| 00:08.24 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-guxtrczeixyyqfqb) |
| 00:12.46 | jeffspeff | i'm looking to make an IVR that processes payments through authorize.net ... anybody have a good starting point for me? i have no clue on this. |
| 00:14.25 | KNERD | ~book |
| 00:14.26 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:14.33 | jpsharp | Probably the easiest way to go is through an AGI script. |
| 00:18.35 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 00:18.53 | leifmadsen | what is the API like for authorize.net? |
| 00:19.05 | leifmadsen | curl? soap? resp? |
| 00:19.18 | jeffspeff | leifmadsen, no clue |
| 00:19.27 | *** join/#asterisk blindhog (~josh@adsl-75-62-197-176.dsl.rcsntx.sbcglobal.net) |
| 00:20.15 | leifmadsen | looks like there are several APIs, so it depends what you need to do |
| 00:20.19 | leifmadsen | and which API you choose |
| 00:20.41 | leifmadsen | there are also lots of example code from what I see |
| 00:20.52 | leifmadsen | so probably could hack together somethign using an AGI easy enough |
| 00:21.07 | leifmadsen | this is my first time looking at the site, so I suspect you could get the data you need from authorize.net |
| 00:22.07 | jeffspeff | i started to look at the different API's, i also did a quick search through their site for IVR, Asterisk or phone, all to no avail; thought i'd check in here and see if anybody might be working on the same thing. |
| 00:22.15 | jeffspeff | leifmadsen, thanks thogh |
| 00:22.22 | jeffspeff | *though |
| 00:22.33 | leifmadsen | wel the asterisk part is going to be very simple |
| 00:22.41 | leifmadsen | all you're going to be doing is passing data you receive from asterisk tot he api |
| 00:23.09 | leifmadsen | you really just need to develop the ability to integrate somehow with them through a script, then just let the script accept data that is passed by asterisk |
| 00:23.31 | leifmadsen | you'll play a prompt, capture CC data, etc... then just pass that to the AGI to do all the heavy lifting |
| 00:23.34 | jeffspeff | leifmadsen, hopefully it is as simple as you make it sound; i'll toy with it in a test context and see what i can break. lol |
| 00:23.47 | leifmadsen | the asterisk side, will be simple |
| 00:23.56 | leifmadsen | the API and coding for authorize.net, that's the tricky part I suspect |
| 00:24.00 | jeffspeff | hmm, i think i see a macro in the works |
| 00:24.03 | leifmadsen | asterisk won't be the bottleneck though |
| 00:24.32 | jeffspeff | ok |
| 00:25.09 | *** join/#asterisk bmg505 (~leon@196-209-101-151.dynamic.isadsl.co.za) |
| 00:36.45 | paulc | leifmadsen: You do a bunch of stuff with CURL in the dialplan right? Got a quick minute for a boolean answer qusetion? |
| 00:36.58 | leifmadsen | paulc: I can try :) |
| 00:38.14 | paulc | Cheers... Imagine you have 2 web services.. A returns after 2 seconds. B returns after 20 seconds. You have two separate extensions to call each service. Would you expect a call to A to not proceed until an in progress call to B had completed? |
| 00:39.25 | leifmadsen | paulc: ummm... I'm not sure... can you give a better example? |
| 00:39.41 | leifmadsen | do you mean calls to CURL(), or calls in Asterisk? |
| 00:39.46 | leifmadsen | or calls to the web service... ? |
| 00:39.58 | leifmadsen | "call" is a bit vague in this instance :D |
| 00:40.33 | paulc | Yes, sorry - a call to CURL.. dialplan logic for extension 123 uses CURL to hit webservice A which sleeps for 20 seconds then spits something out |
| 00:40.50 | paulc | similarly extension 456 uses CURL to hit webservice B that sleeps for 2 seconds then spits something out |
| 00:41.19 | leifmadsen | I'd not expect to be able to dial extension 456 until 123 returned |
| 00:41.26 | paulc | Call 123, verbose logging shows the call starting.. call 456 a few seconds later.. and it seems that CURL doesn't hit the URL until after the CURL instance in 123 has returned |
| 00:41.32 | leifmadsen | i.e. I'd expect CURL() to act like Dial() |
| 00:41.45 | paulc | But what about on different channels/ |
| 00:41.47 | leifmadsen | ya I'm confused |
| 00:41.56 | paulc | haha sorry, I'm not explaining it well am I |
| 00:41.58 | leifmadsen | I'd expect the calls to be totally independent of each other :) |
| 00:42.10 | paulc | YES - I'd expect that too! Except it's acting like they're not |
| 00:42.11 | leifmadsen | if you have separate asterisk channels using CURL() I'd expect them to not be related |
| 00:42.16 | leifmadsen | o.O |
| 00:42.25 | leifmadsen | I never noticed that when I was using it.... |
| 00:43.17 | paulc | 2 phones on my desk. I pick up one, dial 123, it CURLS to a local webservice that sleeps looooong. While I'm waiting for it to return, pick up the other phone, dial 456, and it seems like that CURL process/instance doesn't start until the first one (on the other, unrelated/separate call!) has returned |
| 00:43.49 | paulc | I was looking for validation that this seemed "odd".. before I go delving into whether this is Asterisk and CURL related, or potentially something with the web server that's being hit |
| 00:46.46 | paulc | I'll play around tomorrow and write it up a bit better for the forum/mailing list. For now, there's a commute followed by beer and pizza :) |
| 00:46.52 | paulc | doffs hat - thanks for your help |
| 00:46.57 | UnixDev | is there some way to store registrations in realtime database? |
| 00:47.19 | leifmadsen | paulc: very good -- I'd try maybe doing the same two calls from two tabs in a web browser |
| 00:47.25 | leifmadsen | UnixDev: yes... |
| 00:47.44 | paulc | leifmadsen: good litmus test, yes! I'll give that a whirl now before I head out - cheers :) |
| 00:47.55 | UnixDev | leifmadsen: where can I find the schema? i don't see it in mysql or postgres contrib/realtime dir |
| 00:47.57 | leifmadsen | doffs hot |
| 00:48.01 | WIMPy | Is there a proxy involved? |
| 00:48.02 | leifmadsen | s/hot/hat |
| 00:48.43 | leifmadsen | WIMPy: also a good point to check |
| 00:49.36 | leifmadsen | UnixDev: I just look here usually: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1465.html#dynamic_realtime |
| 00:49.52 | leifmadsen | it's now 8:49am and I've been working since 7am, so FTN, peas out |
| 01:05.41 | *** join/#asterisk coppice (~chatzilla@m121-202-98-209.smartone.com) |
| 01:32.40 | PipBoy | hates asterisk keypress sensitivity |
| 01:32.42 | PipBoy | >.< |
| 01:33.03 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 01:36.57 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:36.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:08.02 | PipBoy | exten => docheck,n,Wait(1) can someone tell me what the N is for? trying to find a good resource for understanding this stuff |
| 02:09.07 | *** part/#asterisk PipBoy (PipBoy@ip067.216.145.100.nas.net) |
| 02:09.24 | jpsharp | We would tell you, but you disconnected. |
| 02:40.58 | *** join/#asterisk Robzombie (cce48afa@gateway/web/freenode/ip.204.228.138.250) |
| 02:44.49 | *** part/#asterisk Robzombie (cce48afa@gateway/web/freenode/ip.204.228.138.250) |
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| 03:08.34 | bikcmp | s/disconnected/left the room |
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| 03:34.45 | *** join/#asterisk Robzombie (cce48afa@gateway/web/freenode/ip.204.228.138.250) |
| 03:36.04 | Robzombie | Any asterisk guru's around, got a wierd issue cant seem to resolve. |
| 03:39.18 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 03:40.25 | bikcmp | kicks Robzombie |
| 03:42.03 | Robzombie | http://pastebin.com/TwUs5vKk |
| 04:23.03 | UnixDev | is there some way to get FollowMe to play ringing instead of MOH ? |
| 04:24.34 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xyntrbdusztafhju) |
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| 05:58.53 | Jo_Bell | Hi, I have a speex audio coming out of my application. My application users require to get into conference with the third party SIP phones. If no transcoding is needed at my application end, it will be nice. Can I pass my Speex audio data as it is to Asterisk and create a conference with third party SIP softphones? Are the softphones/ SIP devices generally capable of decoding Speex? |
| 06:03.34 | kaldemar | asterisk supports speex. if you have phones that do not, asterisk can do the transcoding for you. |
| 06:09.07 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:11.41 | Jo_Bell | kaldemar: That means, my application can just pass the Speex data after SIP session initiation, using RTP. Is this correct understanding? |
| 06:12.23 | kaldemar | yes. |
| 06:14.22 | Jo_Bell | Kaldemar: Whether each external third-party phone supports Speex or not, will that also have to be set as certain Asterisk parameters at the respective servers where the external phones are registered? |
| 06:19.12 | kaldemar | asterisk needs to be configured for the phones to use it. |
| 06:24.59 | *** join/#asterisk Vince-0 (c4d7a482@gateway/web/freenode/ip.196.215.164.130) |
| 06:27.57 | Jo_Bell | Kaldemar: Suppose my application client is registered at Asterisk server A. The client may be contacting an external SIP phone registered at an asterisk server B. The external phone can even be registered on a non Asterisk PBX. Are the supporting codec information etc passed at registration time itself as part of SIP protocol? Or is it asterisk specific? |
| 06:28.48 | *** join/#asterisk din3sh (~din3sh@41.212.200.149) |
| 06:29.02 | din3sh | good morning all |
| 06:30.47 | kaldemar | Jo_Bell: registration is a way to let the other end know where a device is. it has really nothing to do with anything else. it is also not a login type of procedure. |
| 06:31.58 | Jo_Bell | Kaldemar: Then how are the codec details of the clients known to asterisk server? Are they specified in some conf files? |
| 06:32.21 | kaldemar | Jo_Bell: the codec negotiations are made during call setup. and your client will not be contacting any phone, it will connect to a conference on an asterisk server. the phones will also connect to the conference, either directly or through something, which is your choice. |
| 06:32.21 | Dennisvj | Codec negotiation is done on SIP Invite |
| 06:33.06 | kaldemar | Jo_Bell: you allow and disallow codecs for devices in sip.conf. |
| 06:33.30 | kaldemar | Jo_Bell: ultimately, the phones let asterisk know about their capabilities when a call is made. |
| 06:34.48 | Jo_Bell | kaldemar: ok. |
| 06:34.59 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
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| 07:13.20 | Jo_Bell | kaldemar: In order to call out and receive incoming calls from PSTN, I can use a third party PSTN gateway which is interfaced with asterisk right? Which is the best or the mostly used gateway according to asterisk community's general observation? |
| 07:16.26 | kaldemar | you can use an ITSP, or get your own lines that you use with telephony hardware. |
| 07:16.44 | kaldemar | there are certainly many opinions on this. |
| 07:16.46 | kaldemar | ~book |
| 07:16.46 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:17.16 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-OutsideConn.html |
| 07:19.53 | *** join/#asterisk engrxyz (~qweq@host81-150-217-167.in-addr.btopenworld.com) |
| 07:20.48 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:24.49 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 07:25.09 | jacc0 | hi all! |
| 07:25.31 | jacc0 | how do I pass variables from my dialplan to my (php)agi script? |
| 07:26.25 | jacc0 | sorry, found the answer already |
| 07:27.05 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 07:27.46 | EmleyMoor | Is there a way to see the IP that a SIP call came from? (either for calls gone, or if not possible, just for future ones |
| 07:27.50 | EmleyMoor | ) |
| 07:41.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:4cf7:a9f7:f351:fffe) |
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| 07:47.21 | jacc0 | EmleyMoor: in diallan? or just in a log file? |
| 07:47.27 | jacc0 | *dialplan |
| 07:48.58 | jacc0 | http://www.voip-info.org/wiki/view/Asterisk+func+sippeer |
| 07:49.31 | jacc0 | ${SIPPEER(ip)} will hold the ip address |
| 07:50.13 | *** join/#asterisk khurram (6e5dcd82@gateway/web/freenode/ip.110.93.205.130) |
| 07:50.29 | khurram | hello everyone |
| 07:50.34 | jacc0 | or in CLI? 'sip show peers' |
| 07:50.41 | jacc0 | hi khurram |
| 07:50.52 | khurram | i have successfully installed ubuntu on vmware and successfully run asterisk 10 on it |
| 07:50.59 | jacc0 | congratz |
| 07:51.05 | *** join/#asterisk joobie (~joobz@unaffiliated/moo0o0ooo00o0o0o) |
| 07:51.22 | joobie | hey guys.. anyone know of a setting that would be exiting my dial() if it goes for more than 60 seconds? |
| 07:51.24 | jacc0 | vmware might cause some extra jitter |
| 07:51.43 | khurram | i do not understand this extention systems are they all virtual extentions just could be any nummber or what ? |
| 07:51.46 | joobie | i have 3 numbers that dial one after another, kinda like 3 dials in a row |
| 07:52.06 | joobie | i dial each number for 30 seconds.. it dials 2, and then it exits saying "exited non-zero on.." |
| 07:52.10 | khurram | vmware causes extra jitter please explain more , hows and avoids |
| 07:52.15 | joobie | if i decrease the dial time to 10 seconds for each dial, it works |
| 07:52.56 | jacc0 | how to avoid extra jitter; don't use vmware |
| 07:53.38 | khurram | for just test proposses i had to do some RnD so i am doing it on vmware the original client might have it been running on real server |
| 07:53.40 | jacc0 | joobie: http://www.voip-info.org/wiki/view/Asterisk+func+timeout |
| 07:53.58 | joobie | thanks jacc0 |
| 07:54.03 | jacc0 | vmware is good for testing |
| 07:54.35 | jacc0 | 7 hours till my holliday starts! |
| 07:54.37 | jacc0 | :P |
| 07:55.01 | khurram | jacc0: what is the next step how can i test asterisk and is it possible to call asterisk server from windows if yes can you please throw some light on what to read further etc. |
| 07:55.17 | jacc0 | ~book |
| 07:55.17 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:55.36 | jacc0 | I use x-lite from countrepath to make calls from windows |
| 07:55.53 | jacc0 | http://www.counterpath.com/x-lite.html |
| 07:56.00 | jacc0 | (damn, I'm fast) |
| 07:56.03 | jacc0 | :p |
| 07:56.25 | khurram | jacc0: thanks |
| 07:56.45 | joobie | jacc0, i tried exten => 4444,1,Set(TIMEOUT(absolute)=0) but it still exited |
| 07:57.00 | jacc0 | O? |
| 07:57.03 | khurram | jacc0: are there any free phones to test ? |
| 07:57.09 | jacc0 | then the client may be hanging up4 |
| 07:57.38 | jacc0 | 'free phones' x-lite used to be free. isn't it free anymore? |
| 07:57.53 | joobie | jacc0, it can't be, it exits at the same part of the dialplan each time |
| 07:57.59 | joobie | same second count |
| 07:58.10 | jacc0 | client max. ringtime could be 30sec |
| 07:58.22 | jacc0 | add an answer() before you dial out |
| 07:58.34 | joobie | hmm ok |
| 07:58.50 | joobie | each time before i dial out? |
| 07:58.54 | joobie | or just at the start (first time) |
| 07:58.55 | jacc0 | nope |
| 07:59.07 | jacc0 | just the first line of your dialplan extension |
| 07:59.18 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:59.32 | joobie | ok trying again |
| 07:59.35 | jacc0 | opr take a look at your clients configuration |
| 07:59.41 | kaldemar | joobie: show your extension and a CLI output of a call. |
| 07:59.52 | joobie | so i have something like, timeout, answer,dial,wait,dial,wait,dial .. etc |
| 08:00.02 | jacc0 | that's cool |
| 08:00.13 | joobie | hope this works :P |
| 08:00.23 | jacc0 | I'm not sure why you would want to wait |
| 08:00.35 | joobie | just threw that in for testing purposes |
| 08:00.37 | jacc0 | dial(sip/100,30) will make it ring 30 sec |
| 08:00.51 | joobie | nod, im using a 20s timeout atm |
| 08:00.54 | joobie | per dial |
| 08:01.12 | joobie | jacc0, so far so good.. no hangup.. why does answer() resolve this? |
| 08:01.17 | joobie | i dont follow the logic |
| 08:01.37 | jacc0 | while the client hanges up if the call isn;t answerd in 30seconds |
| 08:01.53 | jacc0 | so you answer right away |
| 08:02.06 | jacc0 | and start dialing after that |
| 08:02.12 | joobie | so it's tricking the phone into thinking that it's answered? |
| 08:02.21 | jacc0 | take a look at your client settings |
| 08:02.31 | joobie | yea I see what you mean |
| 08:02.38 | joobie | thanks jacc0.. very shifty :) |
| 08:02.40 | jacc0 | you might be able to change max. ring time |
| 08:02.40 | joobie | i like it! :P |
| 08:02.51 | jacc0 | I'm hot!! bring it on! |
| 08:02.52 | jacc0 | :P |
| 08:02.53 | joobie | it's ok, this will do. i dont mind an answer locally to my phone |
| 08:02.54 | joobie | ahha |
| 08:02.55 | joobie | ok |
| 08:02.56 | joobie | OK |
| 08:02.58 | joobie | i have another Q.. |
| 08:03.08 | joobie | i gave up on this because i thought it would take too much effort |
| 08:03.20 | joobie | ideally, i want to dial 2 numbers at the same time, using the same sip provider |
| 08:03.38 | jacc0 | dial(sip/100&sip/200,20) |
| 08:03.41 | joobie | the provider allows multiple calls, but i tried dial(sip/provider/number&sip/provider/number) and asterisk doesnt like this |
| 08:03.41 | jacc0 | :) |
| 08:03.53 | jacc0 | asterisk does |
| 08:03.56 | joobie | cmon, i wouldn't throw you that easy one ;P |
| 08:04.01 | joobie | sec when i do the above i get the following back |
| 08:04.12 | jacc0 | :P |
| 08:04.26 | jacc0 | ~pb |
| 08:04.26 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 08:04.31 | joobie | [Jul 11 17:42:29] WARNING[7829]: app_dial.c:1209 dial_exec_full: Skipping dialing interface 'SIP/pennytel/<number>' again since it has already been dialed |
| 08:04.37 | joobie | that is what it comes back saying.. |
| 08:04.50 | jacc0 | you are dialing the same number 2 times |
| 08:04.54 | joobie | i know the multiple dial works, because i use it in another part of the dialplan to dial multiple internal extensions |
| 08:05.03 | joobie | omg |
| 08:05.04 | joobie | u are right |
| 08:05.06 | joobie | :/ |
| 08:05.07 | joobie | my bad |
| 08:05.09 | jacc0 | :P |
| 08:05.12 | joobie | thanks jacc0 |
| 08:05.18 | joobie | you are on a roll :P |
| 08:05.27 | jacc0 | 'cmon, i wouldn't throw you that easy one' I guess you did |
| 08:05.27 | joobie | i dont have any other doozies to throw your way to break that roll :P |
| 08:05.30 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 08:05.47 | jacc0 | :P |
| 08:06.09 | jacc0 | 24 hours till airport check-in |
| 08:06.17 | jacc0 | 1 month to india! |
| 08:06.36 | jacc0 | 7 workinghours left |
| 08:06.48 | jacc0 | bring it on! :P |
| 08:07.04 | *** join/#asterisk Vince-0 (c4d7bcf4@gateway/web/freenode/ip.196.215.188.244) |
| 08:08.32 | joobie | heeh |
| 08:08.42 | joobie | what are you doing in india? |
| 08:08.56 | jacc0 | backpacking starting in new delhi |
| 08:09.02 | jacc0 | :) |
| 08:09.33 | joobie | ahh nice |
| 08:09.51 | joobie | i hear india turning into a popular place to visit |
| 08:11.42 | joobie | you know this dialplan |
| 08:11.52 | joobie | I have this offshore developer working for us |
| 08:12.01 | joobie | he's meant to be fulltime.. but he's just vanished |
| 08:12.20 | joobie | i have 2 of his numbers.. im just dialling both of them in a loop for 30 seconds |
| 08:12.31 | joobie | asterisk is a beautiful thing |
| 08:13.36 | joobie | and now im going to have a smoke |
| 08:13.40 | joobie | while it continues to dial :P |
| 08:13.41 | joobie | brb |
| 08:15.41 | jacc0 | hehehe |
| 08:20.20 | khurram | can some one please help me in understanding the line permit: 192.168.5.0/255.255.255.0 and how to know the values here if asterisk is running inside unbuntu inside vmware on windows7 |
| 08:21.44 | joobie | what do you mean khurram |
| 08:22.20 | khurram | joobie: the line is inside the sip.confg to make two softphones |
| 08:23.04 | joobie | ahh |
| 08:23.17 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 08:23.17 | joobie | that would be based on src ip |
| 08:23.44 | joobie | if you are not doing source nat, then it would be the true destination |
| 08:23.50 | joobie | -destination +source |
| 08:25.50 | khurram | joobie: thanks |
| 08:28.54 | joobie | khurram, if you want a quick way to see if you are doing snat |
| 08:29.01 | joobie | just do a tcpdump on the asterisk box |
| 08:31.52 | khurram | cheers joobie |
| 08:34.34 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e95a.pool.mediaWays.net) |
| 08:40.23 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:43.54 | jacc0 | you might want to setup your VM to use bridged mode |
| 08:44.38 | *** join/#asterisk sbw_DvD (~chinky@D522448D.static.ziggozakelijk.nl) |
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| 09:12.22 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 09:15.29 | *** join/#asterisk hay (~hay@tm.84.52.129.113.dc.cable.static.telemach.net) |
| 09:17.31 | hay | hi all... I have installed Digium Fax for Asterisk and 1 license file which seems to work according to fax show stats.. now I don't know how to start sendfax application - from CLI? I know that a parameter has to be .tiff file, but can't find any sending options even in asterisk GUI... TIA for any help available |
| 09:19.37 | bulkorok | hay: check the admin guide: http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf |
| 09:20.41 | bulkorok | i personally use a php script with an originate via AMI and send the call after answering to the [faxsender] context where I have setup the SendFax app |
| 09:23.45 | hay | bulkorok, thanks... I think I have gone through it already... so you use a PHP script to select a TIFF file, upload it to the server and with the fax number send it? |
| 09:24.54 | bulkorok | well the upload to the server is done with a java agent I didn't build... so this is your own part... but I think a simple html/php form for uploading is not big magic... |
| 09:25.48 | bulkorok | and after upload you have to trigger an originate to the receiving side via AMI and php and after connecting send it to the exten where the sendfax is |
| 09:26.13 | hay | bulkorok, uploading won't be problem, what I find problematic is calling the asterisk SendFax from PHP :) do you perhaps have some examples of it available? |
| 09:26.49 | bulkorok | wait :) |
| 09:29.07 | bulkorok | you have to create a socket to asterisk AMI: $socket = fsockopen("127.0.0.1","5038", $errno, $errstr, 10); |
| 09:29.21 | bulkorok | and via this socket you can send the Originate command |
| 09:29.35 | bulkorok | at first you have to login to ami for sure... |
| 09:29.43 | bulkorok | fputs($socket, "Action: Login\r\n"); |
| 09:30.06 | bulkorok | and then send the originate with the vars you need |
| 09:30.13 | bulkorok | fputs($socket, "Action: Originate\r\n"); |
| 09:30.37 | bulkorok | fputs($socket, "Callerid: 123 <123>\r\n"); |
| 09:30.59 | bulkorok | fputs($socket, "Channel: SIP/gw/number\r\n"); |
| 09:31.08 | bulkorok | fputs($socket, "Exten: faxout\r\n"); |
| 09:31.15 | bulkorok | fputs($socket, "Context: faxsender\r\n"); |
| 09:31.22 | bulkorok | fputs($socket, "Priority: 1\r\n"); |
| 09:31.58 | bulkorok | and with the help of variables you can send the filename of the tiff to the SendFax |
| 09:32.05 | hay | thanks man! now I have something to work on :) |
| 09:32.07 | bulkorok | fputs($socket, "Variable: faxdatei=$tiffile\r\n\r\n"); |
| 09:32.25 | bulkorok | don't forget to logoff from ami after all this |
| 09:32.34 | hay | and it actually doesnt seem so hard :) |
| 09:32.40 | bulkorok | not really... |
| 09:33.40 | bulkorok | I coded a little bash-script to send variable sender-number and receiver-number and for sure faxfile to the php-script |
| 09:43.07 | hay | I have added the following after Login: |
| 09:43.15 | hay | fputs($socket, "Username: username\r\n"); |
| 09:43.15 | hay | fputs($socket, "Secret: password\r\n"); |
| 09:43.36 | hay | but am unable to authenticate as reported in full logfile... |
| 09:43.54 | hay | do all the login params have to be sent in one line? |
| 10:08.39 | hay | oh, I had to use secret, defined in /etc/asterisk/manager.conf... now working as it should :-) |
| 10:42.36 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-ijllpvtbavpjgvmo) |
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| 11:32.07 | *** join/#asterisk _omer (omer@66.165.232.51) |
| 11:38.53 | *** join/#asterisk Jo_Bell (~sasishiju@115.249.39.106) |
| 11:41.36 | Jo_Bell | Hi. I am not asking a right question. But I need answer from your experiences in Asterisk. I am searching out various ITSP in US and am in a fix to choose one. If any of you can share your experience on this, it will be useful to short list one. |
| 11:42.16 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 11:42.31 | Kalidarn | is there an echo command in asterisk console to echo to log files?" |
| 11:42.43 | Kalidarn | i was under the assumption it was echo |
| 11:43.03 | Kalidarn | *CLI> echo test |
| 11:43.03 | Kalidarn | No such command 'echo test' (type 'core show help echo test' for other possible commands) |
| 11:43.39 | Jo_Bell | Basically I want to get PSTN connectivity without having a hardware setup in my asterisk server. |
| 11:44.12 | kaldemar | ~itsp-list |
| 11:44.12 | infobot | itsp-list is probably Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 11:44.12 | bulkorok | Jo_bel: in and outgoing calls!? |
| 11:44.54 | bulkorok | Kalidarn: what do you want in the file!? |
| 11:44.59 | Kalidarn | just some text |
| 11:45.07 | Kalidarn | as ma marker im trying to debug a module |
| 11:45.13 | bulkorok | oh... |
| 11:45.14 | Kalidarn | ie commenting |
| 11:45.20 | bulkorok | NoOp |
| 11:45.53 | Jo_Bell | bulkorok: Yes clients should be able to call out and the incoming callers should contact the ITSP and then our asterisk server |
| 11:45.54 | bulkorok | or Log() for sure |
| 11:46.10 | Kalidarn | yeah but do those work in the CLI |
| 11:46.25 | Kalidarn | so i can echo my testing notes to a log file |
| 11:46.48 | *** join/#asterisk din3sh (~din3sh@41.76.47.95) |
| 11:46.57 | din3sh | hello all |
| 11:47.27 | bulkorok | Kalidarn: no... thy work from dialplan |
| 11:47.36 | din3sh | i cannot receive calls from UK on my PRI, i can receive calls from other countries correctly |
| 11:47.43 | Kalidarn | that's not what i wanted |
| 11:48.20 | din3sh | can pridialplan=xxx be a reason for that? |
| 11:49.04 | Kalidarn | oh wait i can do it by doing |
| 11:49.13 | bulkorok | Kalidarn: console send text!? |
| 11:49.15 | Kalidarn | echo "\n\nBLAHBLAH\n\n">> /var/log/asterisk/full |
| 11:49.17 | Kalidarn | in a terminal |
| 11:49.25 | Kalidarn | at the point i want |
| 11:49.30 | bulkorok | ok... |
| 11:49.36 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:49.36 | bulkorok | it's the hard way... |
| 11:49.47 | bulkorok | forget console |
| 11:50.13 | Jo_Bell | bulkorok: thanks |
| 11:50.44 | din3sh | i cannot receive calls from UK on my PRI, i can receive calls from other countries correctly, what might be the problem? chan_dahdi settings |
| 11:50.44 | din3sh | ? |
| 11:51.09 | bulkorok | din3sh: do you have any logs?! |
| 11:54.19 | kaldemar | Kalidarn: "!echo ..." will do it from CLI |
| 11:54.43 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 11:56.02 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 11:58.45 | *** join/#asterisk Hunterkll (~Hunterkll@c-68-50-253-238.hsd1.md.comcast.net) |
| 11:58.47 | Hunterkll | Hello! |
| 11:59.09 | Hunterkll | I'm debugging and repairing a system that fell into my hands... an asterisk system with IVR configured |
| 11:59.27 | Hunterkll | Google voice inbound is apparently forwarded to a 3rd party sip provider (ipcomms) |
| 12:00.02 | Hunterkll | I can dial, and see session intilizations, but don't get the IVR menu anymore when it picks up, or if it dosn't , get the GV unvailable subscriber message |
| 12:00.15 | Hunterkll | I'm completely lost and don't know where to start looking |
| 12:00.28 | Hunterkll | I've configured a SIP phone to connect to the server and CAN do outbound calling from the phone system, however |
| 12:01.05 | Hunterkll | i'm not sure where to debug |
| 12:01.15 | Hunterkll | this was a 'working' config, prior to several IP moves, firewall configs, etc |
| 12:01.16 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:01.37 | Hunterkll | sip provider / GV havn't changed, it was internal lan only. I've removed all firewall restrictions, restored all ATA device's connectivity, etc |
| 12:01.50 | din3sh | bulkorok: actually when someone from UK is trying to call the number, he hears a busy tone |
| 12:02.02 | din3sh | and i dont see any number coming in the asterisk box either |
| 12:02.12 | din3sh | the logs dont even show the call arriving |
| 12:02.27 | din3sh | the person has tried to call several DIDs |
| 12:02.42 | din3sh | the calls dont arrive on the asterisk box |
| 12:02.47 | bulkorok | "sip set debug on" in CLI and look if there is an INVITE |
| 12:02.52 | din3sh | when ppl when other countries call in |
| 12:03.02 | din3sh | the calls get through properly |
| 12:03.10 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-7-139.bois.qwest.net) |
| 12:03.28 | bulkorok | oh wait... you said PRI?! |
| 12:03.32 | din3sh | sip debug or pri debug? |
| 12:03.38 | din3sh | yeah |
| 12:03.49 | bulkorok | well the pri :) |
| 12:03.52 | bulkorok | then |
| 12:04.01 | din3sh | my pridialplan is set to local |
| 12:04.05 | Hunterkll | is it normal to get a 401 unauth'd before a 200 ok? |
| 12:04.06 | din3sh | in chan_dahdi |
| 12:04.16 | din3sh | can this be the problem? |
| 12:04.33 | Kalidarn | is |
| 12:04.37 | Kalidarn | dahdi set debug on |
| 12:04.39 | Kalidarn | correct? |
| 12:04.49 | Kalidarn | to enable debugging for the dahdi module |
| 12:04.51 | bulkorok | Hunterkll: yeah... after the 401 the UAC is sending an register with md5 password |
| 12:04.59 | Hunterkll | okay |
| 12:05.21 | Hunterkll | just saw incomming call ipc |
| 12:05.24 | *** join/#asterisk sbw_DvD (~chinky@D522448D.static.ziggozakelijk.nl) |
| 12:05.34 | Hunterkll | so DID is routing to us , after one ring, we pick up |
| 12:05.34 | din3sh | i have set pri intense debug on the span |
| 12:05.36 | Hunterkll | then silence |
| 12:05.42 | din3sh | i still dont see the call coming through |
| 12:05.46 | din3sh | :s |
| 12:05.57 | bulkorok | din3sh: then it's never arriving at your box => call ISP |
| 12:06.10 | Hunterkll | grumbles |
| 12:06.27 | Hunterkll | <-- thrust into inherting entire hackerspace network / services role guy. :P |
| 12:07.28 | Kalidarn | hmm |
| 12:07.29 | din3sh | the thing is that apart calls from UK |
| 12:07.33 | bulkorok | Hunterkll: do you have any logs from CLI?! |
| 12:07.39 | Kalidarn | doesn't seem to be anyway to debug dahdi from the CLI |
| 12:07.42 | Kalidarn | like other modules |
| 12:07.44 | Hunterkll | bulkorok, I have full access to everything |
| 12:07.44 | din3sh | i can recieve local pri calls as well as from other countries |
| 12:07.45 | din3sh | :/ |
| 12:07.50 | Hunterkll | I just don't know jack to where to look |
| 12:07.55 | Hunterkll | it was working forever. then stopped. |
| 12:08.08 | Hunterkll | IP range shifts, I can dial out from a SIP phone that's configured |
| 12:08.21 | Hunterkll | I had been able to dial INTO said sip phone before, too |
| 12:08.37 | Hunterkll | ipcomms is our did provider, apparently, it seems GV forwards calls to that # |
| 12:08.42 | Hunterkll | outbound calls work fine |
| 12:09.00 | Hunterkll | do you want me to CLI, call in, and pastebin logs? |
| 12:09.17 | bulkorok | Hunterkll: yes |
| 12:09.24 | Hunterkll | okay. one sec. |
| 12:09.27 | bulkorok | core set verbose 5 |
| 12:09.30 | Hunterkll | well, perhaps more than one. |
| 12:09.38 | Hunterkll | core set verbose 5 within asterisk -r ? |
| 12:09.42 | bulkorok | yeap |
| 12:10.22 | kaldemar | Kalidarn: what are you trying to debug? |
| 12:10.46 | Kalidarn | well im having some trouble with a line and its posssible its a dahdi bug |
| 12:10.50 | Kalidarn | on freebsd |
| 12:10.52 | bulkorok | din3sh: did you change sth before UK went down?! |
| 12:11.01 | Kalidarn | like with other modules you can just do module set XXXX |
| 12:11.07 | Kalidarn | but it doesn't seem you can do that with dahdi |
| 12:11.09 | kaldemar | Kalidarn: what kind of a line? |
| 12:11.20 | Kalidarn | ie like |
| 12:11.21 | Kalidarn | rtp set debug on |
| 12:11.35 | Kalidarn | to get dhadi messages to appear in the full log |
| 12:11.42 | Kalidarn | *dahdi |
| 12:11.58 | Hunterkll | need to config putty to keep more terminal lines |
| 12:12.45 | *** join/#asterisk khurram (6e5dcd82@gateway/web/freenode/ip.110.93.205.130) |
| 12:12.48 | kaldemar | Hunterkll: asterisk -vvvr | tee /tmp/asterisk.log |
| 12:12.50 | khurram | hi all |
| 12:13.11 | khurram | two days of contineious research on asterisk and phones are rining in my brain but not in asterisk |
| 12:13.13 | Hunterkll | bulkorok, as an aside, as a semi-major ircop, I thank you for helping everyone at once. :) |
| 12:13.16 | Hunterkll | laso, yes |
| 12:13.19 | Hunterkll | thanks |
| 12:13.53 | khurram | the command on ubuntu->asterisk CLI->sip reload is not doing reload what should i do |
| 12:14.19 | kaldemar | khurram: what is it doing? |
| 12:14.27 | bulkorok | Hunterkll: :) I'll do my very best... |
| 12:14.49 | Hunterkll | I see our stuff playing |
| 12:14.51 | Hunterkll | as if it were answering |
| 12:14.56 | Hunterkll | i don't hear anythign on phone |
| 12:15.06 | Hunterkll | POTS phone hangup on my end |
| 12:15.14 | Hunterkll | terminate and copy log now? |
| 12:15.16 | Kalidarn | actually i might do it without dahdi debugging as i think it might be something else |
| 12:15.22 | khurram | kaldemar: it just does nothing |
| 12:15.31 | beaver_rrr | does ChanSpy affects call in any way? can it break a call? |
| 12:15.40 | bulkorok | Hunterkll: sounds like the RTP traffic is not comming to your phone... do you can handle tcpdump!? |
| 12:15.50 | kaldemar | khurram: do you get any output in CLI? |
| 12:16.00 | khurram | kaldemar: no |
| 12:16.21 | khurram | it does show output when i do CLI> dialplan reload |
| 12:16.42 | Hunterkll | bulkorok, tcpdump is installed |
| 12:16.46 | Hunterkll | want to check my logfile first? |
| 12:16.54 | bulkorok | sure :) |
| 12:16.59 | Hunterkll | okay |
| 12:17.02 | Hunterkll | let me get it out.... |
| 12:17.20 | kaldemar | khurram: pastebin your sip.conf |
| 12:17.26 | Hunterkll | i just reconstructed a juniper / vyatta / netgear / cisco network, this is not my friendly time right now. :P |
| 12:17.30 | kaldemar | khurram: do any other sip commands work? |
| 12:17.42 | bulkorok | khurram: sip reload doesn't has any output when core set verbose is 0 |
| 12:17.53 | bulkorok | type core set verbose 5 and the sip reload... |
| 12:18.06 | bulkorok | there should come "Reloading SIP" |
| 12:18.32 | bulkorok | Hunterkll: sounds like a nice S&M time :-D |
| 12:18.44 | Hunterkll | I have some dog collars nearby and am wearing a suit... |
| 12:18.48 | Hunterkll | >_> |
| 12:18.58 | bulkorok | :) |
| 12:19.08 | Hunterkll | at least i have beer... |
| 12:19.39 | bulkorok | is @ work => no beer :( |
| 12:20.04 | khurram | kaldemar: http://paste.ubuntu.com/1086079/ |
| 12:20.10 | Hunterkll | http://epfarms.org/~kaiba/asterisk.log |
| 12:20.13 | Hunterkll | that's my log |
| 12:20.43 | Hunterkll | is sitting at a desk with 3 laptps and 2 racks, just trying to get stuff working. XD |
| 12:21.07 | Hunterkll | cisco this, cisco that, vlan this, vlan that... |
| 12:21.25 | kaldemar | khurram: don't see any fatal syntax errors there. how did you determine that the reload does not work? |
| 12:22.48 | khurram | well after i added the two phones at the bottom of sip.conf ; i am supposed to reload the sip.conf through command line CLI> sip reload and this command does nothing no message comes up syaing Reloading sip |
| 12:23.02 | khurram | the message does come up when i do dialplan reload |
| 12:23.31 | bulkorok | khurram: check with 'sip show peers' |
| 12:23.39 | khurram | no peers yet |
| 12:24.18 | Hunterkll | bulkorok, offhand, everything is nat'd at the edge, possible problem there? |
| 12:24.19 | khurram | all empty fresh copy of asterisk just made it to run in windows 7 -> vmware->ubuntu->asterisk |
| 12:25.08 | bulkorok | Hunterkll: NAT is everytime hell... check the nat option in the sip peer |
| 12:25.26 | kaldemar | khurram: "module show like chan_sip" |
| 12:25.28 | Hunterkll | I don't understand what you said bast 'check the' |
| 12:25.45 | khurram | is that an asterisk CLI> command ? |
| 12:26.16 | Hunterkll | also, this was a 'known working' without asterisk config modification. so... |
| 12:26.32 | khurram | Module Description Use Count chan_sip.so Session Initiation Protocol (SIP) 0 1 modules loaded |
| 12:26.35 | Hunterkll | and that included ip shifts leaving it working, and edge gateway work leaving it working |
| 12:26.57 | bulkorok | well then it should be sth with firewall/routing stuff... |
| 12:27.10 | Hunterkll | i've elimated any internall firewalling |
| 12:27.13 | Hunterkll | it's purely nat'd now |
| 12:27.15 | bulkorok | the asterisk CLI doesn't look like there ist sth wrong |
| 12:27.19 | khurram | kaldemar: ? |
| 12:27.22 | Hunterkll | any subnet can talk to any subnet |
| 12:27.35 | Hunterkll | vlan routing has no firewalls |
| 12:27.57 | Hunterkll | flow graphs show no drops, too |
| 12:27.59 | WIMPy | din3sh: I assume your box is not located in the UK then? |
| 12:28.20 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-llxnyyeyvnghdtws) |
| 12:28.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:28.31 | Hunterkll | bulkorok, and yea, i can call outbound from our # fine.... weird... |
| 12:28.41 | Hunterkll | ipcomms being douchey? :) |
| 12:28.51 | Hunterkll | i don't know much about them other then being free inbound did provider |
| 12:30.01 | bulkorok | could be... you can check the RTP in asterisk with 'rtp set debug on' |
| 12:30.56 | Hunterkll | WHOLE lotta RTP packets being sent |
| 12:31.23 | [TK]D-Fender | Hunterkll, show us the CALL |
| 12:31.27 | Hunterkll | <PROTECTED> |
| 12:31.28 | Hunterkll | <PROTECTED> |
| 12:31.28 | Hunterkll | <PROTECTED> |
| 12:31.30 | [TK]D-Fender | ~pb |
| 12:31.31 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 12:31.32 | [TK]D-Fender | PASTEBIN |
| 12:31.36 | Hunterkll | I know |
| 12:31.44 | Hunterkll | I sent a logfile to bulkorok earlier already |
| 12:31.51 | Jo_Bell | bulkorok : for testing the PSTN functionality with ITSP say teliax, suppose i got a package from them, which are the details that I need to enter in the sip.conf file? |
| 12:32.39 | Hunterkll | bulkorok, would some kind of rtp log be helpful? |
| 12:32.41 | kaldemar | khurram: interesting. your sip.conf is not to blame if what you pasted is really all you have. |
| 12:32.51 | khurram | yes |
| 12:33.07 | kaldemar | khurram: what do you get with "module unload chan_sip.so" and "module load chan_sip.so"? |
| 12:34.11 | khurram | kaldemar: module unload chan_sip.so Unloaded chan_sip.so |
| 12:34.15 | khurram | works fine |
| 12:34.44 | khurram | kaldemar: module load chan_sip.so Loaded chan_sip.so SIP channel loading... |
| 12:35.36 | kaldemar | khurram: "core set verbose 10" and try again. |
| 12:36.02 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.51.199) |
| 12:36.57 | Hunterkll | I just had a STUPID idea |
| 12:37.07 | bulkorok | Hunterkll: take a look to the IPs the rtp debug is spitting out and check if they are correct.. |
| 12:37.11 | Hunterkll | I think our damn TRAFFIC LIGHT interface being down might be screwing us |
| 12:37.25 | bulkorok | hu!? |
| 12:37.35 | [TK]D-Fender | Less thought, more show :) |
| 12:37.39 | Hunterkll | our phone system controls the space's traffic light if triggered correctly |
| 12:37.44 | [TK]D-Fender | Where's the call with SIP debug? |
| 12:38.07 | Hunterkll | http://epfarms.org/~kaiba/asterisk.log |
| 12:39.13 | Hunterkll | 1 sec pulling some cable |
| 12:40.38 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:40.44 | [TK]D-Fender | [0KReliably Transmitting (no NAT) to 64.154.41.150:5060: Contact: <sip:2014771307@10.101.0.10:5060> |
| 12:40.54 | [TK]D-Fender | Hunterkll, Your * is not configured properly to work from behind NAT |
| 12:40.59 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 12:41.16 | [TK]D-Fender | I expect dead audio |
| 12:42.41 | Hunterkll | [TK]D-Fender, then... how did it work before? |
| 12:42.42 | Hunterkll | :/ |
| 12:42.55 | [TK]D-Fender | ~sipnat |
| 12:42.56 | infobot | [~sipnat] 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 12:42.58 | [TK]D-Fender | ^^ |
| 12:43.18 | [TK]D-Fender | in 1.6+ substitute "canreinveite" with "directmedia" |
| 12:43.26 | [TK]D-Fender | Go fix your settings |
| 12:45.46 | Hunterkll | understood, investigating |
| 12:46.19 | Hunterkll | I just don't knwo why it chose last month to break when it had worked with the same config for ... 1.25 years? |
| 12:48.22 | [TK]D-Fender | maybe a router changed that was compensating. |
| 12:48.58 | Hunterkll | Hn. |
| 12:49.19 | Hunterkll | edge router and internal router hasn't changed since it was working. configs, perhaps, but seems 'near identicle' |
| 12:49.49 | Hunterkll | got the traffic light's ip power control online, anyway |
| 12:50.17 | [TK]D-Fender | "near identical" != 'identical". Either way, go fix up those settings |
| 12:50.29 | Hunterkll | I can't figure out how to get an asterisk version |
| 12:50.37 | Hunterkll | asterisk -v gives me nothing, nor nothing in CLI |
| 12:50.46 | bulkorok | core show version |
| 12:51.04 | Hunterkll | 1.8.3.3 |
| 12:51.07 | Hunterkll | should be new enough then! |
| 12:51.21 | bulkorok | 1.8.14.0 just appeared today |
| 12:51.30 | Hunterkll | [TK]D-Fender, can i avoid this crap with port forwards? or is having NAT still a barrier |
| 12:51.47 | [TK]D-Fender | Hunterkll, The guide tells you what to set. Please follow it... |
| 12:51.53 | Hunterkll | understood! |
| 12:51.53 | [TK]D-Fender | and yes you need port forwards |
| 12:52.07 | [TK]D-Fender | Amonst other things |
| 12:53.15 | Hunterkll | set service "RTP" + udp src-port 0-65535 dst-port 5004-5004 |
| 12:53.27 | Hunterkll | from the salvaged juniper config |
| 12:53.34 | Hunterkll | i assume this might be... kind of important? |
| 12:53.48 | Hunterkll | also protocol tcp lines, as well |
| 12:54.13 | bulkorok | rtp + udp forwarding is what you need... |
| 12:54.22 | [TK]D-Fender | rtp is carried over UDP |
| 12:54.47 | Hunterkll | so it appears to be an omission in my knoweldge of juniper configs. very well |
| 12:54.48 | [TK]D-Fender | and not a SINGLE dest port |
| 12:54.53 | [TK]D-Fender | rtp.conf <--- |
| 12:55.19 | Hunterkll | the config I read the RTP line from was the 'last known working' |
| 12:57.15 | [TK]D-Fender | Hunterkll, Now is a really really great time to throw out all memory of "before". |
| 12:57.21 | khurram | thanks it worked |
| 12:57.22 | Hunterkll | Agreed. |
| 12:57.28 | khurram | kaldemar: thanks |
| 12:57.55 | Hunterkll | I have all internal network and ChaosVPN working. I have all routing done, with only minimal 'before'. all etherchannels up and running. |
| 12:58.41 | *** join/#asterisk n3hxs (~ed@ann100fw01.answernet.com) |
| 12:59.35 | Hunterkll | Now... I shall read NAT docs on asterisk |
| 13:04.29 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 13:06.45 | *** join/#asterisk nir (~quassel@bzq-84-111-107-170.red.bezeqint.net) |
| 13:10.01 | din3sh | banging his head on his desk |
| 13:10.23 | Hunterkll | don't do that! |
| 13:10.28 | din3sh | lol |
| 13:10.30 | Hunterkll | i brought a cinderblock for that purpose... |
| 13:10.42 | din3sh | the call from UK still not reaching the PRI |
| 13:10.54 | Hunterkll | heh |
| 13:10.57 | din3sh | all other countries are able to call |
| 13:11.00 | din3sh | in |
| 13:11.08 | Hunterkll | i have zero inbound :P |
| 13:11.34 | din3sh | setting pridialplan=unknown doesnt help |
| 13:12.34 | Hunterkll | adds nat=yes and reboots |
| 13:12.39 | Hunterkll | just for the hell of it |
| 13:13.06 | din3sh | all pri calls come in ok, apart from calls coming from UK!!!!!!!!!!!!!!!!!!!!!! |
| 13:13.12 | din3sh | :/ |
| 13:15.13 | [TK]D-Fender | ~osmosis |
| 13:15.14 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 13:15.16 | [TK]D-Fender | :) |
| 13:15.41 | bulkorok | what the... ?! |
| 13:16.06 | bulkorok | din3sh: do you can see the UK call comming to your box?! |
| 13:17.09 | bulkorok | what is your ISP telling you!? |
| 13:17.25 | din3sh | no |
| 13:17.46 | din3sh | the call doesnt even come in the pri |
| 13:17.56 | din3sh | the caller on the other side says he hears busy tone |
| 13:18.15 | bulkorok | then it's a provider problem I would guess... |
| 13:19.11 | [TK]D-Fender | Correct |
| 13:20.41 | Katty | hello my asterisk does not work at all how to fix plz??? it makes gurgle noise??? |
| 13:21.28 | Hunterkll | Katty, put it in water! |
| 13:21.34 | Hunterkll | gurggling noise = needs water to breath |
| 13:21.42 | Katty | *hee* |
| 13:21.44 | Katty | adds to list |
| 13:21.52 | WIMPy | din3sh: I assume your box is not located in the UK then? |
| 13:22.05 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-18.nwrknj.fios.verizon.net) |
| 13:22.15 | WIMPy | And pri*dialplan if for outgoing only. |
| 13:22.26 | din3sh | WIMPy:correct |
| 13:23.04 | WIMPy | Looks like a routing issue then. |
| 13:23.05 | Hunterkll | [TK]D-Fender, followed guide, canreinvite=no, nat=yes, host set right... still same result |
| 13:23.07 | din3sh | an update, calls from 442072XXXXX01 comes in |
| 13:23.28 | din3sh | caller from 442072XXXXXXX02 hears busy tone |
| 13:23.33 | WIMPy | din3sh: Rather than area code, better ask for the provider. |
| 13:24.12 | Hunterkll | In USA, some CLECs disable some area codes due to exchange costs, for VoIP providers |
| 13:24.21 | WIMPy | How many callers do you have trying? Maybe someone has international calls barred? |
| 13:24.33 | din3sh | 3 of them from UK |
| 13:24.41 | din3sh | i talked to the one of the callers |
| 13:24.56 | din3sh | says since the * box has been setup last week |
| 13:25.16 | din3sh | sometimes his calls get through |
| 13:25.22 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:25.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:25.24 | din3sh | other times he hears busy tone |
| 13:26.43 | din3sh | here i can see dahdi channels free but the call doesn come through |
| 13:26.57 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 13:34.49 | *** join/#asterisk blindhog (~josh@adsl-75-62-197-176.dsl.rcsntx.sbcglobal.net) |
| 13:35.20 | [TK]D-Fender | Hunterkll, I'm not seeing your configs and the new call attempts... |
| 13:36.44 | Hunterkll | neither am I |
| 13:37.00 | Hunterkll | i'll pasted modified sip.conf config, bt the log result is exactly the same, barring timestamps |
| 13:37.46 | Hunterkll | only modifications: http://pastebin.com/MVxRCEeU |
| 13:43.44 | [TK]D-Fender | User-Agent: Asterisk PBX 1.8.3.3-0ubuntu1~maverick1 |
| 13:44.03 | [TK]D-Fender | <[TK]D-Fender> in 1.6+ substitute "canreinveite" with "directmedia" <------- |
| 13:44.45 | [TK]D-Fender | fix this as I told you, then provide new configs and new call & register attempts and show me what you've done on the router side |
| 13:45.57 | Katty | ATTENTION |
| 13:46.00 | Katty | I HAVE ICED COFFEE. |
| 13:46.01 | Katty | that is all. |
| 13:47.35 | bulkorok | i have hot coffee... |
| 13:48.20 | [TK]D-Fender | is not suitably caffeinated this morning. |
| 13:48.26 | newtonr | green tea, in the houseee! |
| 13:48.52 | Katty | does green tea have much caffeine? |
| 13:49.10 | newtonr | on average a little less than your average coffee |
| 13:49.20 | newtonr | keyword average |
| 13:49.22 | newtonr | a lot of variables |
| 13:49.47 | Katty | i've no idea how the boy makes his coffee |
| 13:49.51 | Katty | but it's awesomesauce. |
| 13:50.11 | Katty | i /do/ know he grinds the beans himself |
| 13:50.39 | Katty | but i can safely say it's packed full of caffeine. |
| 13:50.50 | Katty | sneaks additional caffeine into newtonr's tea |
| 13:51.35 | newtonr | nooo! |
| 14:02.00 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:02.04 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 14:04.31 | Katty | WELL FINE |
| 14:04.35 | Katty | takes caffeines back. |
| 14:04.55 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:05.13 | *** part/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 14:05.36 | Katty | i hope danny gets out of the dr's soon. |
| 14:06.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:06.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:07.35 | leifmadsen | efffffffffffff!@!!! my asterisk is broken I need the helpz!@ |
| 14:07.50 | Katty | applies bandaids and ice creams. |
| 14:08.29 | leifmadsen | Katty: wait wait wait |
| 14:08.31 | leifmadsen | what kind of ice cream? |
| 14:09.04 | Katty | mmmm, pistachio. |
| 14:09.26 | leifmadsen | bah that helps nothing!@ |
| 14:11.17 | Katty | :< |
| 14:11.30 | Katty | ice cream fixes everything! |
| 14:11.55 | leifmadsen | DOES IT?! |
| 14:12.04 | leifmadsen | I might disagree |
| 14:12.13 | leifmadsen | now if you said something like moose tracks, then perhaps I could be convinced |
| 14:12.36 | Katty | pfff. |
| 14:13.07 | Katty | you can make your own moose tracks |
| 14:13.28 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 14:13.35 | leifmadsen | step 1) obtain a moose |
| 14:13.39 | leifmadsen | step 3) profit |
| 14:17.02 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:17.07 | chuckf | but getting the moose to walk through ice cream is no trivial task |
| 14:17.30 | Katty | hi chuck |
| 14:17.34 | Katty | happy belated birthday |
| 14:17.40 | chuckf | hi Katty, and thanks |
| 14:17.48 | Katty | how're you dear? |
| 14:18.17 | chuckf | doing okay. playing catchup after a couple of busy days of doing other things |
| 14:18.28 | chuckf | all work related stuffs |
| 14:19.19 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 14:19.48 | wdoekes | leifmadsen: step 2) put a nonce in the bye? |
| 14:20.17 | leifmadsen | wdoekes: yes! oh so I found out where this "nonce" lived |
| 14:20.28 | leifmadsen | apparently I'm getting a BYE with Proxy-Authorization headers... |
| 14:20.39 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:21.14 | wdoekes | .. which is fine. but I'd guess that asterisk doesn't check them |
| 14:23.00 | leifmadsen | wdoekes: aye that's what I suspected |
| 14:23.20 | leifmadsen | it was messing with an edgemark though that is doing ALG |
| 14:23.27 | leifmadsen | so asterisk was never seeing the BYE |
| 14:23.48 | wdoekes | that would pose a problem ;) |
| 14:24.00 | leifmadsen | although I have a 1.4 server and a 1.8 server, and the 1.4 servers don't seem to have it, so I'm not entirely sure what is placing that header on SIP packet |
| 14:24.11 | leifmadsen | wdoekes: especially with SLA and device states :D |
| 14:24.59 | wdoekes | odd |
| 14:25.20 | leifmadsen | ya, box is behind the same opensips proxy, so I'm not sure if that is adding it or not |
| 14:25.29 | leifmadsen | or if asterisk is, although I've never seen that header in a BYE |
| 14:26.13 | leifmadsen | the issue was with the edgemark though, they said they were incorrectly dropping the BYE when the nonce in the proxy-authorization header was blank, but a newer firmware fixes that |
| 14:27.02 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:27.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:27.03 | wdoekes | what make of UA sent the BYE? |
| 14:28.10 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 14:29.42 | leifmadsen | actually, looks like the BYE is generated by Asterisk |
| 14:29.52 | leifmadsen | I'm looking at the trace again |
| 14:32.29 | leifmadsen | wdoekes: huh well chan_sip.c does have 3 instances of Proxy-Authorization in it but it doesn't appear to be actually doing much |
| 14:33.25 | wdoekes | that changes things.. did asterisk start sending auth with in-dialog requests? |
| 14:33.57 | wdoekes | "Also used for authentication of BYE" |
| 14:36.05 | *** join/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 14:36.46 | leifmadsen | wdoekes: ummm good question |
| 14:37.06 | wtfitsme | In Asterisk 1.6.2 is there the possibility of us P-Asserted-Identity header to fix the caller id of attended transfers? |
| 14:37.30 | wtfitsme | of using (not "of us") |
| 14:39.10 | wdoekes | leifmadsen: looks like it always called transmit_request_with_auth for a SIP_BYE |
| 14:40.00 | wtfitsme | Actually does anyone have the Callerid of the attended transfered call working correctly in 1.6? |
| 14:40.42 | WIMPy | wtfitsme: You may be asking a few years late. |
| 14:40.55 | wtfitsme | WIMPy: Ha! |
| 14:41.14 | wtfitsme | WIMPy: Watcha mean? |
| 14:42.21 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:42.21 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:42.52 | WIMPy | 1.6 is historic. |
| 14:43.50 | wtfitsme | That's is what I figured you meant…Unfortunately I need to stick with this version as we have a lot of custom AGI's and what not... |
| 14:45.40 | WIMPy | Upgrading from 1.6 shouldn't make too much difference configuration wise. |
| 14:45.54 | WIMPy | Unless the config is even older. |
| 14:46.13 | wtfitsme | true and yes the config is ancient! |
| 14:47.44 | oej | Hey, this is not Microsoft land. We don't force people to upgrade just because there are new versions out there. If an old version works, that is fine. We do limit what the dev team supports, but that's a separate issue. Go ahead and use version 1.0 if that works for you! |
| 14:48.35 | oej | Having said that, the 1.6.x releases was a disaster from many points of view and I don't recommend using them. |
| 14:48.57 | coppice | windows 7 has only just passed the market share of XP,and 1.6 is newer than XP |
| 14:49.02 | WIMPy | Has been the worst branch in my experience as well. |
| 14:49.16 | leifmadsen | wdoekes: hmmm, always as in, historically always? |
| 14:49.16 | oej | Coppice! Hi! |
| 14:50.01 | oej | A BYE should be authenticated. |
| 14:50.09 | wtfitsme | oej: Indeed Olle, Do you have any insight in to getting callerid correct in attended transfers…I thought I might get this working with P-Asserted-Identity… any thoughts? |
| 14:50.16 | leifmadsen | oej: interesting, ya I didn't see that in a 1.4 trace |
| 14:50.26 | WIMPy | And I expect it to take at least 8 more versions until it meets my definition of "works". |
| 14:50.54 | oej | wtfitsme: No, I have no insights into that part. Sorry. |
| 14:51.02 | wtfitsme | oej: thanks |
| 15:01.02 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:10.18 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:17.10 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:32.39 | *** join/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 15:36.14 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 15:38.00 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:42.47 | *** join/#asterisk din3sh (din3sh@41.136.81.237) |
| 16:02.15 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:02.49 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.12) |
| 16:08.43 | *** join/#asterisk curfont (~q@87.117.198.78) |
| 16:09.16 | curfont | Assuming I have some OpenVox A400E with 4xFXO (red modules), why does dahdi_genconf put them as fxs? |
| 16:10.00 | *** join/#asterisk superstraw (~user.name@unaffiliated/superstraw) |
| 16:10.05 | *** part/#asterisk superstraw (~user.name@unaffiliated/superstraw) |
| 16:11.14 | rjvvliet | curfont, if i'am correct , a FXO port uses FXS signalling. |
| 16:12.52 | curfont | Oh I see rjvvliet, so "FXO" modules will appear as FXS signalling? |
| 16:13.09 | curfont | And FXO modules are to connect my asterisk to the PSTN "plug in the wall", correct? |
| 16:13.40 | curfont | makes sense, because the context says "from-pstn" |
| 16:13.56 | rjvvliet | FXO is for an Office connection, FXS is for a Station connection. |
| 16:14.00 | Qwell | ~fxofxs |
| 16:14.00 | infobot | fxofxs is probably An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 16:14.07 | Qwell | infobot: really? |
| 16:14.07 | infobot | REALLY! |
| 16:14.09 | rjvvliet | thanks Qwell |
| 16:14.35 | curfont | cool, cool |
| 16:14.36 | Qwell | stupid lag |
| 16:14.51 | rjvvliet | qas doing it fron head, alsway a nice checkup ;-) |
| 16:14.51 | *** join/#asterisk iPod-nano (~davey@c-98-209-216-68.hsd1.mi.comcast.net) |
| 16:14.57 | curfont | had to blacklist netjet and it worked |
| 16:15.34 | curfont | btw, do I need to initialise wctdm always to the country? |
| 16:15.38 | curfont | or can it read from a conf? |
| 16:16.30 | rjvvliet | curfont: sorry cant help you with that. |
| 16:16.42 | curfont | np |
| 16:16.43 | iPod-nano | Hey, I'm running Asterisk 1.6 on a Debian box and setting the bind port to anything other than 5060 results in nothing. Asterisk doesn't even show a device trying to connect, and yes, my PAP2 was configured to the new port. Any ideas? |
| 16:17.00 | curfont | iPod-nano: iptables? |
| 16:17.12 | rjvvliet | iPod-nano: just a blind though firewall. |
| 16:17.17 | rjvvliet | oeps to late...... |
| 16:17.18 | Qwell | ~upgrade asterisk |
| 16:17.18 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 16:17.47 | curfont | I am running Asterisk 10.5.1 :D |
| 16:18.33 | curfont | oh wait, 10.6 is standard now |
| 16:18.44 | Qwell | curfont: days behind |
| 16:18.47 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 16:18.51 | curfont | yesterday? |
| 16:18.51 | curfont | :P |
| 16:19.05 | iPod-nano | Oh really? Upgrading will magically fix what is likely a blocked port issue. Come on, that's kinda ridiculous. |
| 16:19.21 | curfont | might upgrade later |
| 16:19.37 | iPod-nano | The #debian channel was no help, I'm inexperienced with iptables. |
| 16:19.38 | Qwell | iPod-nano: What upgrading will do is make people here care slightly more. |
| 16:19.44 | curfont | if I make install, will it break everything or will it cleanly upgrade? |
| 16:20.04 | [TK]D-Fender | curfont, Show us the problem. |
| 16:20.15 | [TK]D-Fender | Show us that it's listening. Show us the CPDUMP of the inbound attempt |
| 16:20.17 | [TK]D-Fender | ~pb |
| 16:20.18 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 16:20.19 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:20.28 | curfont | [TK]D-Fender: are you targetting the wrong person? |
| 16:20.39 | curfont | you mean "iPod-nano" |
| 16:20.56 | [TK]D-Fender | Yes, I am |
| 16:21.02 | iPod-nano | OK, new question: how did we jump from 1.8 to 10.6? |
| 16:21.15 | Qwell | ~asterisk10 |
| 16:21.15 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/, or a Standard Release. It was released on 2011-12-15, with maintenance until 2012-12-15. Asterisk 10 will be end of life on 2013-12-15. |
| 16:21.15 | [TK]D-Fender | iPod-nano, quickly |
| 16:21.19 | iPod-nano | Aren't we missing a few versions, guys? |
| 16:21.32 | Qwell | keep up - that was a year ago |
| 16:21.46 | curfont | so did firefox |
| 16:21.50 | [TK]D-Fender | iPod-nano, Windows 98 -> Windows 2000. a 1902 difference. |
| 16:22.14 | [TK]D-Fender | iPod-nano, Get over the number chanegd. Slackware jumped to 13.37 <- |
| 16:22.35 | anonymouz666 | Slackware jumped from 4 to 7 |
| 16:22.40 | iPod-nano | Not that I'm defending Microsoft, but the 98 was abbreviated where Windows 00 would've made no sense, really only a two year difference. |
| 16:23.46 | [TK]D-Fender | iPod-nano, Yes they jumped. There. We're done. Lets mov on. |
| 16:23.56 | [TK]D-Fender | iPod-nano, No, your version is not at fault. Show us EVIDENCE |
| 16:24.08 | [TK]D-Fender | iPod-nano, And maybe we can pin down what's going wrong |
| 16:24.33 | curfont | I think thats his point [TK]D-Fender, he cant find evidence of an inbound request |
| 16:24.40 | iPod-nano | Well that's the thing, nothing at all happens in the logs. |
| 16:24.45 | curfont | So there are two possibilities |
| 16:24.50 | curfont | 1. Your client isnt connecting |
| 16:24.52 | curfont | 2. Firewall |
| 16:24.59 | iPod-nano | Which is why I think there's something blocking the port. |
| 16:25.16 | curfont | Or you havent set it to listen to other ports? |
| 16:25.19 | iPod-nano | But the #debian guys were really of no help and suggested coming in here. |
| 16:25.22 | curfont | Maybe reload/restart? |
| 16:26.19 | iPod-nano | Actually, curfont, Asterisk wasn't even running when I changed the config file, so I started it up anew, after changing my PAP2 to the new port, of course. |
| 16:27.01 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:27.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:27.30 | iPod-nano | I thought I disabled all the firewall nonsense when I installed Debian, but maybe I'm wrong. |
| 16:28.00 | [TK]D-Fender | Disto isn't the issue |
| 16:28.05 | [TK]D-Fender | FIREWALL. |
| 16:28.12 | [TK]D-Fender | packets make it or they don't |
| 16:28.31 | iPod-nano | I've been suspecting a firewall issue all along, but I have no idea how to use iptables. |
| 16:31.37 | [TK]D-Fender | iptables --list |
| 16:31.39 | [TK]D-Fender | ~pb |
| 16:31.39 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 16:31.42 | [TK]D-Fender | ^^ |
| 16:34.28 | iPod-nano | http://pastebin.com/nuFbwwPt |
| 16:36.06 | jpsharp | That's a wide open iptables. |
| 16:36.35 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:36.41 | iPod-nano | And... wouldn't that mean nothing should be blocked? |
| 16:37.02 | jpsharp | Nothing is blocked. |
| 16:37.14 | *** join/#asterisk outtolunc (me@c-76-21-78-122.hsd1.ca.comcast.net) |
| 16:37.26 | kaldemar | iPod-nano: "netstat -unlp" <-- is asterisk listening on the port you configured it to bind to? actually, how did you configure the port? |
| 16:38.11 | iPod-nano | kaldemar, I went into sip.conf and under [general] I put port=XXXX. |
| 16:38.22 | kaldemar | iPod-nano: that is not a valid parameter. |
| 16:38.41 | *** join/#asterisk citrusfizz (~IRC@c-50-137-218-206.hsd1.mn.comcast.net) |
| 16:39.02 | iPod-nano | Um... I mean bindport. |
| 16:41.26 | *** join/#asterisk PipBoy (PipBoy@ip067.216.145.100.nas.net) |
| 16:42.45 | kaldemar | iPod-nano: well, is asterisk listening on the port you conrfigured with bindport? does "module show like chan_sip" list chan_sip.so as loaded? |
| 16:43.03 | Katty | hello my asterisk does not work at all how to fix plz?? it is beeping??? |
| 16:43.56 | iPod-nano | Yes to both. |
| 16:44.45 | *** join/#asterisk GameGamer43 (users.5533@id-5533.hampstead.irccloud.com) |
| 16:45.44 | kaldemar | iPod-nano: what is between asterisk and the PAP2? |
| 16:46.09 | iPod-nano | kaldemar, absolutely nothing. They're both on a local network. |
| 16:46.23 | iPod-nano | Plugged into the same switch, even. |
| 16:47.21 | iPod-nano | Just on a hunch, I'm going to power cycle everything. |
| 16:48.14 | kaldemar | enable sip debug to see any messages coming in |
| 16:51.28 | *** join/#asterisk vastina (~vastina@70.99.103.122) |
| 16:51.29 | iPod-nano | kaldemar, again, nothing at all happens in the logs. |
| 16:52.02 | jpsharp | Can you ping your ATA from the asterisk box? |
| 16:52.26 | Qwell | ~upgrade asterisk |
| 16:52.26 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 16:52.35 | Qwell | It is not a suggestion. |
| 16:52.47 | iPod-nano | jpsharp, affirmative. |
| 16:54.00 | outtolunc | wow, i see nothing much has changed ;) |
| 16:55.02 | pabelanger | huh? |
| 16:55.25 | kaldemar | iPod-nano: don't look at logs, look at CLI |
| 16:55.52 | iPod-nano | kaldemar, I misspoke, that's exactly what I've been doing. |
| 16:59.18 | kaldemar | iPod-nano: "tcpdump -ni <your_interface> host <PAP2_host> and port 5070" and try to make a call. |
| 17:01.34 | *** join/#asterisk din3sh (~din3sh@41.136.81.237) |
| 17:01.41 | din3sh | NOTICE[11345]: chan_dahdi.c:3108 my_handle_dchan_exception: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 |
| 17:01.41 | din3sh | NOTICE[11346]: chan_dahdi.c:3108 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 |
| 17:02.31 | din3sh | should i be worried about dropped calls with the 2 notices above? |
| 17:02.47 | jpsharp | Are the regular in your log? |
| 17:02.50 | jpsharp | they |
| 17:03.24 | din3sh | 10 times in a day |
| 17:03.35 | din3sh | around* |
| 17:04.11 | outtolunc | din3sh: if multiple times (not just once on startup), then first thing i would do is replace the cable betwen the NIU and the t1 interface card |
| 17:06.45 | din3sh | i get "PRI got event: HDLC Abort (6)" on span 2, span 2 is cross cable from asterisk to a meridian digital system |
| 17:06.55 | iPod-nano | kaldemar, pastebin.com/v4zz6BSG |
| 17:07.03 | din3sh | whats the reason for these notices? |
| 17:07.28 | iPod-nano | That's what happens any time I try to place a call. |
| 17:08.01 | kaldemar | iPod-nano: so the PAP2 is sending to 5060 and asterisk listens on 5070. |
| 17:08.26 | iPod-nano | The PAP2 is .81 and the server is .150 |
| 17:08.45 | kaldemar | iPod-nano: go kick the PAP2 some more. either you did not configure it right or it doesn't work right. |
| 17:12.02 | outtolunc | din3sh: the hdlc abort 6 (without a companion 8) is usually a timing issue |
| 17:12.44 | iPod-nano | kaldemar, I know I should've done this sooner but when I connected a softphone to port 5033, it worked. |
| 17:12.52 | iPod-nano | Now I'm really confused. |
| 17:13.12 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 17:13.17 | outtolunc | din3sh: timing can be affected by interrupt overload also (depending on system, and what you are doing, writing, etc) |
| 17:13.27 | iPod-nano | Ugh, now I feel stupid. |
| 17:13.29 | kaldemar | iPod-nano: why are you confused? |
| 17:14.30 | iPod-nano | The PAP2 has a "Port" field in its config page, and I've been changing that to 5033. I just cleared that field and added :5033 to the "Proxy" field. Worked. |
| 17:16.13 | iPod-nano | I guess it's true what they say: check the one thing you know can't possibly be wrong. |
| 17:16.14 | din3sh | HP DL 380, 1x Quad, 8gb memory, 30extensions as of now |
| 17:17.06 | din3sh | dahdi_tool shows no missed interrupt |
| 17:17.23 | outtolunc | din3sh: 30 extensions is hardly anything.. as for load (evening recording/encoding/database/etc) unless you have a shared interrupt for your nic |
| 17:17.42 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 17:18.13 | outtolunc | din3sh: how often do you see the abort 6 by itself? |
| 17:20.38 | outtolunc | din3sh: hopefully you aren't one of those guys that runs a gui on your asterisk box and have a hires mouse attached. ;) |
| 17:21.40 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 17:22.44 | jpsharp | My asterisk box is my desktop. Why do calls drop when I play Skyrim? |
| 17:24.29 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 17:25.02 | Hive | Is there a way to tell what channel a queue member is talking on through a CLI command? |
| 17:25.23 | Hive | *using a CLI command |
| 17:26.32 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:26.56 | jpsharp | queue show shows all the queue members and their channels |
| 17:28.05 | newtonr | and then of course from there, "core show channels" and "core show channel <channel>" |
| 17:28.16 | newtonr | to get even more detail |
| 17:28.19 | jpsharp | Or you can dump all the active channels and look to see which ones have an application of Queue |
| 17:28.29 | jpsharp | yeah, what he said. |
| 17:29.06 | Hive | hmm |
| 17:29.39 | Hive | what i'm trying to do is get the channel of each person who is on a queue call for a specific queue |
| 17:29.59 | [TK]D-Fender | queue show [queuename] |
| 17:30.00 | [TK]D-Fender | ^ |
| 17:30.08 | Hive | but then |
| 17:30.15 | Hive | I need to get their channel, so that I can chanspy them |
| 17:30.23 | [TK]D-Fender | Dump the channel list |
| 17:30.30 | Hive | and cross refrence the two? |
| 17:30.34 | [TK]D-Fender | clearly |
| 17:30.38 | Hive | such a pain |
| 17:30.41 | Hive | :P |
| 17:30.53 | [TK]D-Fender | Yes, but I have hopes you'll smoth out :) |
| 17:30.58 | [TK]D-Fender | smooth* |
| 17:30.59 | Hive | lol |
| 17:31.51 | Hive | thanks as always :D |
| 17:34.18 | Hive | Is there a way to filter 'core show channels' by apptype? |
| 17:34.28 | [TK]D-Fender | "man grep" |
| 17:37.09 | rjvvliet | Hive: i maybe late with this but does this help : http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueStatus |
| 17:37.31 | rjvvliet | the queueentry event contains a channel. |
| 17:39.37 | Hive | That looks like what I'm basically trying to do |
| 17:40.26 | Hive | queueentry event? Like the Queue(queuename) command? |
| 17:41.02 | rjvvliet | look at the page al litte doen in the sample data. it contains a QueueEntry list |
| 17:41.25 | rjvvliet | so its part of the QueueStatus AMI Action |
| 17:41.29 | rjvvliet | as a response |
| 17:41.45 | Hive | Oh i see |
| 17:42.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:47.46 | Katty | hello my asterisk does not work at all how to fix plz?? it is beeping??? |
| 17:47.57 | Hive | :P |
| 17:48.11 | newtonr | Katty: do you have a script posting that? |
| 17:48.35 | Katty | not last time i checked. |
| 17:49.01 | newtonr | ah, lol, you posted the same at (11:43:02 AM) |
| 17:49.17 | Katty | and? |
| 17:50.32 | Hive | It's still beeping! |
| 17:50.37 | Katty | YES |
| 17:50.52 | Hive | Have you tried turning your phone volume to 0, that might fix the beeping at least |
| 17:51.03 | newtonr | was just curious for now! :) |
| 17:51.29 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:51.36 | Katty | i don't like that fix. |
| 17:51.38 | Katty | try a new one. |
| 17:52.52 | Hive | Put your head next to a giant speaker, turn the volume up all the way. |
| 17:52.59 | Hive | Do that for about 10 minutes |
| 17:53.06 | Hive | Then see if you can still hear the beeping |
| 17:53.12 | Katty | >.< |
| 17:53.15 | Hive | :P |
| 17:53.42 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:53.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:53.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:55.57 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 18:00.50 | *** join/#asterisk bchia (~Adium@nat/digium/x-yaykzdbumogwlmem) |
| 18:08.56 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 18:13.04 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:15.23 | *** join/#asterisk HyperNerdV2 (~HyperNerd@winery.babcockwinery.com) |
| 18:15.31 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 18:15.50 | Ice_Strike | Hi, I have notice some files has not been recorded in the monitor folder |
| 18:16.08 | Ice_Strike | sometime it work, and sometime it doesnt |
| 18:16.11 | Ice_Strike | what is causing this? |
| 18:16.33 | Ice_Strike | in the cli, it can see it say MixMonitor |
| 18:16.48 | Ice_Strike | not its not recording into monitor folder |
| 18:23.11 | [TK]D-Fender | Where do WE see any of this? |
| 18:24.00 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 18:24.09 | Ice_Strike | [Jul 11 18:58:23] ERROR[17805]: utils.c:1130 ast_carefulwrite: write() returned error: Broken pipe |
| 18:24.16 | Ice_Strike | is this might be a problem? |
| 18:24.27 | pabelanger | fix your agi script |
| 18:24.40 | nny | i have a provider who is returning 503 on any reason it can't complete calls. This ends the channel, but I am trying to setup redundancy and add a Dial statement after to try another provider. Advice? |
| 18:25.38 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 18:26.09 | [TK]D-Fender | I've not seen a Dial() that aborts where you can't continue processing the dialplan... |
| 18:27.30 | leifmadsen | nny: tell Dial() to continue in the dialplan (there is a flag for it, I think it's 'g'), then check ${DIALSTATUS} value |
| 18:28.48 | Ice_Strike | Is there anything wrong with this dial plan? Sometime file is being recorded but in the CLI it showing its excuting MixMonitor() correctly... http://pastebin.com/wAJhRxNx |
| 18:29.15 | Ice_Strike | and sometime it doesnt record when the call answered |
| 18:29.48 | Ice_Strike | It is really strange |
| 18:29.50 | [TK]D-Fender | Well the path is rather absolute there |
| 18:30.14 | Ice_Strike | Yes it is |
| 18:30.15 | curfont | so, it seems asterisk isnt picking up my dahdi channels |
| 18:30.21 | curfont | even though genconf makes it nice |
| 18:30.51 | [TK]D-Fender | Ice_Strike, Also... that dialplan seems to imply it'll only record for... 6 SECONDS + the time some AGI runs.... which we don't see the content of |
| 18:31.26 | [TK]D-Fender | curfont, Do we see your configs for the card & dialplan and CLI output on an incoming call in a pastebin somewhere? |
| 18:31.29 | [TK]D-Fender | ~pb |
| 18:31.29 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 18:31.35 | rjvvliet | curfont: if i'am not mistaken genconf craeet a dahdi_channels.conf file wich you shold include in dahdi.conf |
| 18:31.49 | curfont | aha, good point rjvvliet |
| 18:31.50 | Ice_Strike | AGI is just execute to the database that is all |
| 18:31.51 | curfont | thanks |
| 18:32.10 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:32.26 | Ice_Strike | [TK]D-Fender Indeed I do want to just record for 6 seconds which it work when I make a call. It just that some files is not recorded which is strange. |
| 18:32.29 | Ice_Strike | AGI is just execute to the database that is all |
| 18:32.44 | curfont | rjvvliet: "echo "#include dahdi-channels.conf" >> /etc/asterisk/chan_dahdi.conf" |
| 18:32.46 | curfont | in fact |
| 18:34.06 | nny | leifmadsen: thanks |
| 18:34.18 | rjvvliet | curfont: Yep, that should be it... |
| 18:36.11 | curfont | ERROR[2729]: chan_dahdi.c:16996 build_channels: Unable to reconfigure channel '1' |
| 18:36.13 | curfont | oops |
| 18:36.24 | curfont | something wrong in my config possibly |
| 18:37.31 | rjvvliet | curfont: you may have a double channel mentioned |
| 18:37.50 | rjvvliet | you cannot reconfigure a channel if already configure earlyer.. |
| 18:38.12 | rjvvliet | maybe a PB helps |
| 18:38.19 | curfont | rjvvliet: it needed a reboot |
| 18:38.22 | curfont | instead of a reload |
| 18:38.33 | rjvvliet | aaa, thats also true. |
| 18:39.02 | rjvvliet | curfont: Sorry i'am not using DAHDI that much, i'am only learning it for my DCAP in a few months... ;-) |
| 18:39.19 | curfont | everything works now :D |
| 18:39.22 | curfont | except this warning |
| 18:39.23 | curfont | WARNING[3157]: chan_dahdi.c:4981 dahdi_train_ec: Unable to request echo training on channel 3: Invalid argument |
| 18:39.40 | curfont | Executing [s@from-pstn:4] Dial("DAHDI/3-1", "SIP/cisco230,60") in new stack |
| 18:39.42 | curfont | yay though |
| 18:39.45 | *** join/#asterisk mahlon (mahlon@198.145.180.83) |
| 18:39.46 | rjvvliet | curfont: i'ts the echo config, |
| 18:40.15 | Ice_Strike | [TK]D-Fender -- Executing [callrecord@classification:4] MixMonitor("SIP/scd-000005ed", "/var/spool/asterisk/monitor/num_ib/classification/temp_0dgg3gs.wav") in new stack |
| 18:40.21 | curfont | echotraining=800 |
| 18:40.23 | curfont | this? |
| 18:40.34 | curfont | Got the info from: http://wiki.openvox.cn/index.php/OpenVox_A400E_User_Manual_for_dahdi |
| 18:40.41 | curfont | probably for older asterisk though |
| 18:40.41 | Ice_Strike | [TK]D-Fender its not appearing in var/spool/asterisk/monitor/num_ib/classification/ |
| 18:40.47 | Ice_Strike | when i try again, it work |
| 18:40.49 | Ice_Strike | why? |
| 18:41.03 | rjvvliet | curfont: echo is mostly card depandand, |
| 18:41.13 | curfont | my card is a400e heh |
| 18:41.20 | curfont | same as the wiki says |
| 18:41.50 | rjvvliet | curfont: mmmm. |
| 18:43.04 | *** join/#asterisk Neptu (~Neptu@41.251.190.138) |
| 18:47.30 | rjvvliet | curfont: does the carddriver and the DAHDI version correct? |
| 18:47.50 | rjvvliet | curfont: do they match according to the manufacturer |
| 18:51.52 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-127.mobile.uci.edu) |
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| 18:55.10 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-77-197.pppoe.mtu-net.ru) |
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| 19:44.45 | Katty | hellllllllloooooooooooo nurse. |
| 19:45.19 | WIMPy | Di you pay extra for a night nurse? |
| 19:48.13 | *** join/#asterisk asilva (~asilva@gandalf.ai.unesp.br) |
| 19:48.49 | asilva | Hello, does anyone knows where to find a good how to for building an Asterisk - HA ? |
| 19:49.44 | PipBoy | well.. the easy way is to just virtualize it :P assuming you take care of timing |
| 19:50.27 | *** join/#asterisk HyperNerdV2 (~HyperNerd@winery.babcockwinery.com) |
| 19:50.49 | asilva | Already use asterisk in virtual enviroment? my problem is how to monitor both sides and share informations? and use a ACTIVE / PASSIVE model |
| 19:51.19 | *** join/#asterisk Neptu (~Neptu@41.251.190.138) |
| 19:51.23 | PipBoy | well i was suggesting to use your hypervisor to just do HA on the vm |
| 19:51.30 | PipBoy | thats what I do |
| 19:52.01 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 19:56.58 | asilva | PipBoy: you use VMWare ? |
| 19:58.07 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 19:59.00 | pabelanger | Well that sucks, the SPA942 will not use time-offset from dhcp |
| 19:59.59 | WIMPy | The only thing that doesn't suck about the SPAs is the hardware I think. |
| 20:00.02 | nny | pabelanger: odd, thought that was suppose to work.. i should confirm it works with the newer cisco spa models |
| 20:00.30 | nny | WIMPy: dunno, I have ~ 300 504gs out in the field on various setups, all seem to work happily |
| 20:01.25 | WIMPy | I was told the software hasn't changed much to the older models and they were at the edge of usable. |
| 20:03.49 | leifmadsen | I have a spa941 or 942 in my basement that I haven't used in years |
| 20:04.01 | leifmadsen | same with a unison sip phone |
| 20:04.21 | WIMPy | I've got the SPA962 and that's also only catching dust. |
| 20:04.23 | carrar | What else is in the basement/ |
| 20:04.25 | carrar | ? |
| 20:04.41 | leifmadsen | cisco 7960, polycom ip500 |
| 20:04.57 | leifmadsen | I probably have 4-5 sip phones in a box down there |
| 20:05.00 | carrar | should probably move |
| 20:05.04 | leifmadsen | I just moved here :) |
| 20:05.07 | carrar | heh |
| 20:05.19 | leifmadsen | they really just need to be recycled or provided to another home |
| 20:05.37 | nny | WIMPy: the cisco spa firmware has been fairly modified. For example you can edit the soft button order etc on a cisco SPA phone. |
| 20:05.44 | _Corey_ | leifmadsen: you still in canada? |
| 20:05.55 | carrar | mine the precious metals out of them |
| 20:06.20 | leifmadsen | _Corey_: aye |
| 20:06.29 | _Corey_ | cool... |
| 20:06.49 | WIMPy | nny: Can you easily change between calls now, no matter what account they are linked to? Does it still ist all calls as missed calls, even when asnwered? |
| 20:06.54 | *** join/#asterisk jovy88 (~jaymurtii@99-159-68-212.lightspeed.tukrga.sbcglobal.net) |
| 20:07.12 | pabelanger | Yup. Lame SPA942! Y u no work like polycom |
| 20:07.13 | WIMPy | (answered on another phone that is) |
| 20:07.29 | leifmadsen | I thought that was an asterisk feature |
| 20:07.37 | leifmadsen | well the phone has to support it |
| 20:07.41 | leifmadsen | but thought that was post 1.4 |
| 20:08.26 | WIMPy | There had been patches for 1.4 I think. But I've never been that good in history :-) |
| 20:08.29 | nny | WIMPy: the first part I have 8 accounts on my 962 that I can place on hold and pick up, not sure if that is what you mean. The missed call thing? No, matter of fact I'm not sure how a phone is suppose to address that properly |
| 20:08.58 | nny | WIMPy: er the 3nd part yes, it still shows a missed call as missed even when another extension answers it |
| 20:09.02 | nny | 2nd* |
| 20:09.05 | nny | typing fail today |
| 20:09.14 | leifmadsen | nny: he means if you call multiple phones, and another device picks it up, that the missed calls are not shown on the phone |
| 20:09.35 | WIMPy | nny: Handling of calls on the same account and on another account was done differently and changing to a call on another account with two calls was impossible. |
| 20:10.12 | WIMPy | Or you first had to change to the wrong call. |
| 20:11.02 | nny | WIMPy: not sure what you mean by changing. If I call my phone on 2 different accounts, I can answer each and put each on hold. i can't make them interact directly however |
| 20:11.10 | WIMPy | But I had been warned about Sipura, just like infobot does about Grandstream. |
| 20:11.39 | nny | leifmadsen: WIMPy afaik the cisco still shows up as unanswered if another phone answers it. Does polycom handle this gracefully? (if so, how?) |
| 20:11.52 | WIMPy | nny: The issue comes when you have two calls on hold on one account and an active call on another account. |
| 20:12.06 | nny | WIMPy: stand by i'll test to confirm |
| 20:12.23 | WIMPy | The bad thing is that it can't handle more than two calls on one account, either. |
| 20:13.02 | WIMPy | Someone mentioned not long ago that Polycom has the same issue, but I have no idea if that's current info. |
| 20:15.29 | nny | WIMPy: i'll test both, stand by |
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| 20:16.42 | nny | WIMPy: i'm curious about the answered calls thing though, I agree that it's annoying, was unware other phones handled it more gracefully. I wasn't clear how they would know another phone picked up |
| 20:17.01 | WIMPy | Asterisk tells them. |
| 20:17.44 | jeffspeff | so, i've made a context that reads some info and sets as vars, i've got a php script in the agi-bin dir. when i try to call the php via AGI and pass the vars to it i get an error that it failed to execute. here's the php and the extensions context http://pastebin.com/mA3HV6bU |
| 20:17.53 | WIMPy | Just the same way it has worked in the PSTN for ages. |
| 20:19.06 | nny | WIMPy: is it part of the SIP session? |
| 20:19.15 | nny | WIMPy: like (some code) another phone picked up? |
| 20:19.31 | WIMPy | Yes. It's in the BYE. |
| 20:19.57 | [TK]D-Fender | jeffspeff, You shuold not have any of that HTML crap before your PHP code |
| 20:20.52 | [TK]D-Fender | jeffspeff, and as you're calling that script as though it's executable, You have none of the standard header to indicate the parser |
| 20:20.55 | [TK]D-Fender | ~book |
| 20:20.55 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:21.08 | [TK]D-Fender | ^^^ read up on your PHP AGI basics |
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| 20:22.40 | nny | WIMPy: Firmware version 7.4.7 includes support for the SIP “Reason” header in CANCEL messages. No configuration is required. |
| 20:22.56 | nny | WIMPy: so short answer, the newer cisco spa phones do what the others already did ;) |
| 20:23.13 | nny | but the 962 doesn't, the linksys SPA phones are pretty much obsolete |
| 20:23.16 | WIMPy | Right. CANCEL, not BYE. |
| 20:23.31 | jeffspeff | [TK]D-Fender, ok, i'll check it out |
| 20:23.32 | jeffspeff | ty |
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| 20:24.09 | WIMPy | It's so annoying that so many stuff you took for granted so far doesn't work any more on the VOIP front. |
| 20:24.33 | nny | WIMPy: if you know of any other bumps with the older SPAs let me know and I can test. I'd hate to be a proponent of a specific piece of hardware only to learn I has the stupid later |
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| 20:26.38 | WIMPy | The user experience was rather iffy to me. There are probaly lots of things I suppressed. |
| 20:29.10 | WIMPy | Oh, I'm not sure if I ever found a "dialplan" for the SPA that lets you dial #. |
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| 20:30.02 | WIMPy | And off course no overlap dialling support, which would make all those kind of issues go away by itself. |
| 20:30.54 | WIMPy | themselfes. |
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| 20:33.55 | nny | WIMPy: yeah the dialplan strings have always been awkward with them. I'll test some more, haven't personally found any real issues with them, always good to know what I might be missing though. |
| 20:35.23 | WIMPy | The whole concept of dialplans in the phone is weired. |
| 20:35.48 | nny | WIMPy: agreed |
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| 20:44.46 | Hunterkll | my problems earlier were not asterisk config related |
| 20:44.48 | Hunterkll | it was all firewall |
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| 20:56.49 | Katty | hello my asterisk does not work at all how to fix plz?? it is meowing?!?! |
| 20:57.14 | WIMPy | Feed it. |
| 20:57.28 | Katty | adds to list |
| 20:57.52 | Katty | so far i think i like pour drano into system the best |
| 21:02.57 | _Corey_ | Katty: how did that Drano work out for you yesterday? |
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| 21:43.57 | curfont | ok, why in some iax2 links i need to specify the user/pass in the extensions and in others i dont? |
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| 22:28.03 | TechSmurf | Can anyone suggest a good audio editor for system recordings? |
| 22:28.21 | WIMPy | uses audacity |
| 22:31.29 | nny | son of a b&@*&! |
| 22:31.46 | nny | wonders when vitelity capped their per minute usage out at 10 channels |
| 22:31.51 | jpsharp | audacity is the way to go. |
| 22:32.08 | TechSmurf | Is there any way to make audacity play nicer with gsm and ulaw files? |
| 22:32.14 | nny | the must HATE money |
| 22:32.17 | nny | they* |
| 22:34.03 | jpsharp | Or they hate getting stiffed with large bills when someone's system gets rooted. |
| 22:38.13 | nny | jpsharp: that doesn't sound like something that should penalize all of their customers |
| 22:38.55 | jpsharp | No, but you could probably call them and say 'HEY! Cut that out on my account'. |
| 22:39.10 | nny | jpsharp: yup. I just have to commit to 250 a month or more as a reseller first |
| 22:39.15 | nny | jpsharp: not worth the effort |
| 22:39.29 | nny | jpsharp: considering how many minutes that is vs. actual usage |
| 22:39.49 | TechSmurf | takes the silence as a no |
| 22:40.11 | jpsharp | TechSmurf: I've always done the editing in Audacity, then used SOX to convert to GSM. |
| 22:40.12 | nny | TechSmurf: thought audacity worked with ulaw |
| 22:40.43 | nny | maybe not. I just work with WAV and convert as needed |
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| 22:43.16 | vastina | ls |
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| 22:45.12 | nny | [vastina@vastinaspentiumbeast]# porn moreporn hiddenporn cookierecipes |
| 22:46.21 | TechSmurf | I'm trying to stitch together some of the asterisk core/extra sounds, but audacity lacks drag/drop for it |
| 22:47.00 | nny | TechSmurf copy/paste in audacity |
| 22:47.27 | TechSmurf | I just grabbed an older version of wavepad... it's handling it a bit better |
| 22:47.52 | TechSmurf | audacity won't even open the files right... keeps forcing me to import each one |
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| 22:50.01 | TechSmurf | hmm |
| 22:50.12 | TechSmurf | or I guess I could be smart and download the core/extra sound packs in wav |
| 22:50.15 | TechSmurf | idjit |
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| 22:57.58 | vastina | nny: wtf? |
| 22:58.15 | vastina | i was in the wrong window, cascaded wm, :0 |
| 22:58.54 | vastina | pentiumbeast |
| 22:58.55 | vastina | lol |
| 23:00.01 | vastina | and you put me in a root session, nice |
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| 23:37.02 | generalhan_ | anyone here familar with loading firmware on Xorcom Astribank 2 models ? I have looked all over, and tried several things to no avail. |
| 23:37.32 | generalhan_ | i saw a post that mentioned a new version of xpp_fxloader, which i picked up ... but it fails when i try to load |
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| 23:43.46 | generalhan_ | if anyone has any suggestions please let me know, i am at a complete loss. |
| 23:44.47 | generalhan_ | running asterisk version 1.8.13.1 and dadhi complete 2.6.1 -- if that helps at all |
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| 23:52.18 | sruffell | tzafrir_laptop: ^^ are you around by any chance? |
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