| 00:01.07 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 00:01.14 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-rc5 (2010/10/18), 1.6.2.13 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon |
| 00:01.49 | *** join/#asterisk eugeneoden (~goden@99-62-173-58.lightspeed.austtx.sbcglobal.net) |
| 00:01.58 | leifmadsen | Asterisk 1.8.0 Release Candidate 5 has been released. The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform |
| 00:01.58 | leifmadsen | compatibility IPv6 changes. In addition, the availability of the English sound |
| 00:01.58 | leifmadsen | prompts with Australian accents has been added. See the release announcement for more info: http://www.asterisk.org/node/51453 |
| 00:08.32 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 00:10.41 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:13.17 | *** join/#asterisk trelane` (~trelane@funtoo/staff/trelane) |
| 00:20.42 | *** join/#asterisk rdircio (~rdircio@189.137.181.208) |
| 00:20.56 | rdircio | hi |
| 00:21.15 | jsgoecke | hi |
| 00:21.26 | rdircio | what would be the correct syntax to push "async" to a originate command in the asterisk cli? |
| 00:21.38 | rdircio | i'm doing: channel originate DAHDI/1/0445537482420 application playback hello-world Async |
| 00:21.53 | rdircio | but i don't seem to get feedback about the call |
| 00:22.52 | rdircio | i'd like to get the status.... (no answer/answered/congested/line offhook) |
| 00:22.52 | rdircio | etc |
| 00:26.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 00:27.39 | *** join/#asterisk fofware (~Fabian@host165.186-109-184.telecom.net.ar) |
| 00:28.03 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 00:36.04 | pabelanger | rdircio: Then don't use Async |
| 00:36.18 | pabelanger | otherwise you have to monitor the events to see the status |
| 00:36.42 | rdircio | i want to get that feedback |
| 00:36.46 | rdircio | but, how? |
| 00:39.09 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 00:39.17 | pabelanger | rdircio: Like I said, you don't want to use async then |
| 00:39.32 | rdircio | i'm not |
| 00:39.39 | rdircio | but i don't get anything |
| 00:39.50 | pabelanger | <rdircio> i'm doing: channel originate DAHDI/1/0445537482420 application playback hello-world Async |
| 00:40.01 | rdircio | i yanked it out |
| 00:40.11 | rdircio | but not getting call progress |
| 00:40.47 | pabelanger | pb your CLI |
| 00:41.17 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:41.20 | rdircio | http://pastebin.ca/1966388 |
| 00:43.14 | rdircio | getting nothing :9 |
| 00:43.31 | pabelanger | connect to asterisk first, then call originate. |
| 00:44.48 | rdircio | a bit better |
| 00:44.48 | rdircio | http://pastebin.ca/1966394 |
| 00:44.53 | rdircio | but... want to script that |
| 00:44.56 | rdircio | :) |
| 00:46.22 | pabelanger | try: asterisk -rx "originate DAHDI/1/0445537482420 application playback hello-world" |
| 00:47.11 | rdircio | got no feedback at all |
| 00:47.12 | rdircio | :( |
| 00:47.31 | rdircio | http://pastebin.ca/1966396 |
| 00:48.19 | ChannelZ | originate isn't a command |
| 00:49.02 | pabelanger | Looks like you'll have to use AMI |
| 00:49.17 | ChannelZ | it's "channel originate" |
| 00:49.33 | rdircio | i also tried that |
| 00:49.34 | rdircio | :) |
| 00:49.38 | rdircio | both do make a call |
| 00:49.46 | rdircio | none gives feedback about it |
| 00:50.15 | ChannelZ | scrolls back to read what you're even trying to accomplish |
| 00:50.31 | pabelanger | ChannelZ: Yes, it is a command |
| 00:51.09 | pabelanger | rdircio: just tested it locally, you seem to be correct. I would have expected the -rx "originate" to return something |
| 00:51.15 | rdircio | i want to make a call to a dahdi channel, play a message and hangup, get the status of the call... |
| 00:51.38 | pabelanger | rdircio: I know using AMI, will return a result |
| 00:51.50 | rdircio | what sort of api you use for ami? |
| 00:51.55 | p3nguin | How about using a local channel? |
| 00:51.56 | rdircio | i am trying asterisk-java |
| 00:52.03 | rdircio | not sure if it's any good |
| 00:52.14 | rdircio | a local channel? |
| 00:52.16 | pabelanger | I use a simple PHP class I wrote |
| 00:52.29 | rdircio | opening a socket to asteisk manager? |
| 00:52.33 | pabelanger | Yes, local channel is the other option |
| 00:52.47 | pabelanger | rdircio: Yup |
| 00:53.07 | rdircio | tried that... |
| 00:54.19 | rdircio | sort of just hangs |
| 00:54.26 | pabelanger | rdircio: For example: http://svn.radar-project.org/svn/radar/trunk/library/Asterisk/Manager.php |
| 00:54.45 | rdircio | that site requries a username/pass |
| 00:55.04 | pabelanger | heh, though anonymous access was permitted. |
| 00:55.06 | *** part/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 00:55.35 | rdircio | emm |
| 00:55.46 | rdircio | i tried anonymous/myemail |
| 00:55.48 | rdircio | nope |
| 00:55.50 | rdircio | erggg |
| 00:56.47 | rdircio | cannot access.. |
| 00:56.48 | rdircio | pb? |
| 00:56.49 | rdircio | :) |
| 00:57.18 | pabelanger | rdircio: Send an email off to enable it, not sure how long it will take |
| 00:57.23 | *** join/#asterisk TJ_stamm (~TJ@c-98-213-150-32.hsd1.il.comcast.net) |
| 00:58.15 | ChannelZ | AMI isn't that hard, go look at voip-info |
| 00:58.30 | ChannelZ | http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 00:59.02 | rdircio | interesting |
| 00:59.04 | rdircio | :) |
| 00:59.07 | rdircio | thx |
| 01:04.59 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 01:05.18 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 01:05.49 | joobie | hey guys.. ive got one polycom 320 that whenever a voicemail is left for the user, the LED doesn't light up.. it works on all the other phones - any idea how to start debugging this? had a look at the voicemail config and it all looks the same...... |
| 01:06.55 | Chainsaw | joobie: Well, it depends on how the MWI (Message Waiting Indicator, which is the official name for that fancy light) is triggered. |
| 01:07.39 | Chainsaw | joobie: I have noticed that some of the features like buddies stop working if you force the phone to use SIP over TCP exclusively (no UDP). So if that sounds like something you just did for that one phone, best undo it. |
| 01:09.26 | joobie | Chainsaw, the configs of these phones are similar afaik |
| 01:09.32 | joobie | they were all based on the same template config |
| 01:09.36 | joobie | but ill confirm later.. |
| 01:09.47 | joobie | any other ideas? don't have access to the phone atm to test but would be good to have a few under my hat |
| 01:09.54 | Chainsaw | joobie: There are two sides to this. The config on the phone and the dialplan on the Asterisk box. |
| 01:10.06 | joobie | ahh |
| 01:10.08 | Chainsaw | joobie: What is special about this one phone. Is the station ID out of the normal range? |
| 01:10.14 | joobie | no |
| 01:10.25 | joobie | it looks identical to the others |
| 01:10.27 | Chainsaw | joobie: Is it sitting behind a router/firewall of its own? |
| 01:11.13 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:11.18 | joobie | no |
| 01:11.29 | Chainsaw | joobie: Does the usual operator of this phone handset look flushed or otherwise panicky? |
| 01:11.30 | joobie | it's in the same lan segment as the other phones that work |
| 01:11.35 | joobie | haha :) |
| 01:11.46 | joobie | from what I can tell, it's not user error |
| 01:11.54 | joobie | but im speaking to them remotely saying "is the light coming on" |
| 01:12.08 | joobie | and they say "no".. so im presuming they are not color blind |
| 01:12.13 | Chainsaw | joobie: And you are positive that the user and you are thinking of the same light? |
| 01:12.50 | Chainsaw | joobie: Because the combination of circumstances sounds rather unlikely, unless you have not answered one of the previous three questions honestly. |
| 01:13.36 | joobie | Chainsaw, there's only one light on the phone on the 320 |
| 01:13.49 | joobie | and the above is honest... |
| 01:14.17 | Chainsaw | joobie: Then you win. It will forever be a mystery. The actual cause will be lost in the mists of time. |
| 01:14.18 | joobie | im just sifting through a 'sip show peer X' compared to a phone that is working to see any differences in there.. all looks ok so far |
| 01:15.09 | joobie | ok one diff |
| 01:15.16 | joobie | the phones upports g729 and alaw |
| 01:15.22 | joobie | the other phone that works supports just g729 |
| 01:15.41 | Chainsaw | So the configuration is not similar. |
| 01:15.43 | Chainsaw | Focus on that. |
| 01:15.47 | joobie | firmware is the same on the phone working / not working |
| 01:15.59 | Chainsaw | Sure, but the configuration is not the same. |
| 01:16.16 | Chainsaw | Whether that is on the phone side or on the Asterisk server side is for the moment irrelevant. Focus on differences, not similarities. |
| 01:17.01 | joobie | ok |
| 01:17.16 | joobie | i have disallow=all and allow=g729 set.. going to get them to reboot and test |
| 01:42.14 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:58.24 | *** join/#asterisk Daejeo (889f100e@gateway/web/freenode/ip.136.159.16.14) |
| 01:59.12 | Daejeo | hello guys :) |
| 02:00.17 | *** part/#asterisk TJ_stamm (~TJ@c-98-213-150-32.hsd1.il.comcast.net) |
| 02:00.45 | Daejeo | I am going to buy a new nokia cellphone, and i want to use it with asterisk. anyone knows ..........which version is compatible with asterisk? |
| 02:02.53 | pabelanger | how do you plan to interface it? |
| 02:02.59 | lanning | I like the N97. And it has built-in SIP support... :) |
| 02:05.01 | Daejeo | pabelanger: it will behind NAT |
| 02:24.49 | *** join/#asterisk DrDigi (~mmurphy@75-146-139-38-Stockton.hfc.comcastbusiness.net) |
| 02:32.15 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 02:39.45 | *** join/#asterisk ritztech (ritztech@ip70-189-221-21.lv.lv.cox.net) |
| 02:39.52 | *** join/#asterisk ickmund (~magnus@cli-5b7e85d4.bcn.adamo.es) |
| 02:40.14 | ritztech | is there dialtone outbound provided on dahdi |
| 02:42.32 | [TK]D-Fender | ritztech: When do you provide dialtone OUT? |
| 02:42.51 | [TK]D-Fender | ritztech: When you go out, the OUTSIDE gives you dialtone.... |
| 02:47.13 | sshock | the echo test is weird; it doesn't seem to work unless it is preceded by a Playback() command |
| 02:47.28 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:48.28 | sshock | I tried adding an Answer() step first, but that didn't help. |
| 02:49.49 | rdircio | now doing AGI |
| 02:49.56 | rdircio | to generate a call and deliver amsg |
| 02:50.02 | rdircio | although i only get "Originate successfully queued" |
| 02:50.19 | rdircio | how do i get more info? busy, hangup, no dialtone |
| 02:50.21 | rdircio | :( |
| 02:50.23 | sshock | oh well, not a big deal; just weird |
| 02:50.23 | rdircio | :) |
| 02:53.58 | ritztech | hmmm it seems Really weird its like really intermittent ( i have a field tech on site testing with a butt set and its like im not getting Dial tone) |
| 02:56.35 | ritztech | is there a way to wait for dialtone before prceeding |
| 02:56.49 | [TK]D-Fender | rdircio: Originate spawns a completely new call. It is not related to anything else. What you originate must look after itself |
| 02:56.58 | rdircio | hmmm |
| 02:57.07 | rdircio | so, how do i .... |
| 02:57.12 | rdircio | get its status? |
| 02:59.04 | ritztech | so can i wait for dialtone BEFORE outpulsing digits.... |
| 02:59.05 | [TK]D-Fender | rdircio: Go look at what KIND of channel you are originating. that will determine what you can see. |
| 02:59.14 | rdircio | DAHDI |
| 02:59.15 | ritztech | is that a BIG W instead of a pause |
| 02:59.27 | [TK]D-Fender | ritztech: w <------------------------- |
| 02:59.35 | [TK]D-Fender | ritztech: Thats right. |
| 02:59.41 | [TK]D-Fender | ritztech: MORE FUCKING W'S |
| 02:59.48 | [TK]D-Fender | ritztech: pLEASE DEAL WITH THIS :P |
| 03:00.13 | [TK]D-Fender | ritztech: What it to wait a sec before starting? SHOVE THEM IN FRONT. |
| 03:00.30 | [TK]D-Fender | ritztech: Want it to dial slower? shove them in BETWEEN DIGITS |
| 03:02.52 | ritztech | i have i put 10 of them and i hear a pause and now its too long |
| 03:04.20 | ritztech | whenever a call works i get dialtone from the Mitel But everytime we Dont get dialtone it doesnt As someone has a Butt-set on the other end |
| 03:07.59 | sshock | this sounds just like a conversation from last night |
| 03:11.15 | ritztech | i just did this waitfordialtone=yes in chan_dahdi see how it works |
| 03:12.18 | pabelanger | ritztech: if you want to access ${DIALSTATUS}, you'll need to use a local channel and do it via the dialplan |
| 03:14.08 | pabelanger | EG: http://forums.digium.com/viewtopic.php?t=5077 |
| 03:14.42 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:14.53 | ritztech | whats dialstatus ? |
| 03:15.15 | jsgoecke | http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS |
| 03:15.55 | pabelanger | ritztech: *CLI> core show application Dial |
| 03:18.04 | ritztech | shazzy |
| 03:18.51 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 03:22.09 | ritztech | is there a way to WAIT for dialtone before outpulsing Digit |
| 03:22.10 | ritztech | s |
| 03:23.04 | ritztech | seems like tahts what the field tech is hearing and seeing on his ButtSett |
| 03:24.06 | sshock | ok, here's an interesting one: when I try to dial a user the is not connected, my SIP client prompts me for authentication |
| 03:24.26 | sshock | why would it do that? why not just return a busy signal or unavailable or whatever |
| 03:25.26 | sshock | oh, duh; I need a rule after the Dial |
| 03:25.42 | [TK]D-Fender | ritztech: not AFAIK. |
| 03:25.43 | sshock | ok, what should I put as the next step after Dial? Just Hangup()? |
| 03:26.40 | sshock | I thought asterisk by default when it ran out of rules did a busy signal or something; isn't autofallthrough supposed to do that? |
| 03:27.30 | [TK]D-Fender | sshock: Did youSET it? |
| 03:27.38 | sshock | yep |
| 03:27.43 | [TK]D-Fender | sshock: So far you haven't shown us ANYTHING |
| 03:27.55 | sshock | autofallthrough=yes in my general section |
| 03:28.58 | sshock | and there's one rule for extension 800 in my extensions.conf: exten => 800,1,Dial(SIP/joe) |
| 03:29.17 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-evydkdlhhsildesb) |
| 03:29.53 | sshock | when I try to call 800 and joe is not connected, it prompts me for authentication; that's weird it should have given me a BUSY or something |
| 03:30.07 | [TK]D-Fender | sshock: PASTEBIN IS YOUR FRIEND <------- |
| 03:30.08 | [TK]D-Fender | ~pb |
| 03:30.09 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 03:31.46 | sshock | I know what a pastebin is; what more do you need to see? |
| 03:32.24 | sshock | you want to see the whole extensions.conf? |
| 03:34.36 | [TK]D-Fender | sshock: What is "prmpt for authentication" mean? How is it "asking"? You aren't SHOWING US what is happening, we are only gettting a broken description |
| 03:35.19 | [TK]D-Fender | sshock: you say things are set. We only have your word for it. |
| 03:35.32 | sshock | ok, let me paste a link to my extensions.conf |
| 03:35.38 | [TK]D-Fender | sshock: and show us the CALL |
| 03:35.47 | sshock | the "prompt for authentication" is what my SIP client (twinkle) is doing |
| 03:36.47 | sshock | here's the extensions.conf: http://pastebin.com/jEHswh8j |
| 03:37.44 | sshock | if both tammy and phillip are connected, I can call between them |
| 03:38.30 | sshock | but if, e.g., I disconnect phillip and I try to call phillip from tammy, then my sip client pops up an "authenticate to realm" dialog |
| 03:38.41 | [TK]D-Fender | sshock: Show. Us. The. FAILURE. |
| 03:39.01 | sshock | the failure is that it should simply say "line busy", not prompt me for authentication |
| 03:39.25 | [TK]D-Fender | ....... |
| 03:40.40 | rdircio | WOOOO |
| 03:41.11 | sshock | hmm, maybe these to-gizmo rules are conflicting... |
| 03:41.21 | sshock | (I haven't fully set that one up yet) |
| 03:41.47 | sshock | ok, I commented out include => to-gizmo, but it's the same problem |
| 03:42.05 | [TK]D-Fender | waits for the monologue to end..... |
| 03:42.11 | sshock | if I add an exten => phillip,n,Hangup(), then the problem goes away but it does a hangup rather than a busy |
| 03:42.21 | sshock | ok, monologue over :) |
| 03:43.25 | rdircio | woooo |
| 03:43.26 | rdircio | http://pastebin.com/JipP8HxN |
| 03:44.14 | rdircio | that gives me the call status http://pastebin.com/isK2RAip |
| 03:44.27 | rdircio | thx guys :) |
| 03:46.17 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 03:47.13 | rdircio | i [heart] * |
| 03:49.34 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 03:50.18 | sshock | I shouldn't be required to do a Hangup() or something after every Dial() should I? |
| 03:51.10 | sshock | I do see in the example file a Goto(s-${DIALSTATUS},1) after some of the Dial commands, |
| 03:51.28 | sshock | and then there's like exten => s-NOANSWER,1,Hangup |
| 03:52.22 | sshock | but even without any of that, shouldn't the default after a Dial() be to go to the next step and if there are no more steps it should hang up or something |
| 03:57.33 | sshock | ok, just checked with wireshark and here's what's happening |
| 03:57.59 | sshock | my client is doing an INVITE for sip:phillip@mydomain.com |
| 03:58.11 | sshock | * responds with 503 Service Unavailable |
| 03:58.16 | sshock | however, then my client tries again |
| 03:58.28 | sshock | and * responds with 407 Proxy Authentication Required |
| 03:59.06 | sshock | why is * trying to tell me I need to authenticate? why doesn't it keep responding with 503 ? |
| 03:59.43 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 04:10.59 | sshock | ok, I think I may have figured it out |
| 04:11.19 | sshock | internally my server can be reached as example.home rather than example.com |
| 04:11.41 | sshock | and I think it is reacting that way when the domain doesn't match the realm |
| 04:12.14 | sshock | anyway, I'll play with it some more; maybe I need to set "domain" in my sip.conf to prevent this problem |
| 04:14.05 | [TK]D-Fender | ... |
| 04:19.00 | sshock | I was just connecting to it using the internal name earlier, because I hadn't set up my firewall yet |
| 04:19.34 | sshock | speaking of which, when I use the public hostname, the registration seems to not always work; maybe I need to open up some more ports on my firewall besides 5060 udp? |
| 04:24.55 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 04:25.38 | russ | I am getting bad call quality on IAX2, I did a packet capture and I see something odd |
| 04:25.48 | russ | the timestamps go like this: |
| 04:26.54 | russ | 17381, 17401, 17421, 17441, 17461, 17461, 17501, 17521, 17541, 17561, 17581, 17581, 17621, etc |
| 04:30.34 | sshock | crap, that didn't fix it |
| 04:35.18 | *** join/#asterisk phonic (~jbabiak@pool-71-180-124-168.tampfl.fios.verizon.net) |
| 04:35.23 | phonic | hi |
| 04:36.37 | phonic | does anyone know if the call presentation parameter in Dial can record/play any other codec besides gsm? It seems to be hardcoded, as I see no configuration options. |
| 04:39.46 | pabelanger | phonic: which parameter (letter)? |
| 04:39.52 | phonic | p |
| 04:40.08 | phonic | specifically the lower case (non database) one |
| 04:41.12 | pabelanger | Hmm, not sure. Looks to be hardcoded to gsm, for whatever reason |
| 04:41.39 | pabelanger | I would _assume_ it would have used the codec defined by the channel |
| 04:42.08 | phonic | yeah, kind of sucks. The parameter/application specific sound files are encoded in both gsm and ulaw, so I had some hope. But on either ulaw or g729 calls, it always uses gsm. |
| 04:42.28 | phonic | and when I disabled the gsm codec modules, the screening wouldn't work at all and errored out |
| 04:43.33 | pabelanger | phonic: Sounds like a bug / feature request. Create a new issue on the tracker and I'll see what we can do to triage it |
| 04:44.10 | phonic | cool, thanks. |
| 04:45.05 | phonic | i wrote (half borrowed) a macro that does it, but I would prefer to just use the p parameter if possible. |
| 04:45.27 | russ | anyway, the packets with duplicated timestamps contain different data |
| 04:45.47 | *** join/#asterisk BoTheBlade (~BoTheBlad@5.3.ppp.dianet.info) |
| 04:45.54 | pabelanger | phonic: It shouldn't be too hard to add a new option to the 'p' flag |
| 04:46.18 | BoTheBlade | Hello everyone! |
| 04:47.51 | phonic | pabelanger: quick follow-up question. I imagine the answer is no, but I figured it would be worth a shot to ask. The way I am trying to set this up is to call a group of numbers at the same time, including a cell phone, and to guarantee attempted delivery of the call even if the cell phone is off and goes right to voicemail. The problem is, if I call it like: Dial(SIP/1&SIP/2&SIP/3,20,tpA(beep)), it will call all three numbe |
| 04:47.51 | phonic | rs, but as soon as one answers, it stops ringing the others. Sort of defeats the purpose. Is there any clean way to make it continue to ring all the numbers until one 'accepts' the call? |
| 04:49.00 | pabelanger | phonic: should like you want FollowMe |
| 04:49.01 | [TK]D-Fender | phonic: * can play any format it can read, for with there are like a dozen formats |
| 04:49.40 | phonic | [TK]D-Fender, yeah but the p parameter doesn't seem to accept any others. |
| 04:49.42 | BoTheBlade | I have a question: is it very hard to modify an Asterisk 1.6.2 Bridge command? Have anyone took a look into it's source? |
| 04:49.49 | phonic | pabelanger, would that work with call screening? |
| 04:50.22 | pabelanger | phonic: not sure honestly. |
| 04:50.48 | [TK]D-Fender | phonic: Incorrect |
| 04:51.13 | [TK]D-Fender | phonic: If * picks a GSM file, then tats because its the best OPTION for the file that is being requested <- |
| 04:51.20 | BoTheBlade | I need this command not to hangup both channels upin hangup of one. I've tried to comment a line in the source which I _assume_ to do it, but no effect |
| 04:51.59 | BoTheBlade | By the way, I'm awfully sorry for messing into your conversation. |
| 04:52.20 | phonic | [TK]D-Fender: Unless there is a configuration parameter I am missing, that doesn't apply to the Dial 'p' parameter. If gsm is not enabled, it will not work period. It will use the ulaw files for the prompts, but it will not record in any other format for the recorded name. |
| 04:53.30 | phonic | the entire call presentation application is built into app_dial and everything appears hard coded to record as gsm only |
| 04:54.02 | pabelanger | phonic: what version of Asterisk are you using? |
| 04:54.44 | phonic | pabelanger: 1.4.31 |
| 04:56.29 | pabelanger | phonic: Mind showing us the problem? PB it |
| 04:56.59 | phonic | pabelanger: the issue with it not working if gsm is disabled? |
| 04:57.00 | pabelanger | the only specific code for gsm is ast_play_and_record |
| 04:57.16 | pabelanger | ya |
| 04:57.55 | phonic | sure, one sec. I'll do it with and without |
| 04:58.12 | pabelanger | I would think it would fail at priv-recordintro, if gsm was not loaded |
| 05:01.15 | phonic | http://pastebin.com/nnrVNmQ5 |
| 05:02.10 | *** join/#asterisk sourcode (~code@ppp-58-8-131-245.revip2.asianet.co.th) |
| 05:02.25 | pabelanger | phonic: try this patch |
| 05:02.25 | pabelanger | http://pastebin.com/4tzGmitd |
| 05:03.18 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 05:04.20 | phonic | pabelanger: thanks. I'll have to check it out tomorrow because I'm running astlinux on a via arch and I will need to cross-compile it and build it into a custom image for the box. |
| 05:05.22 | pabelanger | phonic: Ok, I'm pretty sure it'll fix your problem. Ping after you have tested it and I'll merge the code |
| 05:05.47 | [TK]D-Fender | Or you could just implement this yourself in M() and bypass all of that mess |
| 05:06.02 | phonic | so this will replace gsm with slin? |
| 05:06.28 | pabelanger | phonic: yes, slin should be used, since it is the base format for asterisk |
| 05:07.02 | phonic | cool. and it will cut back on the transcoding overhead which was causing some issues. |
| 05:07.25 | pabelanger | yar |
| 05:07.59 | [TK]D-Fender | SLN implies that it should be transcoding |
| 05:08.06 | [TK]D-Fender | NOT good |
| 05:08.18 | ritztech | penguins |
| 05:08.25 | [TK]D-Fender | It should record in the codec of the call direct so that its just repacking frames. |
| 05:08.54 | phonic | if I changed sln to g729 in your modification, would that work? |
| 05:09.14 | [TK]D-Fender | .... |
| 05:09.25 | [TK]D-Fender | phonic: Codec. Of. The. Call. |
| 05:09.31 | pabelanger | difficult to do, if you inbound call is g729 and outbound is ulaw |
| 05:09.38 | [TK]D-Fender | SLN is just guranteed wrong |
| 05:09.52 | phonic | that is the codec of the call, which is why I asked |
| 05:09.58 | [TK]D-Fender | paif thats the case he's transcoding anyway. No gains <- |
| 05:09.59 | phonic | well, not all, but some |
| 05:10.32 | phonic | there would still be gain. if it records it in gsm, it has to do g729->slin->gsm, correct? |
| 05:11.38 | pabelanger | correct |
| 05:11.48 | phonic | so recording in slin directly would remove the extra step |
| 05:11.55 | pabelanger | the patch would be sln->g729 |
| 05:12.32 | pabelanger | yes. The issue is, unless inbound and outbound are the same codec, there will be transcoding |
| 05:12.48 | phonic | in my case, they generally are |
| 05:12.48 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 05:13.00 | phonic | some use ulaw, some g729 |
| 05:13.25 | phonic | but it's on both segments. we don't really want to do any transcoding on the pbx itself |
| 05:13.26 | [TK]D-Fender | Sounds like there is almost no point in a "patch" |
| 05:14.00 | phonic | if i can record in sln or g729 there is |
| 05:14.22 | [TK]D-Fender | phonic: that is only if both legs are in G.729. Otherwise you're transcoding |
| 05:14.27 | pabelanger | sure it is, removes the dependency of gsm |
| 05:14.28 | [TK]D-Fender | (anyway) |
| 05:14.44 | [TK]D-Fender | Is jus having GSm available at all the point here? |
| 05:14.45 | phonic | [TK]D-Fender, I said three times that they were |
| 05:14.57 | sshock | ok, I just figured out one problem; I don't know if it's a bug in * or not, |
| 05:15.16 | *** join/#asterisk trelane_ (~trelane@funtoo/staff/trelane) |
| 05:15.27 | sshock | but I had bindaddr 0.0.0.0, so it was binding to both my local and external interface |
| 05:15.32 | phonic | according to show translation, g729->slin is 6ms. g729->gsm is 17. That's almost triple. |
| 05:15.57 | sshock | when I try to connect to it using external IP, it was replying with the internal IP |
| 05:16.12 | [TK]D-Fender | sshock: Improper NAT setup <- |
| 05:16.21 | [TK]D-Fender | sshock: Whole pile of setting you need to do. |
| 05:16.26 | [TK]D-Fender | ~sipnat |
| 05:16.26 | infobot | extra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:16.49 | sshock | well, I just changed it to only bind to the external IP, and now it behaves appropriately |
| 05:18.00 | phonic | pabelanger: thanks again for the info and patch. So assuming it works, is that going to be a change in the 1.4.x code across the board (ie: doing away with gsm in favor of slin)? |
| 05:18.25 | pabelanger | phonic: Correct. It should help with both transcoding and gsm dependency. The only other option would be to create another parameter to do what app_voicemail does, records the prompt is all formats you tell it too |
| 05:18.43 | sshock | my * is not behind a NAT device; it is my firewall/server/router itself with a public IP |
| 05:18.50 | pabelanger | phonic: Yes, if I get sign off |
| 05:19.02 | sshock | so I don't think what you said applies |
| 05:19.04 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 05:19.07 | pabelanger | phonic: but report back your results please |
| 05:19.19 | phonic | pabelanger: or it could just choose the best codec based on a call by call basis dynamically :) |
| 05:19.40 | phonic | i will |
| 05:19.58 | pabelanger | phonic: FWIW: app_followme.c uses ast_play_and_record(...,"sln",...) |
| 05:20.21 | phonic | hmm |
| 05:22.43 | [TK]D-Fender | sshock: Well your description is weak and you've shown nothing of value to date. There is nothing to debug and you are just monolguing for the past few hours here. |
| 05:23.15 | sshock | I think it may be an * bug; what should I do to help prove/disprove it? |
| 05:23.23 | sshock | paste debug logs? |
| 05:23.56 | *** join/#asterisk sawgood (~sawgood@173-13-158-26-sfba.hfc.comcastbusiness.net) |
| 05:24.05 | [TK]D-Fender | sshock: No, you should profess its a bug and we'll take your word for it and start searching for geese |
| 05:24.24 | sshock | geese? |
| 05:25.52 | sawgood | So, if you have many remote SIP end points registering to an Asterisk 1.6 box (which is not in a NAT envoirnment) ... but the remote phones are (should all the remote phones be registering) on port 5060 on the Asterisk box? |
| 05:26.10 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 05:26.28 | sshock | sawgood: are you addressing me? |
| 05:26.28 | sawgood | I ask this because I have some registering on 5060 and one or two at 5062, and even more using 'high order' ports like 270005 |
| 05:26.40 | sawgood | in general to the channel |
| 05:27.22 | phonic | if you have a phone registering on port 270005, you have a serious problem. |
| 05:28.00 | sawgood | the remote phone registering using UDP port 27003 is actually working (at least for now) ... I think it stops working after a while (thus my question) |
| 05:28.33 | sawgood | one too many zeros (port 27003) not 270005 (sorry) |
| 05:28.39 | phonic | :) |
| 05:29.10 | phonic | the source port is 27003? |
| 05:29.12 | sshock | I think I found a post to asterisk dev list that illustrates the exact problem: http://lists.digium.com/pipermail/asterisk-dev/2004-July/005217.html |
| 05:29.28 | sawgood | I was reading that "Asterisk" does not 'support' or use port 5061 (yet) as this is reserved for TLS (so I was thinking) that is why some phones go from 5060 to 5062 |
| 05:29.29 | sshock | if so, this bug has sure been around for a long time |
| 05:30.02 | sawgood | phonic: when I do a sip show peers (the Port listed for the remote phone is 27003) |
| 05:30.13 | sawgood | phonic: the phone's GUI states to use 5060 though |
| 05:31.31 | sawgood | I think I might have this narrorwed down even more (with this information): three remote phones registering to the Asterisk box from the same WAN IP (1st phone grabs 5060, 2nd phone grabs 5062, and the 3rd phone grabs 27003) |
| 05:31.31 | phonic | is the remote phones are behind NAT (aka PAT), then the source port could very easily be different. |
| 05:32.02 | [TK]D-Fender | 2004 post? |
| 05:32.04 | kaldemar | sawgood: you're mixing the port that asterisk is listening to and the ports that your phones use. |
| 05:32.07 | [TK]D-Fender | what a waste |
| 05:32.08 | sawgood | right all three of these phones are in a LAN using NAT (PAT) (but two of the three use 'normal' ports |
| 05:32.17 | sshock | what's a waste? |
| 05:32.33 | [TK]D-Fender | [01:29]<sshock>I think I found a post to asterisk dev list that illustrates the exact problem: http://lists.digium.com/pipermail/asterisk-dev/2004-July/005217.html <---- 2004 |
| 05:32.40 | phonic | it wouldn't work if they all had the same source/destination IPs/ports |
| 05:32.47 | sshock | so what exactly is a waste? |
| 05:32.49 | phonic | that's how PAT works |
| 05:33.01 | sawgood | phonic: right (exactly)! I understand that |
| 05:33.28 | sawgood | but why do two of the three phones use a range in the 5060-5062 range and the 3rd grabs something really 'weird' |
| 05:33.45 | phonic | short answer: that's what the router decided to do |
| 05:34.17 | sawgood | So, as long as I have a qualifyfreq= and/or quailfy INVITE SIP messages working, this is ok? |
| 05:34.52 | phonic | as long as the router can keep track of the sessions properly, yeah it should be fine. |
| 05:35.25 | sawgood | I've used a Linksys RV082 for this for a few years now, and for others it has worked just fine ... |
| 05:35.52 | sawgood | I had I think 5 or 6 remote phones working full time behind a RV082 with qualifyfreq=25 set in sip.conf |
| 05:36.16 | sawgood | I never paid attention to the port the Asterisk server was using though (because things worked) |
| 05:36.45 | phonic | can you see the router's translations table? |
| 05:36.55 | phonic | never messed with an RV082 personally |
| 05:36.57 | sawgood | I'm not sure |
| 05:37.12 | sawgood | excellent point though |
| 05:37.37 | sawgood | I just checked another Asterisk box I have with 8 remote phones behind a RV082 (which has been working for about 1 year now) |
| 05:37.57 | sawgood | and most of the remote phones are using 'high order' UDP ports (only 1 phone uses 5060) |
| 05:38.10 | phonic | that's what I would assume to be the norm |
| 05:38.49 | sawgood | So, ruling out SIP port as a problem (here is the concern I am facing) |
| 05:39.07 | sawgood | I have a remote Yealink phone (normally I do not use many Yealink phones as remote phones) ... |
| 05:39.42 | sawgood | The remote Yealink phone works most of the time, but sometimes when I try to call out on the SIP trunk (it fails by simply calling the ring group off all the other phones) |
| 05:40.03 | sawgood | sort of strange (as soon as I reboot the PoE phone) it starts making calls out the SIP trunk again |
| 05:40.48 | sawgood | So, with this Yealink (either it calls out correctly over the SIP trunk, or it fails by calling out to a ring group) |
| 05:41.10 | phonic | it calls the other remote phones directly? |
| 05:41.44 | sawgood | phonic: yes, it calls the other two remote phones (which are on the same LAN together) three remote phones on the same LAN (x101, x102, x105) |
| 05:42.05 | phonic | and you do not see any invites on the * box? |
| 05:42.06 | sawgood | x102 = 5060 x105 = 5062 and x101 = 27003 |
| 05:42.35 | sawgood | Yes, I see the call arrive on the * box (and I see the ring group being called as well) |
| 05:42.56 | phonic | wait. |
| 05:43.21 | phonic | are these 'remote phones' registered with a remote server or a local one? |
| 05:43.24 | sawgood | Perfect timing (I got it to fail) |
| 05:43.38 | sawgood | remote phones going to a remote * box |
| 05:44.07 | phonic | so the * box is sending the invite to the other phones in the ring group? |
| 05:44.31 | sawgood | check this out ...I just make an outbound call (over the SIP trunk) using x101 (I called my Cell phone) |
| 05:44.52 | sawgood | Then, I hung up ... I tried to call from x101 to my cell again, but the call went to the ring group instead |
| 05:44.54 | sawgood | funny! |
| 05:45.22 | kaldemar | sawgood: show a call. that's not funny, probably just a screwed up dialplan. |
| 05:45.37 | sawgood | ok ... let me pastebin this ... |
| 05:45.38 | sshock | ok, now on to more interesting stuff; I have a weird audio problem |
| 05:45.38 | sawgood | brb |
| 05:46.00 | sshock | when I call from 800 to 801, the audio works fine on both sides |
| 05:46.13 | sshock | but when I call the other direction, there's no audio at all for anyone |
| 05:46.25 | sshock | my firewall doesn't show any dropped packets |
| 05:46.44 | kaldemar | sb end |
| 05:48.00 | kaldemar | sshock: care to share a description of the network setup regarding your asterisk box, 800 and 801? |
| 05:48.18 | sshock | kaldemar: sure, I pasted it earlier but I'll paste it again |
| 05:48.37 | sshock | http://pastebin.com/jEHswh8j |
| 05:49.01 | sshock | one user is connected via a softphone on my laptop; the other user is on a Nokia device |
| 05:51.05 | kaldemar | sshock: that's not a description of the network setup, that's a part of your dialplan. |
| 05:51.07 | sshock | either device can connect to exten 600 for echo which works fine |
| 05:51.15 | sshock | oh, sorry |
| 05:51.26 | sshock | you want parts of sip.conf? |
| 05:51.48 | sshock | at the moment both devices are inside LAN |
| 05:52.47 | sshock | it's weird I can call from my laptop to nokia just fine, but the other direction there is no audio |
| 05:52.49 | kaldemar | sip.conf will be helpful but that isn't what i asked for. the network setup, how are the devices connected? asterisk box and the clients in the same LAN? |
| 05:53.01 | sshock | yes, they are all on the same LAN |
| 05:53.33 | sshock | laptop = 192.168.8.105; nokia = 192.168.8.113; * = 192.168.8.1 |
| 05:53.35 | sawgood | sshock: no VLANs on your switch? |
| 05:53.40 | sshock | nope |
| 05:53.57 | sawgood | what type of phone is a Nokia? |
| 05:53.57 | sshock | maybe it is just a problem with my nokia device? |
| 05:54.00 | kaldemar | sshock: then pastebin your sip.conf and a failed call with sip debug. |
| 05:54.01 | sshock | N810 |
| 05:54.12 | sshock | the call doesn't fail |
| 05:54.12 | sawgood | that is a SIP end point? |
| 05:54.41 | sshock | yeah, nokia registers to * as "tammy" and laptop registers to it as "phillip" user |
| 05:54.42 | kaldemar | sshock: if you have no audio, i guess it is failing somehow. |
| 05:55.04 | sshock | what do you mean exactly by "sip debug" ? |
| 05:55.17 | kaldemar | sshock: you know the asterisk CLI? |
| 05:55.46 | sshock | yes |
| 05:55.51 | sshock | asterisk -r |
| 05:56.00 | kaldemar | sshock: attach to it, enable sip debug with command "sip set debug on" and make a call. pastebin the output. |
| 05:56.10 | sshock | ok, thanks; will try now... |
| 05:58.24 | kaldemar | also enable verbosity if not already enabled. core set verbose 10 is the command. |
| 05:58.39 | sshock | ok |
| 06:00.09 | kaldemar | i'm out in 2 minutes but someone else will then have atleast some kind of a chance to tell you what the problem is. |
| 06:04.28 | sshock | ok |
| 06:05.04 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 06:05.28 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 06:06.59 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:07.53 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 06:12.53 | sshock | (fyi, I'm trying to look at these debug logs myself first to see if I can see the problem...) |
| 06:35.02 | sshock | ok, I give up; anyone hanging around that wants to look at this? |
| 06:37.18 | ChannelZ | you aren't the first person to come in here who can't get their Nokia to work right. It makes me wonder. |
| 06:37.23 | *** join/#asterisk michaelhiatt (~michaelhi@ppp59-167-184-100.static.internode.on.net) |
| 06:43.49 | sshock | yeah, I'm starting to wonder if maybe the Nokia just has issues, though I know it does work with my gizmo account |
| 06:43.59 | sshock | perhaps it is something weird like it only works when there is a STUN server |
| 06:48.09 | sshock | I still haven't figured out my other problem though, where * is sending a 407 Proxy Authentication Required after a call failed |
| 06:48.40 | *** join/#asterisk gnarf (~gnarf@home.gnarf.net) |
| 06:49.11 | gnarf | having problems with a SIP peer putting a call on hold and then being unable to retrieve the call... 1.6.2.13 |
| 06:49.24 | gnarf | unhold/hold commands seem --- iffy? |
| 06:49.29 | ChannelZ | That proxy auth is the problem I remember from the last person |
| 06:49.29 | gnarf | its a cisco phone |
| 06:50.11 | ChannelZ | problems how |
| 06:50.20 | sshock | another person that wasn't me? |
| 06:50.57 | kaldemar | sshock: just pastebin the output, you'll be more likely to get answers. |
| 06:52.04 | gnarf | like right now i have a call on hold... and if i unhold it, music still plays, and no audio is transferred |
| 06:53.14 | sshock | kaldemar: ok, I'll start with the no audio when calling from nokia to laptop: http://pastebin.com/5Vj7x38D |
| 06:55.44 | *** join/#asterisk karthik (~raghu@122.183.17.90) |
| 06:56.53 | sshock | I found some posts online that seemed to indicate this is a known problem with Nokia N810/N900, and the workaround is you have to use a STUN server even though in this scenario obviously you shouldn't need one. |
| 06:57.08 | sshock | so anyway, don't spend too much time looking at it, because that is probably the problem |
| 06:58.00 | ChannelZ | sshock: I don't think that other person was you. |
| 06:58.18 | ChannelZ | Unless you had a different nick. A female sounding one, Kelly or something maybe |
| 06:58.45 | ChannelZ | gnarf: does the console show anything occuring when you try to resume the call? |
| 06:58.57 | sshock | oh, ok; good to know I'm not the only one with issues with Nokia |
| 06:59.00 | gnarf | ChannelZ: occasionally, but no |
| 06:59.01 | kaldemar | sshock: maybe one of your clients doesn't like re-invites. or direct media. see "MEDIA HANDLING": http://svn.digium.com/svn/asterisk/tags/1.4.21/configs/sip.conf.sample |
| 06:59.14 | gnarf | sometimes this works, most of the time it doesnt |
| 06:59.23 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 06:59.26 | schmidts | good morning |
| 06:59.41 | sshock | kaldemar: I didn't notice any REINVITE messages in those logs... |
| 07:00.06 | kaldemar | sshock: that's because there is no message by the name REINVITE. just INVITE. |
| 07:00.21 | sshock | oh, ok |
| 07:00.40 | sshock | ok, let me try setting canreinvite=no |
| 07:03.17 | sshock | holy freakin' cow it worked! |
| 07:06.13 | ChannelZ | so the phone isn't paying attention to reinvites right... or something is lying and it's not able to connect where it's being told to |
| 07:06.27 | ChannelZ | is your phone and asterisk all on the same physical network? |
| 07:06.29 | sshock | I guess |
| 07:06.35 | sshock | stupid phone |
| 07:06.44 | sshock | at the moment, yes |
| 07:06.48 | ChannelZ | hmm |
| 07:07.05 | gnarf | now i cant put calls on hold |
| 07:07.07 | sshock | somewhat of a bummer to have to disallow reinvite |
| 07:07.10 | gnarf | its not starting moh at least |
| 07:07.19 | gnarf | seems the hold/unhold is getting ignored sometimes |
| 07:07.23 | sshock | but I can probably live with it, since most the time I'll be receiving calls here at home |
| 07:07.53 | ChannelZ | oh wait I'm mixing you and gnarf up |
| 07:08.00 | ChannelZ | your problems that is |
| 07:08.18 | sshock | oh, sorry |
| 07:08.27 | sshock | feel free to ignore me |
| 07:08.33 | sshock | thanks kaldemar for the suggestion |
| 07:08.56 | sshock | I may have disabled reinvites anyways, just because I'm thinking of recording all phone calls |
| 07:08.58 | ChannelZ | heh no I'm just doing other things at the same time and not paying attention |
| 07:12.15 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:13.49 | gnarf | ChannelZ: canreinvite=no is set |
| 07:14.34 | ChannelZ | is this SIP? |
| 07:14.41 | ChannelZ | that you're using on your cisco |
| 07:15.01 | *** join/#asterisk jkroon (~jkroon@dsl-242-8-144.telkomadsl.co.za) |
| 07:15.18 | schmidts | gnarf are you using 1.6 cause its not canreinvite anymore its called directmedia |
| 07:15.29 | gnarf | same setting? |
| 07:15.30 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 07:15.47 | ChannelZ | yeah |
| 07:16.32 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.177.194.233) |
| 07:16.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.194.233) |
| 07:17.32 | sshock | ok, here's a log for my next problem, where * is sending 407 Proxy Authentication Required instead of just 503 Service Unavailable: http://pastebin.com/AGK5CmEZ |
| 07:17.35 | *** join/#asterisk mpe_ (~mpe@94.127.49.1) |
| 07:17.57 | sshock | this is when I try to call an extension and the other phone isn't connected |
| 07:19.25 | schmidts | sshock * acts like it should do |
| 07:19.42 | sshock | it is supposed to do that? |
| 07:20.23 | schmidts | yes only if you set your sip account to insecure=no i think |
| 07:20.27 | schmidts | mom i will have a look |
| 07:21.24 | schmidts | insecure=invite |
| 07:22.25 | sshock | hmm, but I don't get it; the documentation says if autofallthrough is set (which it is), then if it runs out of rules it will terminate the call with BUSY, CONGESTION, or HANGUP |
| 07:22.46 | sshock | it doesn't say anything about prompting you for authentication (which why would it need to do that anyways? I've already registered with it) |
| 07:23.25 | kaldemar | sshock: autofallthrough has nothing to do with SIP authentication. the 407 happens before the call even enters your dialplan. |
| 07:23.41 | sshock | oh |
| 07:23.59 | sshock | alright, let me try insecure=invite, but then I still need some more explanation |
| 07:24.13 | kaldemar | registering is just a way to let asterisk know where the peer is. it must authenticate every call anyway. |
| 07:25.28 | sshock | ok, that didn't help |
| 07:25.52 | schmidts | then try insecure=port,invite |
| 07:26.10 | sshock | hmm, so if it must authenticate every call, shouldn't my client do that automatically without popping up a dialog to prompt me for credentials? |
| 07:26.20 | schmidts | aaah moment |
| 07:26.21 | sshock | schmidts: yeah, I just thought of that |
| 07:26.26 | sshock | but it didn't help either |
| 07:26.43 | schmidts | you can try insecure=very cause it should work in your 1.4 version |
| 07:26.47 | sshock | I can call people just fine, without any prompts for password |
| 07:27.11 | sshock | it's only when I try to call an extension that is not available because the other phone is not connected |
| 07:27.25 | schmidts | the prompt for password is only your client, this has nothing to do with asterisk |
| 07:28.10 | sshock | schmidts: insecure=very gives a warning about it being deprecated and to use insecure=invite,port instead |
| 07:28.11 | sshock | schmidts: it does have to do with * because * is the one telling my client it needs to authenticate |
| 07:28.12 | sshock | 407 Proxy Authentication Required |
| 07:29.51 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 07:30.08 | sshock | kaldemar: you are wrong about it not going into the dialplan |
| 07:30.12 | gnarf | any other thoughts, other than canreinvite / directmedia = no for the hold issue? |
| 07:30.21 | sshock | I added a Verbose() line in there and it shows it |
| 07:30.25 | ChannelZ | the last time I saw SIP debugs of this, * tells the phone to authenticate but then the phone never does. |
| 07:30.31 | kaldemar | sshock: read what i said again. |
| 07:30.35 | ChannelZ | so it tries another invite, which Asterisk then refuses. |
| 07:31.03 | kaldemar | sshock: read the "before" part twice to make sure. |
| 07:31.04 | ChannelZ | or it does a REGISTER but without auth (again). I don't remember exactly. |
| 07:31.15 | sshock | kaldemar: so are you saying it does both a 407 and the dialplan? |
| 07:31.45 | schmidts | sshock the 407 has nothing to do with the dialplan its just the sip part which says, i dont know you please tell me who you are |
| 07:32.00 | schmidts | after asterisk recognize your credentials then it try to do the dialplan |
| 07:32.03 | sshock | schmidts: but why would it be sending that though? |
| 07:32.12 | kaldemar | sshock: the server sends a 407 to the client to let the client know that it must authenticate. |
| 07:32.27 | schmidts | gnarf which version do you use? |
| 07:32.49 | kaldemar | sshock: asterisk sends it because you haven't told it otherwise. |
| 07:33.07 | sshock | but it's not doing it all the time; only when I dial this other phone that isn't connected |
| 07:33.21 | kaldemar | sshock: it's supposed to do it for every call. |
| 07:33.36 | sshock | ok, let me do a debug of a normal call here... |
| 07:33.47 | kaldemar | sshock: every call must the authenticated unless you tell asterisk otherwise. |
| 07:34.02 | kaldemar | sshock: don't bother. |
| 07:34.18 | sshock | but it doesn't make sense |
| 07:34.43 | sshock | if it does it all the time, why does my client only pop up a dialog prompting me for authentication in this one situation? |
| 07:34.45 | doolittlework | hi there i am having problems with my snom phone extension keys, if i program a function key as an extension(to monitor) the light flashes when incomming call to that extension, and stays on if user is busy on a call, for some reason this has stoped working, any tips on where to start looking for problem? |
| 07:35.21 | doolittlework | Someone sugested hints, do i need to incorporate hints into the dialplan? |
| 07:35.32 | gnarf | schmidts: 1.6.2.13 |
| 07:36.35 | schmidts | doolittlework i suggested it yesterday, and yes normaly you need hints like this exten => hint,1,SIP/userxyz |
| 07:36.37 | kaldemar | sshock: because your client is buggy? |
| 07:37.19 | doolittlework | schmidts: why was this working before without having to add in extensions.conf |
| 07:37.34 | schmidts | doolittlework dont know, what have you changed? |
| 07:37.40 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:37.48 | doolittlework | nothing schmidts |
| 07:37.52 | sshock | kaldemar: who knows, maybe twinkle is buggy yeah |
| 07:38.28 | schmidts | doolittlework :D my 3 year old child told me that everytime ;) |
| 07:38.55 | sshock | anyone else have twinkle want to test it? |
| 07:38.55 | schmidts | gnarf sorry i havent got your problem (login too late) can you explain what exactly are your problems |
| 07:38.56 | doolittlework | schmidts: true but honest daddy i did not do it it was jimmy |
| 07:39.31 | gnarf | so, sometimes (as in not always, but seems to be a lot more lately) pressing hold/unhold on my cisco 7960 doesn't seem to actually register with asterisk |
| 07:39.32 | schmidts | doolittework can you show me your config for this key |
| 07:39.49 | gnarf | doesn't start (or stop) music on hold |
| 07:39.50 | *** join/#asterisk oej (~olle@fw01d.snowmen.se) |
| 07:40.06 | doolittlework | set as extension 200@192.168.1.1:5060 |
| 07:40.34 | schmidts | gnarf have you allready took a sip debug? |
| 07:41.04 | gnarf | http://pastebin.me/04575282ca5265d1777e3b66cd33cea2 |
| 07:41.07 | schmidts | doolittlework and 192.168.1.1 is your *, right? |
| 07:41.43 | sshock | my guess is it's probably the ordering; if 407 comes after the 503, it causes this bug in twinkle |
| 07:42.02 | sshock | stupid twinkle |
| 07:42.31 | schmidts | gnarf mom i have to look if sending an invite with no ip in the sdp is the right way to start MOH |
| 07:43.12 | doolittlework | schmidts:yes |
| 07:44.46 | *** join/#asterisk oej (~olle@fw01d.snowmen.se) |
| 07:45.05 | schmidts | doolittlework i hate snoms at all they do stupid things all the time so its sometimes hard to trace if asterisk does wrong or just the phone |
| 07:45.06 | *** join/#asterisk michaelhiatt (~anonymous@ppp59-167-184-100.static.internode.on.net) |
| 07:46.22 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 07:46.32 | schmidts | gnarf do you see any warning in your log? |
| 07:46.42 | gnarf | schmidts: nope, and no notice of holding either |
| 07:47.16 | schmidts | gnarf strange ;) |
| 07:47.51 | gnarf | and here is a history of the call id |
| 07:47.51 | gnarf | http://pastebin.me/b28564106a3cb3688b84b5e3d8e30b28 |
| 07:50.31 | schmidts | gnarf thats many reinvites in there, not so good at all |
| 07:52.08 | gnarf | ive put the call on hold / unhold about 20 times prolly |
| 07:52.31 | schmidts | ok thats why ;) |
| 07:52.56 | sshock | so all the issues I've dealt with so far have been bugs... |
| 07:53.01 | sshock | http://pastebin.com/3mPrKBYr |
| 07:55.30 | *** join/#asterisk jmpiloq (~jmpiloq@static-87-79-94-28.netcologne.de) |
| 07:55.45 | *** join/#asterisk oej (~olle@fw01d.snowmen.se) |
| 07:56.51 | Kevin` | is the first one actually a bug according to the protocol? the cause is obvious, dealing with connectionless traffic, but should it be fixed there? |
| 07:57.26 | sshock | Kevin`: I'm not sure |
| 07:57.36 | sshock | perhaps it should keep track of some state so it can be smarter |
| 07:58.44 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 07:59.21 | Kevin` | you could also work around that with a nat rule btw, but it would be far more ugly than picking an ip |
| 08:01.15 | *** join/#asterisk bjornts (~Adium@it010225.klientdrift.uib.no) |
| 08:02.00 | schmidts | gnarf i am still not sure if your cisco does something wrong or this is a bug in asterisk |
| 08:02.20 | sshock | yeah |
| 08:04.02 | gnarf | http://pastebin.me/ab716a46d643669e73a0b09544eaf399 |
| 08:04.06 | gnarf | check this paste out |
| 08:04.15 | gnarf | the first hold/unhold doesn't do shit |
| 08:04.20 | gnarf | the second one starts MOH |
| 08:04.27 | gnarf | but unhold doesn't stop MOH |
| 08:04.33 | gnarf | and the third does nothing |
| 08:06.38 | schmidts | i dont see what going wrong in here cause i dont know the moh handling good. |
| 08:06.57 | schmidts | could you please open an issue with this log |
| 08:07.18 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 08:07.31 | stix | Hi guys. Can I timestamp what's happening in the CLI? |
| 08:10.01 | schmidts | stix start asterisk with -T parameter |
| 08:10.26 | stix | oh great :) |
| 08:11.09 | schmidts | asterisk -h has some very nice options to use, have a look ;) |
| 08:15.09 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:17.18 | gnarf | schmidts: https://issues.asterisk.org/view.php?id=18160 |
| 08:17.23 | gnarf | seem a good enough explaination? |
| 08:18.43 | gnarf | this is so strange |
| 08:18.51 | gnarf | i have no clue how to look any further |
| 08:19.02 | gnarf | although interesting enough, this particular phone is having problems registering too |
| 08:19.28 | *** join/#asterisk joobie (~joobie@CPE-138-217-158-153.lns6.lon.bigpond.net.au) |
| 08:20.04 | joobie | hey guys.. WMI indicator not appearing on a polycom 320.. works on other phones.. configs are teh same as the other phones.. any tips on trying to debug this? |
| 08:26.32 | *** join/#asterisk coppice (~chatzilla@m121-202-49-28.smartone-vodafone.com) |
| 08:28.15 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 08:28.43 | gnarf | hrm looking at that sip log |
| 08:29.35 | gnarf | it seems 103 INVITE is not getting the "Peer audio RTP is at port 0.0.0.0:16756" line |
| 08:32.27 | gnarf | hrm |
| 08:32.46 | gnarf | it seems 103 INVITE just replys like nothing changed |
| 08:32.52 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 08:33.15 | gnarf | the 105 INVITE detects RTP/audio descriptions before replying |
| 08:33.47 | schmidts | gnarf the 105 invite doesnt recognize the change of ip, thats could be the problem |
| 08:34.02 | gnarf | doesn't ? |
| 08:34.14 | gnarf | you mean the 103 INVITE doesn't? |
| 08:35.25 | schmidts | in 103 there is an ip in the c= row which describes the ip where to send the rtp stream |
| 08:35.38 | schmidts | in 104 its set to c=0.0.0.0 which starts Moh |
| 08:35.49 | schmidts | but in 105 the sdp information is not recognized |
| 08:35.54 | gnarf | 105 started the MOH |
| 08:35.56 | gnarf | 103 didn't |
| 08:36.14 | schmidts | 104 start the moh |
| 08:36.39 | gnarf | <PROTECTED> |
| 08:36.45 | schmidts | sorry you are right ;) |
| 08:36.47 | gnarf | or between the 105 OK and the 105 ACK |
| 08:37.05 | gnarf | 104 was a "resume" |
| 08:37.13 | gnarf | 103 and 105 are the comparisons |
| 08:37.48 | gnarf | 105 seems to detect the RTP/audio/capabilities/Peer audio change between the request and the trying |
| 08:37.50 | gnarf | where 103 doesn't |
| 08:40.06 | schmidts | ok could you also add another sip trace with verbose output and the start of the call, cause its important if something has changed in between the original invite and the first hold |
| 08:40.06 | schmidts | i thought 103 is the first invite which starts the call |
| 08:40.06 | gnarf | k |
| 08:40.06 | gnarf | hold on |
| 08:47.17 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 08:49.48 | gnarf | http://pastebin.me/61ae84fe4e17c85a92eb8ad326f00f95 |
| 08:49.53 | gnarf | note the 105 still starts MOH |
| 08:49.56 | gnarf | and the 103 doesnt |
| 08:52.01 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:53.52 | schmidts | i see |
| 08:54.05 | schmidts | i will try this out how it normaly works |
| 08:57.34 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 08:58.51 | *** join/#asterisk kmdm (~kenny@unaffiliated/kmdm) |
| 08:59.21 | kmdm | Hi all, in 1.6.2 - what's the best way to try and troubleshoot: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available |
| 08:59.40 | *** join/#asterisk ranjan (~ranjan@122.172.115.225) |
| 08:59.42 | kmdm | (when 'odbc show' shows the database as connected) |
| 09:00.31 | ranjan | Hi all, i have an asterisk installation in my internal lan. and i want ekiga to connect to that asterisk server from another network. which port should i forward for that? |
| 09:00.40 | schmidts | gnarf hm my cisco spa phone send an a=sendonly header which starts Moh and not c=0.0.0.0 |
| 09:01.23 | gnarf | schmidts: which version of cisco software? |
| 09:01.57 | schmidts | its a cisco SPA the old linksys family ;) |
| 09:02.02 | gnarf | ah |
| 09:02.09 | gnarf | im workin with 7960s |
| 09:02.25 | schmidts | ;) |
| 09:03.16 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 09:04.25 | nextime | hello all. I'm experiencing some trubbles with dahdi and asterisk from when i've upgraded from zaptel to dahdi ( i'm using for both latest debian sid version ) |
| 09:04.51 | nextime | even trying to compile from latest tarball the issue is the same |
| 09:04.59 | schmidts | gnarf i am sorry i cant reproduce this with my phones, maybe there is an option in the sip part of your config to send this a header |
| 09:05.29 | nextime | my te410p card won't work anymore, pri show spans say Provisioned, Down, Active |
| 09:05.49 | dandre | hello |
| 09:05.54 | nextime | with the old asterisk 1.4 and zaptel all was working like a charm |
| 09:06.07 | nextime | is there any know issue about that? |
| 09:06.17 | creativx | how sensitive is asterisk to non-smp kernels? |
| 09:08.43 | dandre | how can I get the status of a dahdi fxo channel? |
| 09:08.43 | dandre | For an fxs I can get it with dahdi show channel but this command doesn't show whether the line is ringing, inuse, ... |
| 09:08.57 | dandre | from the menager interface |
| 09:13.19 | schmidts | gnarf you could try adding this ignoresdpversion=yes to your sip conf |
| 09:13.51 | gnarf | schmidts: i just upgraded the firmwared on the phone to the newest i could find |
| 09:13.58 | gnarf | can't seem to repro the issue now |
| 09:14.06 | schmidts | have a look at the sip debug |
| 09:14.34 | gnarf | a=sendonly |
| 09:14.56 | *** join/#asterisk Tim_Toady (~moi@79.103.14.105.dsl.dyn.forthnet.gr) |
| 09:15.07 | schmidts | ok then everything is fine ;) |
| 09:15.12 | schmidts | i will close your issue |
| 09:18.22 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 09:25.10 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 09:32.10 | ranjan | Hi all, |
| 09:32.27 | ranjan | to which port does ekiga connect with asterisk? |
| 09:32.36 | henk | 5060 for sip usually... |
| 09:33.01 | ranjan | henk, but when i use netstat -antlp i dont see a 5060 port |
| 09:33.28 | *** join/#asterisk hipitihop (~denis@202.153.71.87) |
| 09:33.28 | ranjan | henk, i see that asterisk listens on port 2000 |
| 09:33.32 | henk | ranjan: on the asterisk server? |
| 09:33.36 | ranjan | henk, yes |
| 09:34.03 | ranjan | henk, i have an asterisk server installed and its in the lan |
| 09:34.16 | henk | ranjan: of course not, usually sip is used via udp, not tcp |
| 09:35.52 | ranjan | henk, how can i connect to it from another network |
| 09:36.17 | ranjan | henk, i am not getting that idea. :( can you help me |
| 09:36.18 | henk | ranjan: i don't understand the question... |
| 09:36.30 | henk | ranjan: yes, your netstat command simply doesn't show udp ports. |
| 09:36.53 | ranjan | henk, see, i am sitting in a network with 192.168.10.X range of IP |
| 09:37.07 | henk | it doesn't matter. read what i said. |
| 09:37.08 | ranjan | henk, and one of my machine 192.168.10.202 is my asterisk server |
| 09:37.46 | ranjan | henk, ok but |
| 09:38.01 | ranjan | i want to connect to it from another internet connection |
| 09:38.32 | henk | ranjan: so that's a new question? the other one about the listening port 5060 is done, right? |
| 09:38.40 | ranjan | and i have used the virtual server in my adsl modem to open port 2000 |
| 09:39.05 | ranjan | henk, ok listening port is ok. |
| 09:39.11 | ranjan | but please help me with this |
| 09:39.16 | henk | omg, surely you haven't... a modem does not close or open ports. |
| 09:39.36 | henk | this would be much simpler if you had a clue about what you're talking about... |
| 09:39.42 | ranjan | henk, its an ADSL Modem/Router |
| 09:40.12 | henk | yeah, and why port 2000? |
| 09:40.18 | ranjan | henk, sorry if i am talking foolish |
| 09:40.36 | ranjan | port 2000 because i saw that asterisk was listening on port 2000 |
| 09:41.01 | henk | so you're guessing? good luck then, not gonna support that. |
| 09:41.20 | ranjan | and i forwarded the port 2000 to the internal asterisk servers port 2000 |
| 09:42.09 | heffer | ranjan: http://www.voip-info.org/wiki/view/port+forwarding |
| 09:42.10 | ranjan | and when i telnet <ip> 2000 in the asterisk console it shows that --starting skinny session from <ip> |
| 09:42.44 | ranjan | henk, heffer so which port should i really forward? is it 5060? |
| 09:43.01 | heffer | ranjan: 5060 and 10000-20000 UDP |
| 09:43.26 | *** join/#asterisk gladier (~gladier@pedobear.molested.me) |
| 09:43.26 | henk | ranjan: yes. you should _really_ get at least a little clue about what you're trying to use there... |
| 09:43.31 | ranjan | heffer, why 10000-20000 |
| 09:43.50 | ranjan | henk, :( |
| 09:44.15 | henk | there are lots of documentation and howto and tutorials and so on about asterisk and sip is the most well-known protocol for voip, so information about it isn't that scarce on the internet either. |
| 09:44.17 | ranjan | henk, i am just learning , and not professional like you, thats why came here for help :( |
| 09:44.19 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 09:44.25 | henk | ranjan: learn by reading. |
| 09:44.33 | ranjan | henk, k |
| 09:45.05 | heffer | ranjan: 10000-20000 is used for signalling |
| 09:45.14 | henk | ranjan: talk to people if you have a problem with what you read. first read, then _think_, then ask. don't be one of the dumasses skipping the first two steps please... |
| 09:45.30 | ranjan | heffer, ok thank you heffer |
| 09:45.55 | ranjan | henk, ok thank you henk, i will start reading. |
| 09:46.03 | henk | thanks |
| 09:46.27 | gladier | hey guys - i have two sip trunks "friends" on an Asterisk 1.4.36 box ... depending on the order i have them in my sip.conf i get a chan_sip.c:9155 check_auth: username mismatch, have <trunk1-new>, digest has <trunk2-new>; chan_sip.c:15157 handle_request_invite: Failed to authenticate user "0404242679" <sip:0404XXXXXX@114.X.XX.X> |
| 09:48.02 | gladier | for inbound calls |
| 09:49.54 | *** join/#asterisk mpe_ (~mpe@gate.ipvision.dk) |
| 09:50.16 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 09:51.27 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 09:51.28 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:52.03 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 09:54.22 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:01.02 | gladier | looks to be this issue https://issues.asterisk.org/print_bug_page.php?bug_id=9678 |
| 10:01.12 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 10:06.20 | *** part/#asterisk dlirit (~lirant@80.74.100.10) |
| 10:11.39 | gladier | fixed it... just turned on insecure :) |
| 10:11.47 | *** part/#asterisk gladier (~gladier@pedobear.molested.me) |
| 10:13.03 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 10:15.55 | nicola_pav | when i run sip show channels |
| 10:16.13 | nicola_pav | i see many channels stuck on INVITE |
| 10:16.30 | nicola_pav | and all from the same IP which is setup in the freepbx as a trunk |
| 10:16.59 | nicola_pav | 16.192.64.1 413 7c4654e247a 00102/00000 0x0 (nothing) No Init: INVITE |
| 10:17.05 | nicola_pav | any hint? |
| 10:17.22 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:23.13 | *** join/#asterisk cybermick (~michaelhi@ppp59-167-184-100.static.internode.on.net) |
| 10:32.22 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 10:33.52 | *** join/#asterisk skrusty (~skrusty@93-97-20-22.zone5.bethere.co.uk) |
| 10:34.01 | skrusty | morning all |
| 10:34.25 | *** part/#asterisk kmdm (~kenny@unaffiliated/kmdm) |
| 10:40.25 | creativx | hm |
| 10:40.47 | *** join/#asterisk wierdo (~jimmy@212.25.51.150) |
| 10:40.51 | creativx | why does playback suddenly not find localized files any more? /lib/asterisk/sounds/<countrycode>/file.gsm exists. |
| 10:41.56 | creativx | have not changed indications.conf |
| 10:43.14 | kaldemar | sound files are in /var/lib... by default. is your asterisk.conf varlibdir pointing to the right place? |
| 10:43.59 | creativx | yeah |
| 10:44.11 | creativx | i tried moving the file up one level, then it worked |
| 10:44.14 | creativx | wicked |
| 10:47.54 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 10:50.38 | *** join/#asterisk Arnoz (~Arnoz@91.183.59.105) |
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| 11:24.07 | creativx | oh joy i wonder how much shit will break when going from 1.2 to 1.6 |
| 11:24.08 | creativx | :D |
| 11:31.25 | schmidts | creative like everything :D |
| 11:32.16 | schmidts | i have tried it 10 times, after several patches and there is still no end of trying and testing |
| 11:33.28 | creativx | hehe |
| 11:33.34 | creativx | i think i will start with 1 phone.. |
| 11:34.41 | creativx | our old 1.2 box did not enjoy going p2v, so now our sip audio is choppy like hell with even the smallest io/cpu utilization of the vm |
| 11:34.50 | skrusty | anyone here had any issues with SPA5xx series phones behind NAT? |
| 11:35.02 | skrusty | MOH seems to drop the call :s |
| 11:35.11 | skrusty | but not with 4xx series |
| 11:36.32 | skrusty | odd one i know (well, nothing is odd with NAT!) |
| 11:43.54 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 11:45.25 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:46.17 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 11:54.04 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 11:59.56 | *** join/#asterisk xoveruk (~xover@193.220.59.2) |
| 11:59.58 | xoveruk | hi all |
| 12:00.11 | xoveruk | how can i see if a SIP user has registered with asterisk? |
| 12:00.28 | xoveruk | sip show users? |
| 12:00.33 | schmidts | sip show peer xyz |
| 12:00.53 | xoveruk | what is the difference between a user and a peer? |
| 12:00.58 | schmidts | or you can also use sip show peers but this will give you the whole list |
| 12:02.24 | xoveruk | schmidts when i do this i can see extensions which have been created, but how can i tell if they are live or working? |
| 12:03.21 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 12:03.38 | *** join/#asterisk tvollmer (~tvollmer@p579F8530.dip0.t-ipconnect.de) |
| 12:03.40 | *** join/#asterisk wphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:04.06 | tvollmer | Hi can anyone help me to set up an 1und1 VOIP Account in asterisk? |
| 12:04.43 | *** join/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 12:04.52 | WIMPy | tvollmer: Find all the 8 servers and create peers for them. |
| 12:05.27 | schmidts | xoveruk you will see the ip adress and if you have qualify=yes you will see if they are reachable like this: |
| 12:06.08 | schmidts | user19997/user19997 (Unspecified) D N 5060 UNKNOWN |
| 12:06.08 | schmidts | user19998/user19998 213.129.250.216 D N 5064 OK (30 ms) |
| 12:08.49 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 12:08.57 | xoveruk | is quality just a ping test? |
| 12:09.06 | xoveruk | or is that actually using SIP? |
| 12:09.22 | xoveruk | users are complaining that they cannot make SIP calls on the LAN |
| 12:10.03 | schmidts | xoveruk qualify sends by default every 60 seconds a sip option message to the last know ip of every peer |
| 12:10.28 | schmidts | if there is any reply it will save the response time thats the ms value you will see |
| 12:11.58 | xoveruk | so i need this enabled to be able to tell if they are working? |
| 12:13.01 | schmidts | it would make things easier cause you will see on asterisk if one or more phone is not reachable |
| 12:13.54 | WIMPy | xoveruk: They can'y make ot they can't take calls? |
| 12:13.55 | xoveruk | schmidts: can i define 'qualify' globally or do i need to specify this for each sip client? |
| 12:14.16 | WIMPy | Can be done globally. |
| 12:14.20 | xoveruk | where abouts? |
| 12:14.24 | xoveruk | in general? |
| 12:14.36 | schmidts | sip.conf the general section, yes ;9 |
| 12:14.37 | WIMPy | That is the global section, yes. |
| 12:14.42 | schmidts | s/;9/;)/ |
| 12:15.18 | schmidts | but you will have more sip traffic on this, depending on your amount of peers you could have bandwith problems |
| 12:16.45 | xoveruk | I enabled this but its it still shows as unmonitored |
| 12:16.52 | xoveruk | qualify=yes |
| 12:17.20 | schmidts | qualify only starts working if a peer registers to * |
| 12:17.30 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.177.194.233) |
| 12:17.37 | schmidts | mom |
| 12:17.47 | schmidts | no it should show unknown not unmonitored ;) |
| 12:17.50 | drmessano | yes, dear? |
| 12:18.02 | schmidts | mom = moment :D |
| 12:18.21 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.194.233) |
| 12:18.26 | drmessano | You're adopted anyway |
| 12:18.42 | schmidts | ok mum, whats for dinner? |
| 12:19.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:19.44 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 12:19.55 | drmessano | schmidts, grandstream phones and trixbox ISO's, you brat |
| 12:20.51 | schmidts | drmessano wanna kill me? |
| 12:21.30 | xoveruk | schmidts I have restarted asterisk, would that not force register? |
| 12:22.05 | schmidts | xoveruk i would prefer to restart the phones ;) |
| 12:22.12 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 12:23.12 | *** join/#asterisk marcok2010de (~Marco@213.61.61.138) |
| 12:31.05 | xoveruk | i restarted the server and they still show unmonitored |
| 12:31.34 | schmidts | you should restart the phones not the server |
| 12:31.39 | [TK]D-Fender | No |
| 12:31.47 | [TK]D-Fender | Unmonitored has NOTHING to do with REGISTRATION |
| 12:31.52 | [TK]D-Fender | CRAZY PEOPLE |
| 12:31.59 | creativx | [TK]D-Fender. crazy is as crazy does. |
| 12:32.11 | xoveruk | [TK]D-Fender: what is the story? |
| 12:32.12 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 12:32.23 | [TK]D-Fender | creativx: Crazy has being doing a LOT |
| 12:32.36 | schmidts | aaahh sorry missed your unmonitored, i think you still was talking about force registration ;) |
| 12:32.40 | [TK]D-Fender | xoveruk: that says QUALIFY IS OFF |
| 12:32.58 | xoveruk | I set qualify=on in the general section of sip.conf |
| 12:33.11 | xoveruk | it does not work, do i have to enter it for each extension? |
| 12:33.14 | schmidts | qualify=on or qualify=yes? |
| 12:33.15 | creativx | [TK]D-Fender: its fun :) |
| 12:33.46 | xoveruk | [TK]D-Fender: I need to find out if the phones are working |
| 12:34.07 | schmidts | xoveruk you can also make a database show SIP |
| 12:34.11 | [TK]D-Fender | xoveruk: USE THEM. |
| 12:34.26 | schmidts | if the phones are register you should see the ip and port in the astdb too |
| 12:34.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:34.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:34.46 | xoveruk | I am not in the same office |
| 12:35.00 | [TK]D-Fender | xoveruk: call them |
| 12:35.03 | xoveruk | schmidts: how do i access that? |
| 12:35.19 | schmidts | in the asterisk cli: "database show SIP/Registry" |
| 12:35.27 | schmidts | without the " |
| 12:35.49 | *** join/#asterisk mpe_ (~mpe@94.127.49.1) |
| 12:37.02 | xoveruk | schmidts: I am running that on my asterisk and it doesnt not show anything, yet the phones are working |
| 12:37.02 | *** join/#asterisk aleokos (~quassel@mail.moslift.ru) |
| 12:38.11 | xoveruk | I rebooted my phone and nothing came up in there |
| 12:38.43 | schmidts | is your phone registered? |
| 12:39.18 | [TK]D-Fender | [08:35]<schmidts>in the asterisk cli: "database show SIP/Registry" <- WRONG |
| 12:39.51 | schmidts | why??? |
| 12:40.18 | xoveruk | TK, whatever, its nit picking, who cares. |
| 12:40.20 | [TK]D-Fender | schmidts: Because that has nothing to do with phones registering to your server |
| 12:40.32 | xoveruk | ah right |
| 12:40.44 | [TK]D-Fender | xoveruk: Not nit- picking. |
| 12:41.13 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:41.16 | xoveruk | yeah, thought you meant smething else |
| 12:41.17 | xoveruk | sip show registry: Show SIP registration status (when Asterisk registers as a client to a SIP Proxy) |
| 12:41.21 | schmidts | [TK]D-Fender> and where do you think will asterisk save the ip and port of a registration?? |
| 12:41.51 | xoveruk | Are you saying there is not way of seeing if a phone is registered with asterisk unless you call it? |
| 12:41.54 | [TK]D-Fender | schmidts: go look in "sip show peer [peer]" |
| 12:42.40 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:43.10 | schmidts | [TK]D-Fender yes but you can also see this in astdb |
| 12:43.18 | xoveruk | is contact, the important line? |
| 12:43.35 | xoveruk | what is this astdb? |
| 12:43.45 | schmidts | astdb is the internal asterisk database |
| 12:44.15 | schmidts | Reg. Contact is the field you have to look in sip show peer |
| 12:47.26 | schmidts | [TK]D-Fender saving ip and port into astdb was the issue why russell bryant wrote this astdb patch |
| 12:51.52 | *** join/#asterisk suprstar (~suprstar@216.56.88.4) |
| 12:53.50 | suprstar | i want to disable the recording of voicemails for a given extension, temporarily, like if the user is on vacation or something. maxmsg=0 in the options is invalid, any ideas short of making an agi wrapper function? |
| 12:55.01 | suprstar | there's a deprecated option outgoing=yes that would be perfect, but it's not in 1.6.... |
| 12:57.14 | [TK]D-Fender | suprstar: What would happen instead? |
| 13:00.56 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 13:04.16 | xoveruk | in answer to my original question how can i tell from using the CLI if SIP calls on the LAN are working? |
| 13:04.24 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 13:04.36 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 13:05.36 | Naikrovek | well this is interesting |
| 13:05.42 | wphillips | suprstar: Couldn't you simply remove the Voicemail() application entry from the extension for the duration? |
| 13:05.52 | Naikrovek | apparently there are Java application servers that support SIP now. |
| 13:06.03 | Naikrovek | RSA 7 supports developing SIP servlets |
| 13:06.09 | Naikrovek | and they do mean Session Initiation Protocol |
| 13:06.55 | [TK]D-Fender | xoveruk: LOOK at them |
| 13:07.28 | xoveruk | I told you already, they are HERE |
| 13:07.48 | [TK]D-Fender | xoveruk: what is "here"? And why does it matter? |
| 13:13.20 | xoveruk | here is your face |
| 13:13.32 | xoveruk | I cannot see you face so I cannot see if the your face is connected. |
| 13:13.45 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 13:14.47 | xoveruk | I have to ask if the phones are working, but wanted to see if from the CLI I can tell if they are without being in the same place as the phones. |
| 13:19.18 | dandre | I have two configurations one in asterisk 1.4 and another 1.6.2 |
| 13:19.19 | dandre | the files are roughly equivalent. |
| 13:19.29 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:19.50 | dandre | on the 1.6.2, the commande sip show registration doesn't show anything |
| 13:20.08 | dandre | and on the 1.4 it shows as expected |
| 13:20.13 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:20.23 | dandre | is there any known issue? |
| 13:22.07 | schmidts | dandre you mean sip show registry right? |
| 13:22.23 | dandre | yes sorry |
| 13:24.57 | [TK]D-Fender | xoveruk: working means using. So go use them. |
| 13:32.59 | [TK]D-Fender | BAI BAI |
| 13:34.22 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 13:34.24 | Katty | mew. |
| 13:36.16 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 13:38.55 | FlashDeluxe | hi! Ive got a problem, i want to run asterisk 1.6 with DAHDI 2.2.1 and junghanns duoBRI but dahdi_cfg -vv says, that there is no channel to configure, so in the system.conf there is no channel configured, but dahdi is installed correctly. If I modprobe wcb4xxp and start dahdi with /etc/init.de/dahdi start it says |
| 13:38.57 | FlashDeluxe | Loading DAHDI hardware modules: |
| 13:38.58 | FlashDeluxe | <PROTECTED> |
| 13:39.00 | FlashDeluxe | driver should be 'qozap' but is actually 'wcb4xxp' |
| 13:39.01 | FlashDeluxe | Running dahdi_cfg: done. |
| 13:39.09 | Naikrovek | no pasting please |
| 13:39.13 | Naikrovek | ~pb |
| 13:39.13 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:39.15 | Naikrovek | in the future |
| 13:39.29 | FlashDeluxe | sorry, thhought i was under three lines |
| 13:39.36 | Naikrovek | it's not a line thing |
| 13:39.40 | Naikrovek | it's a pasting-in-general thing |
| 13:40.10 | FlashDeluxe | ok, sorry, next time i will pastebin it ;) |
| 13:40.28 | FlashDeluxe | nevertheless, qozap doesn`t exist on my system and i don`t really know, which driver i have to load |
| 13:45.10 | *** join/#asterisk _LoneCrow (~me@industrial.wavedirect.net) |
| 13:46.36 | *** join/#asterisk thevishy (~thevishy@122.175.83.48) |
| 13:46.39 | thevishy | what is the use of asterisk ? |
| 13:46.58 | thevishy | i just heard about this sowtare from a friend |
| 13:47.22 | schmidts | asterisk is an open source telephone system and PBX |
| 13:47.37 | FlashDeluxe | http://lmgtfy.com/?q=asterisk |
| 13:49.13 | FlashDeluxe | and if i load wcb4xx i get red alarms on both controllers :/ |
| 13:50.25 | thevishy | telephone system and PBX ? |
| 13:51.19 | thevishy | can these corporate people take one phone connection and split to many ? |
| 13:51.43 | schmidts | thevishy yes |
| 13:51.49 | thevishy | I see |
| 13:52.08 | thevishy | so PBX is something that facilitates this |
| 13:52.27 | Naikrovek | google will help you more than we can |
| 13:52.34 | Naikrovek | google PBX, google Asterisk |
| 13:52.37 | Naikrovek | read |
| 13:52.38 | Naikrovek | read |
| 13:52.39 | Naikrovek | . |
| 13:52.43 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-jdmqmladgfkbclit) |
| 13:52.46 | thevishy | Yeah am doing that lol |
| 13:54.32 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 13:54.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:57.57 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:57.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:58.16 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 13:59.12 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:04.47 | *** join/#asterisk jeffik (~chatzilla@69-196-165-181.dsl.teksavvy.com) |
| 14:12.13 | *** part/#asterisk tvollmer (~tvollmer@p579F8530.dip0.t-ipconnect.de) |
| 14:13.54 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 14:15.19 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 14:16.29 | *** join/#asterisk TJ_stamm (~TJ@69.197.107.254) |
| 14:17.45 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 14:18.12 | nicola_pav | hello. I havea sterisk 1.4.30. i need to compile it with tcp |
| 14:18.22 | nicola_pav | any hint? where to start from? |
| 14:18.35 | nicola_pav | re-compile |
| 14:19.17 | leifmadsen | nicola_pav: asterisk 1.4.x does not support TCP |
| 14:19.18 | [TK]D-Fender | nicola_pav: 1.4 doesn't support TCP. 1.6.X does |
| 14:19.25 | leifmadsen | use Asterisk 1.6.2.x or later |
| 14:19.40 | *** join/#asterisk FFForever (FFForever@unaffiliated/ffforever) |
| 14:19.44 | FFForever | Hi ya |
| 14:19.53 | nicola_pav | there is any patch to do it? |
| 14:20.08 | FFForever | Would anyone know of a cheap/good DID provider for screwing around with asterisk? |
| 14:20.26 | leifmadsen | nicola_pav: no, use 1.6.2.x or later |
| 14:20.35 | leifmadsen | FFForever: Google? |
| 14:20.42 | leifmadsen | FFForever: are you in the USA? |
| 14:20.52 | schmidts | leifmadsen google voice? ;) |
| 14:20.55 | leifmadsen | schmidts: yes :) |
| 14:21.04 | FFForever | leifmadsen, I am in the us. |
| 14:21.13 | leifmadsen | FFForever: then use Asterisk 1.8 and use Google Voice |
| 14:21.44 | FFForever | leifmadsen, I am looking for something that isn't gv.... |
| 14:21.45 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 14:22.04 | thevishy | can anyone tell me a application of asterisk ... just real life |
| 14:22.18 | *** join/#asterisk xtuple_pclark (~pclark@static-76-161-157-20.dsl.cavtel.net) |
| 14:22.28 | leifmadsen | why? |
| 14:22.44 | leifmadsen | thevishy: queue? auto-attendant? voicemail system? |
| 14:22.48 | leifmadsen | thevishy: conference system? |
| 14:22.56 | bougyman | thevishy: making a phone call, receiving a phone call. |
| 14:22.57 | FFForever | leifmadsen, because I already have a gv account and I don't really want to toy with it and miss a call |
| 14:23.00 | leifmadsen | thevishy: war dialer? |
| 14:23.05 | xtuple_pclark | hey all |
| 14:23.11 | schmidts | leifmadsen that was a bad one :D |
| 14:23.33 | thevishy | Right this is somthing like what I would need for example if I have a telephone company right ? |
| 14:23.49 | FFForever | People who operate war dialers have a special place in hell reserved for them. |
| 14:24.11 | xtuple_pclark | looking for info on screen pops in QT script using the asterisk API |
| 14:25.30 | bougyman | if you have a telephone company and don't know what you could use a telephony engine for, i'm dubious. |
| 14:26.28 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:26.29 | thevishy | No i dont have a telephone co , but am saying these are things needed by a telephone co: |
| 14:26.39 | *** part/#asterisk xtuple_pclark (~pclark@static-76-161-157-20.dsl.cavtel.net) |
| 14:27.41 | thevishy | it interacts with various devices , does it identify devices using IP ? |
| 14:30.21 | kaldemar | if you want to. |
| 14:30.36 | thevishy | I see |
| 14:31.32 | leifmadsen | thevishy: it can be used anywhere from a home system to running an ITSP |
| 14:31.47 | [TK]D-Fender | theHave you ever seen a "phone system" in ANY company? |
| 14:31.50 | [TK]D-Fender | thevishy: ^ |
| 14:32.11 | [TK]D-Fender | thevishy: You know.. where they have a recepionist, and lots of users and line and call each other, have voicemail, etc? |
| 14:32.22 | [TK]D-Fender | thevishy: THAT is the kind of thing yuo can make with * |
| 14:32.53 | thevishy | Right no I have not seen I am trying to understand these things. I worked on Unix support for Nokia Siemens . They build NMS and also integrate network elements with with NMS basically. |
| 14:33.02 | [TK]D-Fender | thevishy: A system where you can make menus for your calls, do complicated processing, etc |
| 14:33.18 | thevishy | Right I am getting some sense |
| 14:33.31 | [TK]D-Fender | thevishy: How can you not have seen a "phone system" / PBX / Key System EVER? |
| 14:33.47 | [TK]D-Fender | thethat's what * is used to build |
| 14:33.52 | thevishy | Lets say I have a pizza shop I can implement a server and sorta do "logic" on it to distribute the calls etc |
| 14:33.53 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:34.31 | thevishy | I have seen Phone system as in old exchanges where u plug in out |
| 14:34.53 | thevishy | is that what you mean? the one where operators sit and connect calls as per the request |
| 14:35.05 | leifmadsen | thevishy: basically think of any phone system at any company, and Asterisk can do that, but better |
| 14:35.28 | leifmadsen | thevishy: asterisk contains lots of functionality that can be built into anything you want. It's a telephony toolkit. |
| 14:35.38 | thevishy | I see |
| 14:35.47 | [TK]D-Fender | thevishy: Yes, that is just transferring calls. very basic. not even something you really "configure so much |
| 14:36.09 | thevishy | but the question , when I call a company then does the call gets picked by the SERVER ? |
| 14:36.12 | leifmadsen | you can build queues, you can control how those queues operate, you can determine what techonology you use to deliver or accept the calls (IAX, SIP, PSTN, etc...), you can determine who to ring, and when, and with what skills. |
| 14:36.21 | leifmadsen | thevishy: yes, Asterisk would answer the call. |
| 14:36.32 | thevishy | I see |
| 14:36.38 | leifmadsen | think of asterisk as like a voicemail on super drugs that can do anything you want. |
| 14:36.53 | thevishy | lol ok |
| 14:36.57 | leifmadsen | if you wanted, Asterisk could be used simply as a voicemail system |
| 14:37.15 | [TK]D-Fender | thevishy: Picture where when you call in it says "Helo" and asks you what department you want to speak with. Maybe one of those options leads to calling a specific phone. Another option might ring MULTIPLE phones at the same time. Another option jsut lets them leave you a message. Another option plays them a list of services your comapny offers, etc. This is called an IVR. More things... |
| 14:37.16 | [TK]D-Fender | ...you set up with * |
| 14:37.47 | leifmadsen | it could answer the call, play a message, record a message, then save it on the HD, plus email it to you, store it on your IMAP server and integrate with your calendar so the voicemail system would answer automatically when you were marked as busy. |
| 14:38.07 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:38.29 | thevishy | Yes poweful things . I suppose something like yellow pages |
| 14:38.44 | leifmadsen | I don't understand the relationship... |
| 14:38.56 | leifmadsen | yellow pages is a book to me |
| 14:38.59 | thevishy | I mean where I call a tele phone yellow page . |
| 14:39.09 | leifmadsen | you could build such a system, yes. |
| 14:39.27 | thevishy | Right |
| 14:39.59 | thevishy | so basically do you connect your PSTN to server ? |
| 14:40.07 | leifmadsen | yes |
| 14:40.19 | thevishy | Right so instaed of a Receive I keep a Server ! |
| 14:40.26 | thevishy | Receiver* |
| 14:40.27 | leifmadsen | either through a physical connection to the PSTN like with an Analog or Digital line, or virtually through an ITSP over SIP or IAX |
| 14:40.36 | thevishy | ah |
| 14:40.44 | leifmadsen | from the viewpoint of the phone company, Asterisk is the telephony. |
| 14:40.46 | leifmadsen | telephone* |
| 14:40.55 | leifmadsen | a programmable telephone |
| 14:40.56 | thevishy | Yes |
| 14:41.07 | leifmadsen | see http://astbook.asteriskdocs.org |
| 14:41.09 | thevishy | that makes sense |
| 14:41.11 | thevishy | okay |
| 14:41.14 | leifmadsen | some reading is likely in order |
| 14:41.25 | thevishy | thanks will read it up |
| 14:41.36 | thevishy | i am also seeing the cool intro @ http://astbook.asteriskdocs.org |
| 14:41.42 | *** join/#asterisk rafael-ec (~rafael@200.25.197.108) |
| 14:41.42 | thevishy | i mean asteriks homepage video |
| 14:42.46 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:48.57 | FlashDeluxe | hi! can anybody tell me what module a junghanns duoBri card does need to work with dahdi and asterisk? |
| 14:49.25 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 14:50.50 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:51.22 | creativx | heh im amazed |
| 14:51.33 | E-bola | Do anybody here use any type of cluster or HA with asterisk? |
| 14:51.37 | creativx | reinstall of fc4, haggard old distro.. and 1.2 from source.. copied over all config files == works |
| 14:52.01 | E-bola | Im wondering how to best make my asterisk installation highly available |
| 14:52.12 | p3nguin | If the configs are 1.2 compatible. why wouldn't they work? |
| 14:52.31 | creativx | p3nguin: shit is usually not _that_ simple! |
| 14:52.49 | p3nguin | It should be. |
| 14:54.10 | creativx | well I am glad it was :> |
| 14:55.38 | schmidts | E-bola that depends on what service you have |
| 14:56.12 | schmidts | E-bola most just put a sip proxy in front and let asterisk only do call handling but no registration |
| 15:00.08 | skrusty | so, anyone using SPA500 series phones with asterisk behind NAT? :) |
| 15:01.29 | skrusty | E-bola: try Ultra Monkey |
| 15:01.32 | *** join/#asterisk GameGamer43|Mac (~GameGamer@65.27.76.78) |
| 15:03.54 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 15:04.11 | schmidts | skrusty yes |
| 15:05.12 | schmidts | skrusty what you want to know about the spas? |
| 15:05.24 | skrusty | i ahve a few behind NAT, and * is on a public IP |
| 15:05.34 | skrusty | problem is, whenever you put someone on hold, MOH start, but then call drops |
| 15:05.37 | skrusty | any ideas why? |
| 15:05.44 | skrusty | doesn't happen with 400 series |
| 15:06.08 | schmidts | strange, never had this problem. do you use nat keep alive? |
| 15:06.11 | skrusty | rtp audio in both directions is fine on both phones |
| 15:06.13 | skrusty | yeah |
| 15:06.21 | skrusty | qualify is also on at *'s end |
| 15:06.29 | schmidts | hmm no i havent seen this |
| 15:06.30 | skrusty | sip keep-alive turned on at the phone |
| 15:06.42 | skrusty | odd i know... tried with a few routers now, same problem |
| 15:06.54 | schmidts | after which time the call drops? |
| 15:07.01 | skrusty | about 3 seconds |
| 15:07.19 | schmidts | do you have rtptimeout or rtpholdtimeout set? |
| 15:07.25 | skrusty | nope.... |
| 15:07.34 | skrusty | hmm, sip.conf option? |
| 15:07.38 | schmidts | yes |
| 15:07.47 | skrusty | both are not set |
| 15:07.54 | schmidts | ok |
| 15:07.59 | skrusty | im using realtime by the way |
| 15:08.03 | schmidts | no idea ;) sorry |
| 15:08.11 | skrusty | boo :D |
| 15:08.11 | schmidts | i gtg now, cya all ;) |
| 15:08.15 | skrusty | cheers anyway |
| 15:08.52 | drmessano | skrusty, firmware bug? |
| 15:09.41 | skrusty | unsure, spoken with Cisco, they seem to think not, but have escelated it |
| 15:09.43 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 15:09.50 | drmessano | I mean, the moh starts, but then the call drops.. is there an MOH server setting in there you've accidentally populated? |
| 15:10.16 | skrusty | phone is base config (reset to factory settings) and only set proxy, user and nat akeep-alive |
| 15:10.43 | skrusty | keep-alive even |
| 15:11.09 | skrusty | one of those that's got me a bit stumped... esp' as the 400 series doesn't have the same issue |
| 15:11.29 | E-bola | skrusty: last time i looked at ultra monkey (bout 2 years ago) it hadnt been updates in years |
| 15:11.48 | skrusty | E-bola: tbh, neither have I, but thought it was worth a shout :) |
| 15:11.58 | E-bola | schmidts: ya i know alot use openser to handle the HA part, i was just hoping i wouldnt have to learn how to use openser and get by with asterisk |
| 15:12.09 | skrusty | hehe :) |
| 15:12.44 | E-bola | I sort of already have HA via our virtualization platform, i would just prefer to have it "inside" the servers instaid of outside |
| 15:12.54 | E-bola | preferebly something running active/active |
| 15:13.02 | skrusty | yeah |
| 15:13.22 | skrusty | i've been looking at something like that, hence why i mentioned ultra monkey as it seemed like a good way to do it |
| 15:13.27 | drmessano | Lots of guys have clustered asterisk in some way.. multiple frontend boxes, multiple SQL backends for realtime, etc |
| 15:13.36 | drmessano | It doable, if you have a little imagination |
| 15:13.37 | skrusty | as use we sip realtime, the peer info is stored in a shared resource |
| 15:14.40 | E-bola | drmessano: i havent been able to find any up2date blogs or stories about how they did it |
| 15:15.04 | E-bola | and since its quite clear you can do it in a ton of different way's i would love to read about smoebody's experiences with specific solutions before trying to re-invent it all my self |
| 15:15.17 | *** part/#asterisk rafael-ec (~rafael@200.25.197.108) |
| 15:16.31 | drmessano | E-bola, just because no one has put it all out on your plate doesn't mean you can't make a trip to the buffet. Google for realtime.. that's your backend. see how multiple SQL backends are being implemented with Asterisk.. then check maybe DUNDI peering the boxes or some alternatives |
| 15:17.44 | drmessano | E-bola, this is Asterisk.. there is no "right way" to do anything.. there are lots of WRONG ways to implement the pieces, but following those rules and putting them together your way, you can make anything |
| 15:17.53 | [TK]D-Fender | drmessano: DUNDi isn't really for HA. Its pretty much a slighlty more protocol-agnostic E.164 |
| 15:18.23 | E-bola | drmessano: I always prefer to read about how others did something instaid of testing it myself first |
| 15:18.56 | drmessano | [TK]D-Fender, I know leif had a howto a while back for clustering asterisk using DUNDI.. maybe there is a better way. Someone else could suggest something here too |
| 15:19.51 | [TK]D-Fender | drmessano: A better way with the wrong tool? probably not :) Lets say "unintendedly hackable in a semblenece of the desired outcome" :p |
| 15:20.26 | [TK]D-Fender | drmessano: You can clump taht into my bin along with "Sure * supports SLA, it even has SLA.CONF!" |
| 15:20.33 | E-bola | It does seem though that people are using dundi for HA |
| 15:20.43 | E-bola | It sure wouldnt be my first try either.... |
| 15:21.05 | [TK]D-Fender | E-bola: some mix between "because someone wrote a doc and I jsut followed it", "can't think of another way", etc |
| 15:21.26 | [TK]D-Fender | E-bola: And "Don't want to have to learn SER too" |
| 15:22.22 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:23.40 | E-bola | I guess the best thing to do would probably be to go the SER way, since thats seems to be what most are doing, including my 2 providers |
| 15:23.45 | *** join/#asterisk the_weard (~arthur@41-132-172-16.dsl.mweb.co.za) |
| 15:23.50 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:24.56 | [TK]D-Fender | E-bola: Some would take that as a "hint" ;) |
| 15:25.23 | E-bola | Lazyness rules the world :) |
| 15:27.27 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:27.31 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:30.31 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 15:31.25 | skrusty | lo [TK]D-Fender by the way, long time no speak :) |
| 15:38.56 | *** join/#asterisk mcargile (~mikec@rrcs-97-76-33-146.se.biz.rr.com) |
| 15:40.39 | mcargile | In asterisk 1.6 how do I query a local channel with the 'b' option set to find out what the actual channel is behind it? |
| 15:41.23 | [TK]D-Fender | mcargile: "behind it"? |
| 15:41.37 | titter | Can anyone explain what 3RD_T2_TIMEOUT is during an incoming fax with res_fax? I am only getting this from one sender. |
| 15:41.39 | mcargile | * A 'b' option has been added which causes chan_local to return the actual channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. |
| 15:41.58 | [TK]D-Fender | mcargile: If it still exists, then it has the same "bridged to" reference as any other channel |
| 15:42.10 | coppice | titter: the third one is when the FAX engine gives up |
| 15:42.25 | [TK]D-Fender | mcargile: Odd.. never heard of that |
| 15:42.36 | mcargile | That is from the change log |
| 15:43.24 | titter | coppice: Any idea on how I should debug this? Try a lower baud rate maybe? |
| 15:44.25 | coppice | its a myth that lower bit rates help with these problems |
| 15:44.36 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:45.28 | titter | coppice: What would you suggest? |
| 15:51.52 | mcargile | [TK]D-Fender: I guess my question is how do I query a channel from an AGI script in the manor this change is referring to. |
| 15:52.38 | [TK]D-Fender | mcargile: can you point me to docs on it? |
| 15:53.33 | mcargile | [TK]D-Fender: all I can point you to is the Change log referring to it. If I could find the docs I could figure it out. |
| 15:54.34 | geemee | I have an IAX trunk that works OK however calling from site 1 to site 2 the Caller Display is listed as Device XXX. Dialing the other way the Caller display is OK. Where can I start in troubleshooting this? |
| 15:56.05 | *** join/#asterisk leif[mobile] (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:56.05 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 15:56.17 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 15:58.45 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 16:04.37 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 16:06.15 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 16:06.21 | cusco | hello |
| 16:06.27 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:06.49 | leif[mobile] | hi |
| 16:07.38 | cusco | in a ael dialplan, at the end of extension t, I have a goto context,h,1 so it hangs up. on the hangup it performs a mysql query inser to db the hunged up call info... now it seems that calls going trough t are hanged up twice |
| 16:07.47 | cusco | hi leif, on a mobile phone, are you? |
| 16:08.04 | leif[mobile] | yepistles |
| 16:08.06 | leif[mobile] | Yep... |
| 16:08.23 | leif[mobile] | Auotcomplete ftl |
| 16:08.45 | thehar | hehe |
| 16:09.26 | leif[mobile] | Cusco doing _. Anywhere? That's what will happen |
| 16:09.48 | cusco | ow.. is it? |
| 16:09.49 | cusco | let me check |
| 16:10.47 | cusco | inside the context there is no _ anywyere... |
| 16:10.50 | cusco | gah.. he left |
| 16:11.27 | cusco | but the original dialplan that gets the call has _NXXXXXXXX |
| 16:11.53 | cusco | inside it there is a goto context,s,1 |
| 16:11.58 | cusco | in that context there is no _ |
| 16:19.59 | *** join/#asterisk cjk (~cjk@85.93.217.128) |
| 16:20.43 | cjk | hi, i am using disa and i would like to disconnect the calls and return to disa when i hit *0, unfortunately i disconnect the 2 legs of the calls when entering *0. any ideas are welcome |
| 16:22.18 | *** part/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 16:25.34 | cusco | I heard somewere that disa can be insecure.. |
| 16:25.58 | cusco | maybe you should have *0 transfer the cal to some Local/dialplan |
| 16:33.22 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:34.50 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 16:40.32 | [TK]D-Fender | cusco: What else do your Rice Crispies say to you? |
| 16:41.05 | [TK]D-Fender | cusco: It's as secure as the dialplan that lets you call it and the dialplan you point it to. |
| 16:41.20 | [TK]D-Fender | cusco: 1 wrong INCLUDE is the exact same risk |
| 16:42.26 | drmessano | [TK]D-Fender, same thing with Ward @ co at NerdVittles telling everyone anonymous SIP insecure |
| 16:43.04 | drmessano | or & |
| 16:43.10 | [TK]D-Fender | drmessano: People need to get dictionaries |
| 16:43.38 | [TK]D-Fender | drmessano: Insecure is when you can get to things you're not SUPPOSED TO be able to |
| 16:43.54 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:45.28 | drmessano | Sad thing is, they bundles FreePBX in PIAF, and FreePBX has always ensured the dialplan in regard to SIP guests has remained secure. So either anonymous SIP is insecure in Asterisk, they don't trust the way FPBX implements it, or they aren't confident in how they're implementing it |
| 16:45.33 | drmessano | bundle* |
| 16:46.06 | drmessano | It's like selling Quiches and creating mass hysteria in the egg market |
| 16:46.43 | [TK]D-Fender | drmessano: Anonymous SIP packets don't magically land in contexts other than the one specified in [general]. |
| 16:47.11 | drmessano | Absolutely |
| 16:47.24 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 16:48.23 | drmessano | I can't say I have seen anyone in a GUI or non-GUI environment implement a package or bundle asterisk with anonymous SIP incorrectly configured |
| 16:50.05 | drmessano | But yeah, I think the same bozos perpetuated the "DISA is not secure" thing too |
| 16:50.06 | [TK]D-Fender | drmessano: FreePBX could have done a MUCH better job of separating access with very little effort. |
| 16:50.30 | drmessano | How so? |
| 16:51.22 | [TK]D-Fender | drmessano: in general. all devices are treated the same really. |
| 16:51.44 | Naikrovek | devices and users are kinda treated as one but they have a mode where you can separate them |
| 16:51.51 | drmessano | Yes |
| 16:51.53 | Naikrovek | that makes a lot of things possible that otherwise aren't |
| 16:51.54 | [TK]D-Fender | drmessano: would be simple to separate up functionality into contexts and build a "Class Of Service" selection screen |
| 16:52.30 | drmessano | Ah |
| 16:52.31 | [TK]D-Fender | drmessano: make a class that says, yes, this and that trunk, these internal features", and pick the class on the device level |
| 16:52.49 | Naikrovek | that would be nice |
| 16:52.57 | [TK]D-Fender | drmessano: You could then in turn make DISA point to a specific class, etc... |
| 16:53.04 | [TK]D-Fender | My work GUI has this and more. |
| 16:53.15 | Naikrovek | don't suppose you could share that |
| 16:53.20 | [TK]D-Fender | Which... I'd love to dispose of of course :p |
| 16:53.23 | Naikrovek | lol |
| 16:54.03 | [TK]D-Fender | Naikrovek: its closed and it's what got me * in here. That was paramount. Now that it's in, whatever happens, happens, and its all commodity hardware and worst case == best case ;) |
| 16:54.27 | Naikrovek | you write it? |
| 16:54.43 | drmessano | Something along the lines of nested includes, where rather than just removing/adding lines of dialplan, you're creating new contexts for DISA and where you dump your anonymous calls |
| 16:54.43 | [TK]D-Fender | Naikrovek: No, that's why I want to get rid of it :) |
| 16:54.48 | Naikrovek | ah hehe |
| 16:54.54 | drmessano | ? |
| 16:54.58 | Naikrovek | so what kind of gui would you write, if any |
| 16:55.18 | Naikrovek | client/server, webui, dialplan only, other? |
| 16:55.40 | [TK]D-Fender | Naikrovek: I'd aim for a "framework", more than a "one-stop-shop", with sample common bits so you could deploy as a complete solution like FPBX, but make the rest pluggable |
| 16:55.54 | Naikrovek | i was thinking the same |
| 16:56.17 | Naikrovek | make an API that does the work with Asterisk, then hook into it however you please |
| 16:56.20 | Naikrovek | plugin support |
| 16:56.48 | Naikrovek | i wrote something very similar to this while at State Farm, but instead of managing the phone system it managed our entire ticket tracking system |
| 16:57.01 | [TK]D-Fender | Naikrovek: not THAT abstract. it would build configs, but for instance the core dialing logic wouldn't be anywhere as locked away. |
| 16:57.02 | Naikrovek | we wound up using it to tie other ticket tracking systems and monitoring solutions together |
| 16:57.55 | Naikrovek | have you seen yahoo pipes? |
| 16:58.01 | drmessano | HISS |
| 16:58.04 | Naikrovek | lol |
| 16:58.06 | drmessano | You said the Y word |
| 16:58.37 | Naikrovek | yahoo pipes has a nice gui where you pass RSS feeds through it and you can graphically do 'if' statements and control flow and munge the feed into what you want |
| 16:58.45 | Naikrovek | then you subscribe to the modified feed |
| 16:58.52 | Naikrovek | something simliar could be done with a dialplan gui |
| 16:59.14 | Naikrovek | and one could have every bit as much control over the dialplan |
| 16:59.18 | drmessano | "Learn How to Build a Pipe in Just a Few Minutes" <--- Yahoo should build their company out of pipes then |
| 16:59.21 | Naikrovek | i used to work for "HISS" |
| 16:59.28 | Naikrovek | a decade ago |
| 16:59.47 | drmessano | Back when Soccer moms roamed free? |
| 16:59.52 | drmessano | God those were the days |
| 17:00.05 | Naikrovek | 2001-2003 in North Sydney, NSW |
| 17:00.10 | Naikrovek | those WERE the days |
| 17:00.20 | ChannelZ | Yes. In the past |
| 17:00.39 | drmessano | Back then, you didn't need AdultFriendFinder with Yahoo around.. |
| 17:00.47 | Naikrovek | lol |
| 17:00.55 | Naikrovek | HISS has turned to reminisence |
| 17:01.25 | drmessano | Now the only soccer moms on yahoo are from indonesia |
| 17:01.32 | Naikrovek | lol |
| 17:01.37 | Naikrovek | god you crack me up sometimes |
| 17:02.41 | drmessano | Ditto.. I would kiss you right now, but I am trying to get QQ to load.. Need to order some beef and broccoli |
| 17:03.40 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 17:04.30 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 17:05.36 | ChannelZ | no kissing in #asterisk |
| 17:06.13 | ChannelZ | that's how rootkits get spread |
| 17:06.20 | Naikrovek | stupid to do during flu season anyway |
| 17:06.31 | Naikrovek | how do i tell centos that my hardware clock is NOT on UTC |
| 17:07.06 | drmessano | Oh, that's realtime memory |
| 17:07.24 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
| 17:07.27 | drmessano | rm -rf /* recursively set the clock to your current timezone |
| 17:07.36 | Naikrovek | heh |
| 17:07.37 | fauxalliance | Naikrovek, use utc.... |
| 17:07.49 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
| 17:07.52 | fauxalliance | i cant find GMT -3.5 most of the time |
| 17:07.58 | ChannelZ | hwlock ? |
| 17:08.00 | Naikrovek | well i'm on a nice hour boundary |
| 17:09.08 | Naikrovek | okay this is weird. hwclock shows 7 hours in the past, 'date' shows 5 hours into the future. |
| 17:09.27 | drmessano | I guess I need to go find a nice 1ft long 1/4" drill bit so I can remove the air restrictor on this nerf gun |
| 17:09.41 | Naikrovek | which model |
| 17:09.42 | ChannelZ | well localtime is just an offset from the hardware clock |
| 17:09.50 | drmessano | Longstrike CS-6 |
| 17:09.57 | fauxalliance | drmessano, don't you have a legitimate potato canon? |
| 17:10.06 | Naikrovek | ChannelZ: both hwclock and 'date' show CDT which is my timezone |
| 17:10.16 | drmessano | My potato weapons were confiscated before the trial, fauxalliance |
| 17:10.39 | ChannelZ | hwlock can show either. It doesn't really know what mode the hardware clock is in. |
| 17:10.43 | Naikrovek | drmessano: can't take it apart to remove the restrictor? my little maverick revolver shoots a bit better but i had to take it apart to get those restrictors out |
| 17:10.45 | fauxalliance | drmessano, i was just warned not to freeze them again... no long arm regisrty in canada yet ;-) |
| 17:10.52 | Naikrovek | ChannelZ: okay thanks |
| 17:11.45 | fauxalliance | smartphones, not just for watching movies at work anymore... |
| 17:13.09 | drmessano | Naikrovek, the longstrike has a reverse plunger in it, so the air restrictor is a second shaft below the barrel.. you pull the whole assembly out and just drill the end, from the inside.. small 1/4" hole |
| 17:13.30 | Naikrovek | let me know the results |
| 17:13.48 | Naikrovek | had a friend once somehow turn a maverick nerf revolver into a fully auto revolver |
| 17:13.58 | Naikrovek | would fire 6 darts in like 0.1 seconds or something. |
| 17:14.05 | Naikrovek | like a little minigun |
| 17:14.28 | Naikrovek | he hooked it up to an air compressor to do it, but i still say it counts |
| 17:15.08 | drmessano | I also need to find a small piece of thinwall PVC, because the spring retainer is about 3/4 of an inch deep.. you can drop a small 1/2" PVC ring in there and compress the spring another 1/2". They claim 100 feet outdoors and 75 indoors after just modding the spring |
| 17:15.35 | Naikrovek | ooh |
| 17:15.47 | ChannelZ | Why would it be more outdoors? |
| 17:16.21 | Naikrovek | tailwind? |
| 17:16.52 | drmessano | I think ANY wind tends to be a little updraft for those little foam darts |
| 17:16.58 | ChannelZ | Guess it depends which way you're shooting |
| 17:17.17 | ChannelZ | hmm |
| 17:17.52 | drmessano | Same as flying one of those balsa wood airplanes outdoors or indoors.. I always thought it was funny they would sink like a rock indoors |
| 17:18.34 | drmessano | I think a little wind is enough of an updraft, even a headwind, to keep it going |
| 17:18.40 | drmessano | *little* |
| 17:19.44 | drmessano | But yeah.. I can get about 25 feet indoors now, and hitting a cat in the face sounds like a fat kid falling on the sidewalk... so I know there something to work with |
| 17:20.07 | drmessano | Not that I was practicing on the cats |
| 17:20.50 | ChannelZ | I won't tell PETA |
| 17:21.17 | drmessano | People Eating Tasty Animals? |
| 17:21.23 | [TK]D-Fender | People for the Eating of Tasy Animals |
| 17:21.25 | [TK]D-Fender | :) |
| 17:21.25 | ChannelZ | yesh |
| 17:22.18 | drmessano | It's ok... by the time PETA gets here, I am gonna have this thing modded.. Then I am ready to board up the windows, bolt the door, and dare the cops |
| 17:22.30 | fauxalliance | i have more worries of the SPCA |
| 17:22.49 | pabelanger | ~book |
| 17:22.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:23.01 | ChannelZ | I think the nerf at out-of-the-box power is enough to make PETA people shriek |
| 17:24.53 | drmessano | Especially when you soak the darts in bald eagle blood? |
| 17:24.58 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:25.08 | drmessano | Wait, are they still endangered? |
| 17:25.21 | ChannelZ | everything is to them |
| 17:25.30 | Katty | hi. |
| 17:25.38 | ChannelZ | OH HAI |
| 17:31.24 | *** join/#asterisk patrick^ (~patrick_@hq.clearcable.ca) |
| 17:32.59 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
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| 17:35.02 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 17:36.14 | Katty | it's a sleepy day |
| 17:36.26 | Kobaz | nap time |
| 17:36.41 | Katty | already had a nap |
| 17:38.30 | Kobaz | so who'se going to astricon |
| 17:40.00 | *** join/#asterisk zoid_ (~awainer@190.2.14.213) |
| 17:42.29 | *** join/#asterisk Tim_Toady (~moi@79.103.14.105.dsl.dyn.forthnet.gr) |
| 17:42.29 | Naikrovek | would love to but can't |
| 17:42.38 | *** join/#asterisk _zoom_ (~user@41.218.39.168) |
| 17:43.40 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 17:44.16 | z4nD4R | hi |
| 17:44.17 | *** join/#asterisk TJ_stamm (~TJ@69.197.107.254) |
| 17:44.47 | z4nD4R | i hope that you can help me |
| 17:45.12 | z4nD4R | i need run my asterisk with sip over tls |
| 17:45.32 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:45.34 | z4nD4R | but i have severals problems |
| 17:46.35 | z4nD4R | asterisk version 1.8 rc3 ( i testet 1.6.2.5 to ) |
| 17:47.39 | z4nD4R | problem is that after client send clientHello message, sever send serverHello with Alert 40 handshake failure |
| 17:47.55 | z4nD4R | SOMEBODY CAN HELP ME? |
| 17:48.10 | Naikrovek | no yelling please |
| 17:48.21 | Naikrovek | you'll get responses if you give us a way to solve your issue |
| 17:48.24 | Naikrovek | for example: |
| 17:48.31 | Naikrovek | 1) tell us what you're trying to do |
| 17:48.38 | Naikrovek | 2) tell us what is actually happening |
| 17:48.52 | Naikrovek | 3) show us some output that shows us the actual errors |
| 17:49.06 | Naikrovek | use a pastebin |
| 17:49.08 | Naikrovek | ~pb |
| 17:49.09 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:49.35 | z4nD4R | 1) i want set sip over tls |
| 17:49.54 | z4nD4R | 2) tls handshake failure |
| 17:50.10 | z4nD4R | 3) w8 for my sip.conf |
| 17:50.39 | Naikrovek | we'll also need a log showing the handshake failure probably |
| 17:51.23 | z4nD4R | http://pastebin.com/61ghFaAY |
| 17:52.30 | z4nD4R | i sniffed this comunication in wireshark, because aterisk cli nothing shows |
| 17:54.43 | Naikrovek | z4nD4R: okay |
| 17:54.48 | Naikrovek | not a lot of TLS users in here I don't think |
| 17:54.58 | Naikrovek | be patient and someone knowledgable will assist you |
| 17:55.24 | rdircio | not me :( |
| 17:56.21 | z4nD4R | Naikrovek: ok... thx |
| 17:56.26 | z4nD4R | i hope.. |
| 18:11.40 | zoid_ | is it possible to customize the cdr fields when using postgres? I would like to add a dialplan variable |
| 18:12.23 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 18:16.29 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 18:16.36 | KavanS | is portaone ~ asterisk? |
| 18:19.32 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:20.17 | *** join/#asterisk mcargile (~mikec@office2.vicidial.com) |
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| 18:28.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:31.58 | *** join/#asterisk Da-Geek (~Da-Geek@87-194-2-213.bethere.co.uk) |
| 18:37.29 | z4nD4R | aterisk + sip over tls... someone??? |
| 18:39.39 | Naikrovek | z4nD4R: handshake problem still? does either side give a reason why the handshake failed? |
| 18:39.43 | *** join/#asterisk PhunTelTek (4cbce85c@gateway/web/freenode/ip.76.188.232.92) |
| 18:40.11 | z4nD4R | yes |
| 18:40.18 | Naikrovek | what is the reason |
| 18:40.21 | z4nD4R | i reade in fcf |
| 18:40.23 | z4nD4R | thah |
| 18:40.35 | Naikrovek | ... huh? |
| 18:40.40 | z4nD4R | w8 |
| 18:40.48 | *** part/#asterisk rdircio (~rdircio@189.137.181.208) |
| 18:40.55 | z4nD4R | 7.4.1.3. Server hello |
| 18:40.56 | z4nD4R | <PROTECTED> |
| 18:40.56 | z4nD4R | <PROTECTED> |
| 18:40.56 | z4nD4R | <PROTECTED> |
| 18:40.56 | z4nD4R | <PROTECTED> |
| 18:40.58 | z4nD4R | <PROTECTED> |
| 18:41.04 | Naikrovek | please don't paste |
| 18:41.07 | Naikrovek | use a pastebin |
| 18:41.08 | z4nD4R | ok |
| 18:41.10 | z4nD4R | sry |
| 18:41.15 | Naikrovek | np |
| 18:41.57 | z4nD4R | i think... this is a problem |
| 18:42.01 | Naikrovek | yes |
| 18:42.56 | Naikrovek | does your work in wireshark tell you what algorithms each side is proposing to use? |
| 18:43.01 | z4nD4R | in second msg ( serverHello ) become the alert handshake failure |
| 18:43.41 | z4nD4R | client use severals algorithm ( cca 19 ) |
| 18:44.04 | z4nD4R | on server side i set tlscipher=RC4-SHA |
| 18:44.20 | z4nD4R | which support client to |
| 18:45.57 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 18:47.14 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:49.13 | skrusty | odd question, but is it possible to have a catchall style context? I want to be able to give sip entries a specific context, but don't want to have to create each one in the extensions.conf, and thus want to know if there is a way of saying 'use this by default' ? |
| 18:50.59 | Letoric | skrusty that's what the context option is for in sip.conf |
| 18:51.35 | [TK]D-Fender | skrusty: What do you mean "create in extensions.conf"? |
| 18:51.43 | [TK]D-Fender | skrusty: that makes no sense... |
| 18:51.49 | skrusty | let me explain |
| 18:51.55 | skrusty | we have an AGI that handles routing |
| 18:52.03 | skrusty | all calls come into the default context |
| 18:52.09 | [TK]D-Fender | skrusty: You point each peer to a context. You don't need to create a UNIQUE context for every device |
| 18:52.15 | skrusty | this works great, except for when a phone issues 302 |
| 18:52.47 | skrusty | if this happens, a local channel is created and our AGI doesn't know which (virtual) group of phones is responsible |
| 18:53.00 | skrusty | is local channel just uses the overall context |
| 18:53.29 | skrusty | but if i could set the contect for each sip entry to something specific for that group, then the context of the local channel would hold all the information i needed |
| 18:53.47 | skrusty | except, i would then have to create an entry in the extensions.conf for each of those contexts |
| 18:53.50 | skrusty | which i dont wnat to do |
| 18:53.59 | skrusty | does that make sense? :/ |
| 18:54.57 | skrusty | [TK]D-Fender: yeah, i understand... but i want to use the context for the peer entry as a way to define a group of phones |
| 18:55.16 | skrusty | but want to only have one real context in extensions.conf from which my AGI is executed |
| 18:57.23 | skrusty | :/ |
| 18:57.26 | [TK]D-Fender | skrusty: what does the context name actually MEAN to you? |
| 18:57.33 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:57.38 | skrusty | a context is just a section of dialplan |
| 18:57.43 | skrusty | something into which calls drop |
| 18:57.51 | skrusty | as i see it |
| 18:58.20 | skrusty | i set the context for a sip peer, and in normal options, the dailed number is fed into that context, and processed |
| 18:58.33 | skrusty | in general, is that about right? |
| 18:59.14 | [TK]D-Fender | skrusty: >-- do you give a shit about the NAME for anything? |
| 18:59.38 | skrusty | no...? |
| 18:59.49 | skrusty | Local channels are what's casuing me to ask this |
| 19:00.17 | skrusty | because local channels get setup in the context set for that sip peer right? |
| 19:00.27 | skrusty | and for me, at the moment, that's just [default] |
| 19:00.52 | skrusty | but it gives me no idea about the device that requested the channel to be setup, which is proving to be an arse |
| 19:01.09 | skrusty | sorry, this must seem converluted! (sp?) |
| 19:01.33 | skrusty | (convoluted) |
| 19:09.40 | skrusty | ok, lets put it this way, if i have set a sip peer to use a context that doesn't exist, is there a way to default to one that does? At the moment you jsut get an error in asterisk and it drops the call. |
| 19:11.00 | [TK]D-Fender | skrusty: Why don't you just point EVERYTHING to [default] and be done with it? |
| 19:11.30 | skrusty | because of local channels, when a phone sets up a 302 and asterisk creates a local channel, the context is set to default. |
| 19:12.01 | skrusty | at this point, my AGi knowns jack about this channel, as there is no information in it about which device created it (i.e. my sip phone) |
| 19:12.22 | skrusty | this is really my problem, and i am just trying to find creative ways around it |
| 19:13.29 | skrusty | my agi needs to know which phone, or at least which group of phones that local channel has been created for, and by default there's no way of knowing |
| 19:14.11 | skrusty | :s |
| 19:15.27 | [TK]D-Fender | skrusty: It should have channel variablesinherited |
| 19:15.34 | skrusty | the channel "Local/10008@default-test-35c7" means nothing, as we may have 30 different devices using the 'extension' 10008 |
| 19:17.47 | skrusty | they dont seem to be |
| 19:18.52 | skrusty | in the initial channel the callerid is "10002_1" which is the peer name, but in the local channel it's "10002", (omitting the '_2') |
| 19:19.28 | [TK]D-Fender | CalleriD != target transfer ext # |
| 19:19.51 | skrusty | no, but i was wondering why they differed betweeen the two |
| 19:23.59 | Letoric | 3 line paste incoming - [TK]D-Fender: Any thoughts on what's happening here? |
| 19:24.01 | Letoric | [Oct 19 14:23:29] -- Executing [aa-off@AutoAttendant:6] GotoIf("SIP/4108-0000020e", "7200 = 3600?20:15") in new stack |
| 19:24.01 | Letoric | [Oct 19 14:23:29] -- Goto (AutoAttendant,aa-off,20) |
| 19:24.01 | Letoric | [Oct 19 14:23:29] -- Executing [aa-off@AutoAttendant:20] Playback("SIP/4108-0000020e", "for&digits/2&hours") in new stack |
| 19:24.29 | Letoric | it's failing the if check, but still going to the success priority |
| 19:24.29 | [TK]D-Fender | Letoric: Yes, someone didn't make a proper EXPRESSION |
| 19:24.49 | Letoric | oh? It looks like most of my others that work, what did I do wrong? |
| 19:25.00 | [TK]D-Fender | Letoric: Not an EXPRESSION. |
| 19:25.39 | *** join/#asterisk TobSnyder (~Tobias@91-64-185-67-dynip.superkabel.de) |
| 19:25.43 | TobSnyder | hi |
| 19:25.47 | Letoric | I've tried == and =, neither seem to work. I'll research more on expressions, just puzzled that it looks the same as my others that DO work |
| 19:27.46 | [TK]D-Fender | Letoric: $[] <----------------- experssion |
| 19:31.11 | TobSnyder | [TK]D-Fender and some other might remember that I had some troubles with DTMF detection last time. The reason was that I had outbound dail options like t or w defined |
| 19:31.41 | TobSnyder | full information on here: http://pastebin.com/D1Cc8Caa |
| 19:33.45 | skrusty | [TK]D-Fender: thanks for the bit about inherited variables, i didn't know about those, my own fault. using them to identify the original device infomation worked. Cheers! :D |
| 19:34.33 | *** part/#asterisk _zoom_ (~user@41.218.39.168) |
| 19:34.34 | [TK]D-Fender | TobSnyder: . the conference system received 07166411 instead of 071641 <------ T & W don't use normal digits by themselves. You need to start with # or * and its timeout based |
| 19:34.48 | [TK]D-Fender | TobSnyder: And stop using DTMF for crap like that |
| 19:38.19 | TobSnyder | [TK]D-Fender: but with T & W defined asterisk will analyze the audio stream what might bring up some more latency in there? |
| 19:38.39 | TobSnyder | concerning "stop using DTMF for crap like that" => can't understand you |
| 19:38.43 | [TK]D-Fender | TobSnyder: There is no functional effect on latency for this. |
| 19:39.00 | [TK]D-Fender | TobSnyder: STOP DOING DTMF TRANSFERS |
| 19:39.41 | TobSnyder | okay |
| 19:39.58 | TobSnyder | I would like but it's not that easy |
| 19:40.32 | TobSnyder | 1. I am having different analog telephone sets, meaning they have different hook flash times |
| 19:41.54 | TobSnyder | 2. I don't want to say this as you will slap me for...(having some free sip softphones like X-Lite that do not have transfer functionality as those features are only available at non-free versions..) |
| 19:42.52 | TobSnyder | 3. how to enable dynamic monitoring when not using DTMF? |
| 19:42.58 | *** join/#asterisk saint_42 (~saint42@c-76-98-53-209.hsd1.pa.comcast.net) |
| 19:44.31 | TobSnyder | all in all I now have disabled dial options like t and w as this is the solution to get dtmf working |
| 19:45.07 | [TK]D-Fender | TobSnyder: ..... on-demand monitor fine... I guess uing a PC to trigger it wouldn't be so hot. the rest... are just crappy choices. Either way, those features should not be set in a way that interferes with basic digits |
| 19:45.20 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:46.23 | *** join/#asterisk JunK-Y (~junky@64.15.77.94) |
| 19:50.31 | TobSnyder | sure...just wondered why the problem occurs at all |
| 19:50.41 | TobSnyder | "The problem is that the DTMF detection takes some time (35-40 ms). During this time the PCM with DTMF is forwarded over the BRI to the far end equipment that  also tries to detect the DTMFs. It may happen that the far end equipment already detected a DTMF and then Asteris sends the same DTMF again." |
| 19:50.50 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:51.45 | z4nD4R | nodoby can help with sip over tls? |
| 19:51.55 | z4nD4R | *nobody |
| 19:52.52 | *** join/#asterisk mgbowman (~mgbowman@78.96.144.26) |
| 19:53.00 | mgbowman | +1 for asteirks! |
| 19:53.04 | mgbowman | *asterisk |
| 19:53.21 | thehar | z4nD4R: use 1.8 RC i hear it supports magic |
| 19:53.45 | z4nD4R | thehar: i have 1.8 rc3 |
| 19:53.50 | thehar | awesome |
| 19:54.10 | z4nD4R | but tls stil not working |
| 19:56.10 | [TK]D-Fender | z4nD4R: And what docs are you foloowing to enable suppotr for it? |
| 19:56.14 | [TK]D-Fender | following* |
| 19:56.33 | z4nD4R | [TK]D-Fender: http://www.voip-info.org/wiki/view/SIP+TLS |
| 19:57.48 | *** part/#asterisk TobSnyder (~Tobias@91-64-185-67-dynip.superkabel.de) |
| 19:57.59 | *** join/#asterisk TobSnyder (~Tobias@91-64-185-67-dynip.superkabel.de) |
| 19:58.35 | TobSnyder | [TK]D-Fender: if that description is correct, isn't it a bug in asterisk then? |
| 19:59.23 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 19:59.36 | [TK]D-Fender | TobSnyder: What description |
| 19:59.40 | TobSnyder | if it analyses the audio stream for detecting dtmf there is no need to send detected dtmf tones back to bri again... |
| 19:59.43 | [TK]D-Fender | z4nD4R: PB your configs |
| 19:59.56 | TobSnyder | [TK]D-Fender: "The problem is that the DTMF detection takes some time (35-40 ms). During this time the PCM with DTMF is forwarded over the BRI to the far end equipment that  also tries to detect the DTMFs. It may happen that the far end equipment already detected a DTMF and then Asteris sends the same DTMF again." |
| 19:59.59 | z4nD4R | [TK]D-Fender: sip.conf? |
| 20:00.19 | [TK]D-Fender | z4nD4R: clearly |
| 20:01.31 | z4nD4R | [TK]D-Fender: what you mean "" |
| 20:01.34 | z4nD4R | [TK]D-Fender: what you mean "PB your configs" |
| 20:03.27 | [TK]D-Fender | ~pb |
| 20:03.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 20:03.29 | [TK]D-Fender | ^^^^^^^^^ |
| 20:03.46 | z4nD4R | w8 |
| 20:04.21 | z4nD4R | [TK]D-Fender: my sip.conf http://pastebin.com/vTREQCM9 |
| 20:05.14 | z4nD4R | [TK]D-Fender: extensions.conf http://pastebin.com/90c4svnj |
| 20:09.21 | z4nD4R | if you whant i cant send you ( pastebin ) tls handshake failur ... snifing with wireshark |
| 20:09.44 | z4nD4R | *can |
| 20:11.41 | mgbowman | can I play a sound two 2 callers once the call is setup |
| 20:14.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:14.55 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 20:18.51 | mgbowman | are there no up to date docs |
| 20:19.11 | mgbowman | voip-info is getting redicuously out of date |
| 20:19.53 | z4nD4R | hmmm tuttorial was writen on asterisk 1.8 |
| 20:23.08 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 20:23.22 | *** join/#asterisk [hC] (~voxter@74.198.148.46) |
| 20:23.41 | heffer | voip-info also looks exceptionally cluttered |
| 20:23.44 | [hC] | anyone know how to get a PAP2 to accept calls if they are called with a destination number (ie Dial(SIP/pap2/5551212) |
| 20:23.58 | JunK-Y | [hC]: are ya coming to astricon this year? |
| 20:24.07 | [hC] | currently they are rejected unless i leave 5551212 off, but based on how its complicated to modify m dialplan in this instance, im looking to see if there's an option to have the pap2 allow that |
| 20:24.47 | [hC] | JunK-Y: nope. it seems to be run very sloppy this year, i was put up on the site to speak without being asked, then took down without being told, and then someone from TMC called to invite me to buy a ticket to come to the show.. so i said fuck it, i'll stay home and make some money instead :) |
| 20:24.58 | [hC] | JunK-Y: the shitty part is i like hanging out with you guys too :P |
| 20:25.12 | p3nguin | [hc]: Since you don't have multiple phone numbers on the PAP2, there's really no reason to send calls to it like that. |
| 20:25.22 | [hC] | JunK-Y: especially when we get you drunk and you harass polycom :) |
| 20:25.41 | JunK-Y | i never been drunk :) |
| 20:25.42 | [hC] | p3nguin: I know, i can work around it in the dialplan, and i know its not 'right' im just seeing if theres a way to get the pap to ignore it, its the easier path in this case, if i can do it. |
| 20:25.47 | [hC] | JunK-Y: hahahahahahahahhaah |
| 20:25.56 | [hC] | JunK-Y: liar!! :P |
| 20:26.10 | JunK-Y | im coming with david (the guy from astricon 2007) this time. |
| 20:26.19 | [TK]D-Fender | [16:25]<JunK-Y>i never been drunk :) <-- would have made a decent excuse for that pic ;) |
| 20:26.29 | p3nguin | [hc]: You wouldn't send other IP phones calls in the same manner, so why do you want to keep sending calls to the PAP2 like this? Fix your dialplan. |
| 20:26.31 | [hC] | JunK-Y: ah cool. im sure it will be a blast, but sadly no i am not coming. |
| 20:26.38 | Qwell | JunK-Y: LIES! I've seen you (very) drunk. :p |
| 20:26.43 | *** join/#asterisk QbY (~kelvin@96.176.19.11) |
| 20:26.53 | JunK-Y | damn, my reputation is done! |
| 20:27.04 | [hC] | p3nguin: sigh. thanks, i guess? heh - i know im not 'supposed' to do that, id just prefer to not mess with the dialplan in this particular case. |
| 20:28.30 | QbY | building a new box with realtime.. keep getting this error.. [Oct 19 16:26:44] NOTICE[23190]: chan_sip.c:21687 handle_request_register: Registration from '<sip:0012D9122326@11.22.33.44>' failed for '55.66.77.88' - No matching peer found -- I've double checked everything, but am apparently missing something |
| 20:28.32 | QbY | suggestions? |
| 20:29.43 | z4nD4R | QbY: sip show users |
| 20:29.46 | z4nD4R | ? |
| 20:31.45 | QbY | z4nD4R: only users showing are the ones defined in sip.conf... nothing from the database |
| 20:32.29 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 20:32.45 | z4nD4R | QbY: you have defined users ini sip.conf ? |
| 20:33.36 | QbY | z4nD4R: there is one test one |
| 20:34.08 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 20:34.44 | QbY | z4nD4R: i've commented it out, sip reload, and still nothing showing |
| 20:36.13 | QbY | realtime show pgsql status is showing its connected |
| 20:37.40 | mgbowman | is there a way to set user specific variables |
| 20:37.49 | p3nguin | yes |
| 20:38.10 | mgbowman | my use case: detect 2 secure endpoints (tls + srtp) |
| 20:38.14 | p3nguin | Use setvar within the definition for the "user." |
| 20:38.30 | mgbowman | and playback a "secure" audio file to both uers |
| 20:42.36 | mgbowman | I know you can do the A(channel-secure) ... but that only announces the called party |
| 20:42.48 | mgbowman | can I not play a sound to both callers |
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| 20:56.18 | mgbowman | can I not use the M(x) macro in the Dial cmd to playback an audio file to both extens? |
| 21:02.56 | QbY | in realtime.. what's the difference between sipusers and sippeers |
| 21:03.07 | Qwell | one is for users, the other is for peers |
| 21:03.16 | Qwell | sip, specifically. |
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| 21:09.06 | QbY | Qwell: A Cisco phone is a .... |
| 21:09.30 | Qwell | terrible phone |
| 21:09.51 | QbY | hehehehe |
| 21:09.54 | QbY | i meant user or peer |
| 21:09.59 | QbY | what about a polycom |
| 21:11.31 | QbY | apparently everything should just be a peer in 1.6 |
| 21:11.33 | QbY | brb |
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| 21:24.25 | carrar | w00t! |
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| 22:14.14 | ANurmi | will any of the polycom Spectra link phones work with a standard asterisk build or do they require more dedicated hardware (besides wifi router)? |
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| 22:48.47 | ANurmi | does anyone have an opinion on the Polycom SoundPoint IP 430 - VoIP phone |
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| 22:55.17 | paulc | ANurmi: A great phone? (I like most of the Polycom gear) |
| 22:58.47 | ANurmi | paulc: One thing I am confused about with the polycom ip 430, is it talks about a POE built into it, do I need any other hardware on the network side? |
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| 23:09.45 | QbY | ODBC Voicemail Storage issue: http://pastebin.org/322029 |
| 23:10.47 | QbY | ignore the dsn names--they are correct in both |
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| 23:28.10 | leifmadsen | QbY: check the table layout and look at the SQL logs to see why the INSERT is failing |
| 23:28.52 | QbY | layout was fine.. ksqo |
| 23:28.55 | leifmadsen | you won't be able to debug much from the Asterisk side because ODBC hides that information. You have to debug it from the PostgreSQL side by looking at (and enabling) the logs |
| 23:28.57 | QbY | have to turn that off for postgres |
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| 23:38.40 | jdoe | how can I set the callerid for outbound calls made by app_followme? |
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