[sip over TCP? nevermind, I can't help with that] |
| 21:11.01 | igcewieling | yay, 10 more asterisk servers to upgrade tonight |
| 21:11.29 | rrittgarn | sawgood... got your example working in about 3 minutes... you sure you want to pay $50 for that? |
| 21:11.43 | ankurgupta0001 | thanks a lot for the help,will get back to you with details |
| 21:11.57 | igcewieling | sawgood: likely the information you were looking for is in the Asterisk book. |
| 21:12.04 | navaismo | rrittgarn, hey! leave the deal as $100 |
| 21:13.13 | rrittgarn | navaismo: I saw $50 or $100 wasn't sure which... Either way... |
| 21:14.16 | navaismo | alway go for the top XD |
| 21:14.29 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
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| 21:21.27 | saint_ | Is there a way to broadcast a message or audio file over Digium phones ? |
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| 21:23.42 | jmetro | the same way you put messages out over other phones through asterisk? |
| 21:25.23 | saint_ | yeah. like with current phone system, we can page over phone speakerphone and/or overhead speaker. |
| 21:26.05 | saint_ | With Asterisk, I just want to know if it's possible to page using Digium speakers (from the phones) |
| 21:26.34 | jmetro | oh, paging. I think thats standard. Its in the book actually. |
| 21:26.59 | saint_ | yeah. i just found something about that in chapter 11. reading it now. looks like it s doable. perfect. |
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| 21:45.16 | ChannelZ | heh..chapter 11 |
| 21:45.53 | slav3_kitten | isn't that bankruptcy? |
| 21:46.00 | leifmadsen | ya |
| 21:46.11 | slav3_kitten | poor saint_ an his bankrupt asterisk server |
| 21:46.18 | slav3_kitten | we should mourn it |
| 21:48.34 | newtonr | saint_: https://wiki.asterisk.org/wiki/display/DIGIUM/Ringing+Types+and+Intercom |
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| 22:01.57 | artyx | SAweeet. I just lowballed someone on a sangoma b600 |
| 22:02.29 | artyx | and best of all, if it doesnt work, i'll simply swap it out w/ the one at work ;) |
| 22:06.06 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 22:09.44 | sawgood | isn't that 'like' stealing? |
| 22:10.00 | artyx | not really?> |
| 22:10.41 | artyx | Well, no more than sifting through rubbish bins for recyclables |
| 22:10.52 | sawgood | just kiddin ... |
| 22:10.54 | sawgood | ha! |
| 22:10.59 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 22:11.36 | artyx | There are some neighbors that are peeving me off with their trash can |
| 22:11.42 | artyx | but thats OT ;) |
| 22:12.00 | artyx | I can say ford escape goes for the punt, and its GONE! |
| 22:12.15 | sawgood | I like putting stuff out of the curb (some nice stuff) and then seeing how long it makes it |
| 22:12.30 | artyx | before someone nicks it? or do you put a sign up saying free or what |
| 22:12.31 | sawgood | sometimes, they 'pick' stuff off the entire dump, but that is ok too I guess |
| 22:13.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:16.20 | artyx | <PROTECTED> |
| 22:16.58 | artyx | The frame went in no time, but the mattress, well i had to cut that apart and just toss it in the next 3 neighboring apt complex dumpsters :P |
| 22:17.20 | sawgood | sometimes I run the garbage disposal at night to piss off neighbors, but I draw the line at that |
| 22:17.34 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:17.42 | artyx | this... simian liked to to aerobic activity. in a 70r old building. living upstairs |
| 22:17.51 | artyx | But we trip ballz ONE time out of the yr and she calls the fire department on us |
| 22:17.54 | artyx | ;) |
| 22:18.10 | artyx | simmian? cimian? something like that |
| 22:19.08 | sawgood | sa-moan-in? |
| 22:19.09 | artyx | Thats okay. we got her good |
| 22:19.18 | artyx | REALLY really good |
| 22:19.28 | artyx | (after she mixed bleach and ammonia produts in the kitchen sink) |
| 22:19.36 | artyx | the one that overflows into our area |
| 22:19.41 | *** part/#asterisk Simon-- (~sim@2606:6a00:0:28:5604:a6ff:fe02:702b) |
| 22:20.05 | artyx | Never living in a low rent ghetto-rear ended apartment again |
| 22:21.05 | sawgood | what city are youin? |
| 22:21.11 | artyx | Sandy Eggo |
| 22:21.35 | sawgood | I do not know where that is at |
| 22:21.51 | sawgood | oh San Diego Chargers |
| 22:22.17 | artyx | Hey nobody has ever played Commander Keen |
| 22:22.24 | artyx | I learned sandy eggo from that, a long time ago |
| 22:22.44 | artyx | That, and this one joke from the offspring concerth ere.. "What do you get when you lay a waffle on the beach" |
| 22:23.10 | artyx | A sandy $#@!(%^ eggo |
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| 22:43.02 | teloniusz | hi, authentication problem. I have an user in sip.conf: type=user, host=dynamic, username, secret |
| 22:44.17 | teloniusz | I'd like to authenticate only at invite, without registration, but then invite from another asterisk ends with 'extension not found in context default', when contet defined in my user is other than default, of course |
| 22:46.11 | teloniusz | It seems to work when my user sets up registration, but when two users register from different IPs, only one of them can call and the other encounters the same problem, his extensions seem to fallthrough to context default, as if the peer was not found |
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