| 00:01.35 | TheRedOctober | Greetings all. I keep getting a SIP 405 Response: Method Not Allowed on an attempted SIP trunk from a CUCM7 box as provider...can anyone suggest why I am getting this? Authentication is set up properly, from what I can tell |
| 00:09.09 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 00:10.37 | *** join/#asterisk MACscr (~Adium@c-98-214-103-147.hsd1.il.comcast.net) |
| 00:26.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 00:27.48 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 00:27.55 | *** part/#asterisk MACscr (~Adium@c-98-214-103-147.hsd1.il.comcast.net) |
| 00:35.03 | saxa | still without audio on his nat problem :( |
| 00:40.13 | p3nguin | Did you ever check for ALG? |
| 00:42.15 | ChrisInSydney | Good point |
| 00:56.24 | WIMPy | TheRedOctober: I had to at least disable qualify and session timers. |
| 01:15.01 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 01:16.55 | saxa | p3nguin: yes, i have tried with SIP ALG enabled and disabled. |
| 01:17.18 | p3nguin | Always disable it if you intend to let Asterisk control the NAT stuff. |
| 01:17.36 | p3nguin | Did you ever pastebin your configs? |
| 01:49.42 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 01:53.21 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 01:56.18 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 02:02.38 | ChrisInSydney | p3nguin: Had a strange thing. Something in the #dd-wrt channel caused my x-chat2 to exit. Had to clear the history. Wierd, but not suprising |
| 02:05.05 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 02:14.04 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 02:15.15 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 02:17.55 | SeRi | p3nguin: you in? |
| 02:17.55 | p3nguin | SeRi, Leave a message after the beep. *BEEP* |
| 02:18.25 | ChrisInSydney | SerRi: Tell p3nguin the bridge is still up |
| 02:19.07 | ChrisInSydney | :-) |
| 02:21.50 | SeRi | lol |
| 02:25.26 | p3nguin | I know it's up; I hear your musicz. |
| 02:25.50 | ChrisInSydney | Al Kooper |
| 02:25.54 | SeRi | p3nguin: whats going on! |
| 02:35.12 | p3nguin | I'm fighting with a Bourne shell problem. |
| 02:36.10 | WIMPy | Give it a Bourne Ultimatum to comply. |
| 02:36.21 | p3nguin | I have this shell script that I am working with... I have six systems, and the script works perfectly fine on five of them. |
| 02:37.32 | p3nguin | Four GNU/Linux, all work well; two FreeBSD, one works well, one blows up. |
| 02:38.00 | p3nguin | On FreeBSD 6.4, it works well. On FreeBSD 7.3, it blows up. |
| 02:42.40 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 02:45.21 | p3nguin | Now it's really pissing me off. I wrote a new script just for testing purposes, and it seems to work fine! I'm using the same type of format, even. |
| 02:53.23 | ChrisInSydney | is it /r/n ? |
| 03:00.51 | *** join/#asterisk chendy (~chatzilla@204.45.134.74) |
| 03:08.07 | p3nguin | I'm not sure what you mean by that. |
| 03:08.38 | p3nguin | The problem I'm having with the one script is pertaining to \r |
| 03:08.42 | ChrisInSydney | do you have a carrage return / line feed in the file. Its got be that way a few times. |
| 03:08.54 | ChrisInSydney | dos v unix format |
| 03:09.06 | ChrisInSydney | got me |
| 03:09.09 | p3nguin | I just press enter on the end of the line. I'm using vim. |
| 03:09.26 | ChrisInSydney | well I'm fresh out of ideas then |
| 03:09.39 | p3nguin | I'm going to do more testing and possible recreate the conf from scratch. |
| 03:12.20 | p3nguin | ++ $'\r' |
| 03:12.44 | p3nguin | When a variable is null, that's what it shows me if I run the script with bash -vx. |
| 03:16.22 | *** join/#asterisk irishpilot (~irishpilo@122-59-153-152.jetstream.xtra.co.nz) |
| 03:17.00 | p3nguin | Solved! I guess the conf got some special characters from the pastebin. I deleted it and recreated it by hand, and now it works. |
| 03:17.24 | ChrisInSydney | could have been a stray |
| 03:18.01 | irishpilot | Hey guys |
| 03:18.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:19.02 | p3nguin | What's weird is that I pastebinned the files from one system and downloaded it in raw format from the pastebin on both of those FreeBSD systems. |
| 03:19.30 | p3nguin | One worked, one choked. |
| 03:26.30 | *** join/#asterisk radic (~radic@dslb-178-002-214-066.pools.arcor-ip.net) |
| 03:30.07 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 03:36.13 | irishpilot | anybody know anything about using Originate Action |
| 03:36.17 | irishpilot | oh the Manager AMI |
| 03:37.54 | [TK]D-Fender | So where re you stuck on it now? |
| 03:38.01 | irishpilot | hey Fender |
| 03:38.12 | irishpilot | wow thought you'd be asleep as it's daytime here in New Zealand |
| 03:38.39 | irishpilot | I am :( |
| 03:38.56 | irishpilot | I am writing the Java part to call the ChanSpy |
| 03:39.04 | irishpilot | so been reading over and over our conversation last night |
| 03:39.15 | irishpilot | my problem is originateAction.setChannel(callerChannel); |
| 03:39.53 | irishpilot | callerChannel needs to be a SIP/1234 sort of thing |
| 03:40.18 | irishpilot | my understanding was that I want to break into our Dial() channels |
| 03:40.41 | irishpilot | so I should be calling a "channel" not an extension SIP/1234 |
| 03:41.08 | irishpilot | remember this: "[TK]D-Fender: irishpilot, "Originate "Hey, left side, call this dialplan that does chanspy". OK "Well that force answered, didn't it? OK, go to your DIALPLAN I pointed you to". What is there? PLAYBACK. What else is there? Nothing? (DIES)" |
| 03:41.49 | irishpilot | sure I can use Originate to call an extension but not an active channel |
| 03:41.54 | irishpilot | or am I totally ost? |
| 03:41.55 | irishpilot | lost |
| 03:43.14 | irishpilot | the man says Originate(tech_data,type,arg1[,arg2[,arg3]]) |
| 03:43.26 | irishpilot | for the DialPlan |
| 03:44.06 | irishpilot | the AMI man page says: Channel - Channel name to call. |
| 03:44.31 | irishpilot | so maybe I can originate to an already active channel |
| 03:44.42 | [TK]D-Fender | No |
| 03:45.13 | [TK]D-Fender | channel: Local/spy@spycontext/n |
| 03:45.29 | [TK]D-Fender | [spycontext] |
| 03:45.45 | [TK]D-Fender | exten => spy,1,Answer |
| 03:45.58 | [TK]D-Fender | exten => spy,1,ChanSpy(thechannel goes here) |
| 03:46.06 | [TK]D-Fender | That is the left side |
| 03:46.14 | irishpilot | ok all cool with that |
| 03:46.17 | p3nguin | s/1/n/ |
| 03:46.22 | [TK]D-Fender | This "call " is dialplan on BOTH sides |
| 03:47.41 | [TK]D-Fender | Yeah, fix the prio's |
| 03:48.14 | irishpilot | So this is my context done |
| 03:48.18 | irishpilot | context spycontext { |
| 03:48.19 | irishpilot | <PROTECTED> |
| 03:48.19 | irishpilot | <PROTECTED> |
| 03:48.20 | irishpilot | <PROTECTED> |
| 03:48.20 | irishpilot | <PROTECTED> |
| 03:48.20 | irishpilot | <PROTECTED> |
| 03:48.20 | irishpilot | <PROTECTED> |
| 03:48.21 | irishpilot | } |
| 03:48.24 | [TK]D-Fender | irishpilot: PASTEBIN. |
| 03:48.25 | p3nguin | PASTEBIN |
| 03:48.29 | [TK]D-Fender | Never flood like that |
| 03:48.55 | irishpilot | sorry was just showing the 7 lines I did |
| 03:49.01 | irishpilot | didn't mean to flood ya |
| 03:49.03 | p3nguin | ~pb |
| 03:49.03 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 03:49.28 | [TK]D-Fender | irishpilot: You did too muchin there. the left side is ONLY the spy |
| 03:50.00 | irishpilot | oh ok |
| 03:50.17 | irishpilot | so I just ChanSpy there and remove the rest |
| 03:50.24 | [TK]D-Fender | answer, spy |
| 03:50.27 | [TK]D-Fender | that's all. |
| 03:50.31 | irishpilot | ok |
| 03:50.32 | irishpilot | done |
| 03:50.36 | irishpilot | the channel for there |
| 03:50.41 | irishpilot | not sure yet which one it will be |
| 03:50.45 | [TK]D-Fender | the other side waits 1-2 sec, then do a Playback for your warning. |
| 03:50.48 | irishpilot | but will come back to it |
| 03:50.50 | [TK]D-Fender | or integrate the dealy into it |
| 03:50.56 | irishpilot | Im starting to get ya |
| 03:51.02 | p3nguin | <irishpilot> so I should be calling a "channel" not an extension SIP/1234 <-------- SIP/1234 is a channel, not an extension. |
| 03:51.17 | [TK]D-Fender | That is a device |
| 03:51.18 | irishpilot | yep |
| 03:51.23 | [TK]D-Fender | A local channel is a device as well. |
| 03:51.37 | [TK]D-Fender | You can have 1 set of dialplan talking to another. |
| 03:51.47 | irishpilot | I didn't realise that |
| 03:51.58 | irishpilot | so now I create another dialplan |
| 03:52.16 | [TK]D-Fender | with the other half |
| 03:54.32 | irishpilot | I think I have it |
| 03:54.33 | irishpilot | http://pastebin.com/62Y2i0dV |
| 03:55.25 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
| 03:55.29 | irishpilot | (Im liking this paste bin stuff, apologies for flooding the code earlier, it's been 15 years since I've used IRC) |
| 03:55.50 | p3nguin | We didn't like flooding back then, either. |
| 03:56.07 | irishpilot | What I mean is back then there was no such thing as a paste bin, cool concept |
| 03:56.14 | p3nguin | I see. |
| 03:57.48 | [TK]D-Fender | I miss the days of slowing down so as not to fill the teletype buffer :) |
| 03:58.09 | p3nguin | Buffer Overflow |
| 03:58.12 | p3nguin | *crash* |
| 03:58.30 | irishpilot | the old 14,4 modem |
| 03:59.13 | irishpilot | where I lived in the far beyonds of Ireland you would have to wait about 10 seconds for a line of text to even send as the connection was so slow |
| 04:00.41 | irishpilot | are we on the same page with that contact fender, I reckon we are :) |
| 04:02.02 | irishpilot | one is spying and the other is playing |
| 04:03.29 | irishpilot | I just originate a call now to both I guess from my "watcher" script |
| 04:03.58 | irishpilot | and then Bridge Action them together perhaps |
| 04:04.23 | irishpilot | although no you said not to use Bridge Action last night |
| 04:04.23 | ChrisInSydney | <p3nguin>. Just got some feedback on the call |
| 04:04.32 | ChrisInSydney | someone is there |
| 04:06.33 | *** join/#asterisk picard276 (~chatzilla@74.60.166.135) |
| 04:06.39 | [TK]D-Fender | No. |
| 04:06.52 | picard276 | hey guys having a problem wiht .call files |
| 04:07.00 | picard276 | im trying to process more than one at once |
| 04:07.11 | picard276 | but .. if i run a call file... asterisk will not process the next call file until that one is completed? |
| 04:07.23 | [TK]D-Fender | Channel =LEFT SIDE. Context,Exten, Priority, = RIGHT SIDE |
| 04:07.39 | picard276 | i have to manually set the channel? |
| 04:07.42 | [TK]D-Fender | picard276: No, |
| 04:07.47 | [TK]D-Fender | picard276: that wasn't for you |
| 04:07.50 | picard276 | ohh kk' |
| 04:07.55 | [TK]D-Fender | irishpilot: that was for you. |
| 04:08.01 | irishpilot | yep :O) |
| 04:08.07 | [TK]D-Fender | picard276: Same topic though.... * will take them ALL at once |
| 04:08.24 | picard276 | but like i drag the callfile into the spool.. its running |
| 04:08.24 | [TK]D-Fender | picard276: And potentially congest |
| 04:08.34 | picard276 | and then if i drag another one... (just one more call) it wont work |
| 04:08.38 | picard276 | it seems to only do one at a time |
| 04:08.41 | picard276 | (only trying 2 calls) |
| 04:08.47 | [TK]D-Fender | No, and what the hell is "drag"? |
| 04:08.51 | picard276 | copy them |
| 04:08.53 | picard276 | srry |
| 04:08.57 | picard276 | into the spool |
| 04:09.00 | [TK]D-Fender | NEVER copy, only MOVE |
| 04:09.10 | picard276 | ok... ive done with moving too |
| 04:09.17 | picard276 | i move them both |
| 04:09.22 | picard276 | right after each other |
| 04:09.25 | picard276 | only the first one gets processed |
| 04:09.34 | [TK]D-Fender | * will take them all unless some get locked, or the datestamp is in the future, etc |
| 04:10.36 | irishpilot | @fender: Not sure what you mean by Channel =LEFT SIDE. Context,Exten, Priority, = RIGHT SIDE. Are you saying on the Originate we set the channel to be "spyhalf" and the context extn pri as the playback message "playhalf" |
| 04:12.05 | p3nguin | I think he's referring to channel originate CHANNEL <application|extension> DATA. |
| 04:12.14 | p3nguin | Rather than Originate(). |
| 04:12.26 | [TK]D-Fender | irishpilot: Yes |
| 04:13.01 | [TK]D-Fender | irishpilot: When the Local channel you call gets answered it gets dumped where you tell it to |
| 04:13.14 | p3nguin | If you're doing it on the CLI, you don't even need the "playhalf" in an extension at all. |
| 04:13.38 | [TK]D-Fender | irishpilot: end result : dialplan talking to dialplan. However the Channel side latches on to another call, so when it gets audio... out it goes. |
| 04:13.49 | p3nguin | channel originate Local/spyhalf@spycontext/n application Playback somewarningmessage |
| 04:13.50 | [TK]D-Fender | p3nguin: No, triggered from a script |
| 04:14.04 | [TK]D-Fender | p3nguin: He's got a quirky project. |
| 04:14.11 | p3nguin | I'd guess the script still does that. |
| 04:14.24 | [TK]D-Fender | p3nguin: Payed support call, 5 minute nag to "bill for 5 more minutes" |
| 04:14.30 | picard276 | got it thanks TK |
| 04:14.35 | irishpilot | @fender: giving it a whirl brb |
| 04:14.39 | [TK]D-Fender | so press somthing to keep the call alive |
| 04:14.50 | p3nguin | But I don't know the 1st thing about AMI, so I don't really know. |
| 04:14.51 | picard276 | one more question.. this is easy... Where can i find the code for DISA online (the freepbx/asterisk) like dialplan how its passing stuff? |
| 04:15.02 | [TK]D-Fender | p3nguin: Using originate + chanspy to hook in to play the nag |
| 04:15.06 | p3nguin | If it's a shell script using asterisk -rx, that's where I do stuff. |
| 04:15.24 | p3nguin | channel originate Local/spyhalf@spycontext/n application Playback somewarningmessage <------ |
| 04:15.39 | [TK]D-Fender | p3nguin: For this aprt, its the same net effect. He does need a bit more, but I went over all of this with him 12+ hours ago |
| 04:15.56 | [TK]D-Fender | picard276: that makes no sense |
| 04:16.15 | picard276 | im looking to basically.. pass callerID and a Phone Number to a Custom context |
| 04:16.25 | picard276 | so similar to DISA (except i have the numbers already predialed... ) |
| 04:16.36 | picard276 | so i want to pass my custom Numbers and caller ID to a custom context |
| 04:16.58 | p3nguin | Where can you find it? src/asterisk-1.8.9.0/apps/app_disa.c |
| 04:18.10 | p3nguin | DISA is made to do one thing: provide you a dial tone to make a call after you are already "in" the system on a call. |
| 04:18.25 | p3nguin | If you're just making calls to numbers, you use Dial(), not DISA(). |
| 04:18.32 | [TK]D-Fender | How is a number "predialed" and what is the point of DISA? |
| 04:18.48 | [TK]D-Fender | DISA is to give you tone so you can dial. If you have the number what do you need DISA for? |
| 04:19.05 | picard276 | right... |
| 04:19.08 | picard276 | im using a call file |
| 04:19.13 | picard276 | so i have a number im passing thorugh the .call file |
| 04:19.19 | picard276 | but i want to pass that number to my custom context... |
| 04:19.31 | p3nguin | So define that context in the call file. |
| 04:19.39 | p3nguin | Context: your-context |
| 04:19.42 | picard276 | right... |
| 04:19.51 | picard276 | but when im passing to the context i want to pass a # |
| 04:19.57 | p3nguin | So do it. |
| 04:20.00 | picard276 | right.. how? |
| 04:20.04 | picard276 | in the .call file? |
| 04:20.08 | picard276 | thats where im confused |
| 04:20.13 | p3nguin | Extension: 3145555555 |
| 04:20.22 | picard276 | gotchytaa Thanks p3nguin! |
| 04:20.39 | p3nguin | Maybe you need to read the section on call files again. |
| 04:20.41 | p3nguin | ~book |
| 04:20.41 | 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 |
| 04:21.55 | p3nguin | Call files are sort of like a "spooled" version of a CLI originate. You provide a channel to attach to, then an application or extension to run once attached to the channel. |
| 04:22.58 | picard276 | k |
| 04:22.58 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 04:24.01 | p3nguin | Are you trying to make a call to one phone, then connect that call to another phone? |
| 04:28.10 | picard276 | ok that extensio nworked well p3nguin |
| 04:28.21 | picard276 | yes p3nguin |
| 04:28.23 | picard276 | the problem... |
| 04:29.09 | picard276 | when i start the call to my phone ... |
| 04:29.18 | picard276 | it automatically starts going to the extension |
| 04:29.30 | picard276 | it does not wait to call me phone... it just starts dialing the context and extension before my phone has a chacne to pick up? |
| 04:29.35 | picard276 | and i have WaitTime: 30 |
| 04:29.55 | picard276 | but i think that wait time is only to see if hte call will be connected.. it doesnt make the rest of the application wait until my call is connected... ):L |
| 04:31.49 | *** join/#asterisk bmg505 (~leon@196-209-84-238.dynamic.isadsl.co.za) |
| 04:32.23 | drmessano | hmm |
| 04:33.24 | picard276 | if u get what i am sayin |
| 04:33.40 | p3nguin | You want to call your phone, then dial another number. |
| 04:33.46 | picard276 | yes (; |
| 04:33.53 | picard276 | but its dialing before my phone gets connected |
| 04:33.56 | p3nguin | Pastebin your call file example. |
| 04:34.03 | picard276 | kk one sec |
| 04:35.11 | picard276 | http://pastebin.com/CmFqkjvf |
| 04:35.39 | p3nguin | SIP/flowroute/1949..... is your phone number? |
| 04:35.47 | picard276 | yes |
| 04:36.03 | p3nguin | And what does the from-context have in it? |
| 04:36.15 | picard276 | its just the basic thing to call out |
| 04:36.17 | picard276 | nothing specila |
| 04:36.27 | picard276 | it runs a dial(sip/flowroute/${number} |
| 04:36.33 | p3nguin | okay |
| 04:36.40 | [TK]D-Fender | And if it answers right away then that is what Flowroute does |
| 04:36.53 | picard276 | but ... |
| 04:36.58 | picard276 | if i use Application: |
| 04:37.01 | p3nguin | And it's wrong if they are doing it. |
| 04:37.11 | picard276 | in my .call file.. not using context, and Extension |
| 04:37.12 | p3nguin | I can't believe that would do that. |
| 04:37.13 | picard276 | then it works |
| 04:37.17 | picard276 | aka the call waits to be picked up... |
| 04:37.23 | picard276 | once its picked up it runs the application |
| 04:37.43 | p3nguin | s/that/they/ |
| 04:38.23 | p3nguin | It should behave the same way in either case, be it an extension or an application. |
| 04:38.29 | picard276 | right i knw |
| 04:38.31 | picard276 | ) |
| 04:38.34 | picard276 | but its not ): |
| 04:38.41 | picard276 | should i get rid of WaitTime? |
| 04:38.54 | p3nguin | Try it on the CLI with originate and see how that goes. |
| 04:39.29 | p3nguin | channel originate SIP/flowroute/1949yournum extension 1949othernum@from-context |
| 04:40.18 | picard276 | kk one sec |
| 04:40.39 | p3nguin | By the way, the "from-context" name is extremely confusing for a context intended to be used for dialing out to the PSTN. |
| 04:41.05 | p3nguin | to-flowroute, maybe... or out-context? |
| 04:41.16 | [TK]D-Fender | FreePBX <- |
| 04:41.21 | p3nguin | ick! |
| 04:41.30 | p3nguin | Nasty nasty words. |
| 04:41.32 | picard276 | works |
| 04:41.34 | picard276 | p3nguin |
| 04:41.40 | picard276 | when done from CLI works properly |
| 04:41.41 | p3nguin | The originate works correctly? |
| 04:41.46 | picard276 | yeahup (; |
| 04:41.54 | p3nguin | So what would be different using a call file? |
| 04:42.01 | p3nguin | The concept behind it is identical. |
| 04:42.04 | irishpilot | hey Fender, I've a problem. To test I made a basic dial plan to Set(__DYNAMIC_FEATURES=pleaseextend) and then a Dial(SIP/556,,U(watcher)g); All great, I call, when I answer the call on 556 it waits for the watcher script to exit - but the watcher script thread is asleep for 5 minutes. So my call never gets answered because asterisk is waiting for the watcher to finish. Is there a way to make it concurrent? I used a U to go sub, |
| 04:42.04 | irishpilot | maybe that was why you said use a macro m ? |
| 04:42.06 | picard276 | haha ? no originate |
| 04:42.10 | picard276 | ? |
| 04:44.21 | picard276 | yeah idk p3nguin? |
| 04:44.24 | picard276 | kind of wierd no? |
| 04:45.10 | [TK]D-Fender | irishpilot: Not what I told you. I gave you a complete sample already and you aren't showing what you're doing in that gosub as it is.... |
| 04:45.11 | p3nguin | Is it some kind of bug? |
| 04:45.23 | p3nguin | I don't typically use call files. |
| 04:45.34 | p3nguin | I use them, but not very often. |
| 04:45.53 | irishpilot | the go sub just goes and calls the watcher AGI, nothing special. That all works, but calls and waits for it |
| 04:46.32 | irishpilot | I am looking for the 'complete sample' you mention, Im not sure where you wrote that. Rereading your messages from today and last night |
| 04:47.12 | picard276 | its the callerID functionality |
| 04:47.13 | picard276 | ?? |
| 04:47.22 | irishpilot | yesterday you said: TK]D-Fender: irishpilot, m() only launches your external watcher and the GOES AWAY. |
| 04:47.32 | irishpilot | did you mean "then goes away" |
| 04:47.34 | picard276 | its going straight to from-context because im passing the callerID variable.. ? |
| 04:47.40 | picard276 | p3nguin ... could that effect it? |
| 04:47.44 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 04:47.50 | irishpilot | I'll give it a try, also I guess you meant capitol M as small m is for music |
| 04:47.52 | p3nguin | I wouldn't think so, but take it out and find out. |
| 04:48.02 | ChrisInSydney | So sho is on the VUC |
| 04:48.04 | ChrisInSydney | ?? |
| 04:48.13 | p3nguin | I'm still here. |
| 04:48.24 | ChrisInSydney | Is that you ?? |
| 04:48.27 | p3nguin | :) |
| 04:48.37 | irishpilot | you sent this too: TK]D-Fender: irishpilot, m() only launches your external watcher and the GOES AWAY. |
| 04:48.50 | ChrisInSydney | ahh. Tired if Al Kooper ? |
| 04:48.51 | irishpilot | and then [macro-payupbastard] exten => s,1,System(/somewhere/watchhimlikeahawk.pl ${ARG1} &) |
| 04:49.04 | irishpilot | will try that |
| 04:49.11 | p3nguin | Not necessarily tired of it, no. It sounds pretty bad, though. |
| 04:49.39 | p3nguin | They killed us. |
| 04:49.48 | ChrisInSydney | :-( |
| 04:50.00 | p3nguin | 12 hours |
| 04:50.03 | irishpilot | Now I have Dial(SIP/556,,M(macro-payupbastard)g); will try that |
| 04:50.15 | picard276 | p3nguin what do u think? |
| 04:50.23 | p3nguin | (2247.51) <p3nguin> I wouldn't think so, but take it out and find out. |
| 04:50.24 | picard276 | im putting the CID info in and then its autopassing too fast? |
| 04:50.28 | ChrisInSydney | Automatic do you think ?? |
| 04:50.39 | p3nguin | I'm not sure. |
| 04:50.59 | ChrisInSydney | we'll have to ask next time |
| 04:51.02 | p3nguin | Maybe someone in #vuc saw your question and checked the conf. |
| 04:51.25 | p3nguin | I asked before and they wouldn't give me a straight answer... just kept bitching about the fact that we stayed connected. |
| 04:52.01 | p3nguin | There's no reason they need to be recording it, anyway. |
| 04:52.07 | p3nguin | The "show" is over. |
| 04:52.07 | ChrisInSydney | true |
| 04:53.01 | p3nguin | I guess what I'm trying to say is, "If they don't fuckin' like it, they can kiss my big black ass." |
| 04:53.44 | p3nguin | Anyway... |
| 04:54.46 | p3nguin | Way to kill a chat. |
| 04:55.32 | *** join/#asterisk bmg505 (~leon@196-209-84-238.dynamic.isadsl.co.za) |
| 04:55.39 | [TK]D-Fender | Chapter 12 of "How to Not Win Friend and Not Influence People" :) |
| 04:56.18 | coppice | Friend in the singular isn't setting your sights very high |
| 04:57.10 | ChrisInSydney | Not that anyone was chatting |
| 04:57.23 | ChrisInSydney | You dont "Sound" black ;-) |
| 04:57.24 | irishpilot | strange does anybody know about macros in AEl? "No such context 'macro-payupbastard' for macro 'payupbastard'" |
| 04:57.40 | irishpilot | I didn't realize Macro's needed a context |
| 04:57.44 | irishpilot | or do they? |
| 04:57.46 | [TK]D-Fender | macro IS a context |
| 04:57.49 | p3nguin | I don't look black, either, for the record. |
| 04:58.01 | irishpilot | thats what I thought |
| 04:58.09 | irishpilot | strange it is telling me No such context 'macro-payupbastard' for macro 'payupbastard' |
| 04:58.23 | [TK]D-Fender | I don't see race. People tell me I'm white, and I only believe them because I can't dance. |
| 04:58.42 | irishpilot | must be something to do with the name. I call it macro payupbastard() { |
| 04:58.55 | [TK]D-Fender | macro-name |
| 04:59.05 | irishpilot | Oh ok I was calling it as SIP/556,,M(payupbastard)g |
| 04:59.10 | [TK]D-Fender | Don't go dreaming that random white-space is legel |
| 04:59.11 | irishpilot | will add macro- |
| 04:59.18 | [TK]D-Fender | M adds it |
| 04:59.35 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 04:59.38 | [TK]D-Fender | Perhaps AGI does have a special syntax actually.. |
| 04:59.48 | [TK]D-Fender | hard to say... it's all fake and I never recommend AEL |
| 05:00.25 | irishpilot | ya it didnt like it |
| 05:00.25 | irishpilot | No such context 'macro-macro-payupbastard' for macro 'macro-payupbastard' |
| 05:00.31 | irishpilot | lol |
| 05:00.55 | irishpilot | that was from Dial(SIP/556,,M(macro-payupbastard)g); |
| 05:02.25 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 05:02.27 | picard276 | yea i think something is wrong with the .call stuff? |
| 05:02.34 | irishpilot | fixed that |
| 05:02.50 | irishpilot | needs to start as: macro macro-mymacroname() { |
| 05:03.10 | irishpilot | now it says lacks 's' extension, priority 1 but I'll just add an s |
| 05:05.16 | p3nguin | picard276: Did you ever take out the extra settings and try again? |
| 05:05.25 | p3nguin | I've been waiting like 20 minutes. |
| 05:05.27 | picard276 | what settings? |
| 05:05.29 | picard276 | im sorry |
| 05:05.34 | irishpilot | no doesn't like s=> in the macro either |
| 05:05.49 | p3nguin | (2247.34) <picard276> its going straight to from-context because im passing the callerID variable.. ? |
| 05:05.52 | p3nguin | (2247.40) <picard276> p3nguin ... could that effect it? |
| 05:05.54 | p3nguin | (2247.51) <p3nguin> I wouldn't think so, but take it out and find out. |
| 05:06.11 | picard276 | well where will i send the call to if not from-context? |
| 05:06.58 | p3nguin | You asked me if the caller id value was breaking it. Take out the caller id value and test again. |
| 05:07.07 | irishpilot | I think I found a bug with AEL. app_macro.c:311 _macro_exec: Context 'macro-payupbastard' for macro 'payupbastard' lacks 's' extension, priority 1 |
| 05:07.14 | [TK]D-Fender | picard276: Where did you even come up with that name? |
| 05:07.30 | picard276 | what do u mean TK? |
| 05:07.30 | irishpilot | it freaks up if you put an s=> {} in there |
| 05:07.44 | [TK]D-Fender | irishpilot: that shows nothing IN s |
| 05:08.03 | [TK]D-Fender | picard276: "from-context" <------------- where did you even come up wirth this? |
| 05:08.14 | picard276 | its the main context |
| 05:08.15 | picard276 | its just a Dial |
| 05:08.22 | picard276 | from freepbx |
| 05:08.24 | [TK]D-Fender | Context is not a DIal |
| 05:08.33 | [TK]D-Fender | And since when is THAT name used by freepbx? |
| 05:08.34 | irishpilot | exactly |
| 05:09.21 | irishpilot | you can't have an S however as its a macro, if you put it in then it freaks out |
| 05:09.25 | picard276 | it works fine |
| 05:09.30 | picard276 | if i get rid of the CallerID |
| 05:09.56 | [TK]D-Fender | picard276: That is not a proper context to use in FreePBX. |
| 05:10.08 | [TK]D-Fender | CallerID may be it's own issue, but that's another matter |
| 05:10.16 | picard276 | its with any context though |
| 05:10.18 | picard276 | this works completely fine |
| 05:10.22 | picard276 | unless i pass the CallerID |
| 05:10.49 | p3nguin | It may be its own issue, too. |
| 05:12.24 | picard276 | yea it could be a .call asterisk bug |
| 05:12.33 | p3nguin | What's your version? |
| 05:12.49 | [TK]D-Fender | No, I seriously doubt it's a bug. |
| 05:12.51 | picard276 | 1.6 |
| 05:13.02 | [TK]D-Fender | that isn't a version or even a branch |
| 05:13.06 | picard276 | asterisk 1.6 |
| 05:13.11 | [TK]D-Fender | NOT A BRANCH. |
| 05:13.19 | [TK]D-Fender | 1.60 = branch., 1.6.1, 1,6.2 |
| 05:13.23 | p3nguin | Not even a stick. |
| 05:13.33 | p3nguin | Not even a twig. |
| 05:13.38 | picard276 | 1.6.2.20 |
| 05:13.42 | [TK]D-Fender | Nor a splinter |
| 05:14.02 | [TK]D-Fender | picard276: And that isn't the problem. First problem is you aren't looking |
| 05:14.15 | [TK]D-Fender | It's a congenital issue |
| 05:14.24 | picard276 | ? |
| 05:14.46 | p3nguin | Like herpes? |
| 05:14.52 | picard276 | hahaha |
| 05:15.09 | *** join/#asterisk bluregard (~matt@c-24-15-36-70.hsd1.il.comcast.net) |
| 05:15.16 | bluregard | good evening all |
| 05:15.43 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 05:18.07 | irishpilot | I think macro |
| 05:18.16 | irishpilot | I think Macro's are broken in AEL in 1.8 |
| 05:18.49 | irishpilot | can't call it at all, keep getting that Ive no S=> in my macro which makes no sense as its not a context its a macro |
| 05:19.17 | [TK]D-Fender | No, otherwise plenty of other people would have freaked by now. |
| 05:19.59 | p3nguin | A macro runs extension s in the macro-context. |
| 05:20.28 | irishpilot | if you put it in penguin it bails out |
| 05:20.35 | irishpilot | and the manual says not to use it |
| 05:20.49 | p3nguin | Consider using .conf instead of .ael. |
| 05:20.53 | irishpilot | check this |
| 05:20.53 | irishpilot | https://wiki.asterisk.org/wiki/display/AST/AEL+Macros |
| 05:21.00 | irishpilot | I have an AEL that is 3000 lines long |
| 05:21.02 | [TK]D-Fender | irishpilot: PM |
| 05:21.08 | irishpilot | to convert that to AEL would be a nightmare :( |
| 05:21.15 | irishpilot | from ael |
| 05:23.46 | ChrisInSydney | p3nguin: just loiaded 1.8. Looks like the attended transfer caller ID issue doesnt work in 1.8 either |
| 05:23.53 | ChrisInSydney | is there something I am missing ? |
| 05:24.19 | p3nguin | It seems to work for me, so I don't know. |
| 05:24.30 | ChrisInSydney | as a comparrison, it does work with Milkfish (OpenSER) on my dd-wrt router |
| 05:25.30 | ChrisInSydney | A calls B, B calls C, B talks to C, B presses Transfer, A and C are now talking. C sees Bs number still |
| 05:25.41 | ChrisInSydney | should see A |
| 05:26.17 | p3nguin | After the transfer completes, it should update if your channel technology supports that. |
| 05:26.31 | irishpilot | http://pastebin.com/RveW9VJF ok this is just a basic hack together |
| 05:26.33 | irishpilot | not the end product |
| 05:26.43 | [TK]D-Fender | irishpilot: You didn't pay attention earlier. |
| 05:26.51 | irishpilot | eh |
| 05:27.08 | irishpilot | its not finished Fender, I just want to see Watcher work |
| 05:27.11 | [TK]D-Fender | 23:48]irishpilotand then [macro-payupbastard] exten => s,1,System(/somewhere/watchhimlikeahawk.pl ${ARG1} &) |
| 05:27.24 | irishpilot | you didn't pay attention to me :p Im running Java |
| 05:27.27 | irishpilot | not a perl script |
| 05:27.27 | [TK]D-Fender | yours looks NOTHING like what I gave you as a sample |
| 05:27.39 | irishpilot | yours is conf not AEL |
| 05:27.39 | [TK]D-Fender | AGI is blovking and this is NOT supposed to be an AGI |
| 05:27.43 | ChrisInSydney | SIP / SNOM |
| 05:27.55 | [TK]D-Fender | System != AGI |
| 05:28.03 | [TK]D-Fender | Look what else is in there. |
| 05:28.05 | irishpilot | you said "language doesn't matter" |
| 05:28.05 | [TK]D-Fender | This matters |
| 05:28.13 | irishpilot | ;) |
| 05:28.14 | [TK]D-Fender | AGI isn't a language <- |
| 05:28.22 | irishpilot | Agi calls the language |
| 05:28.25 | [TK]D-Fender | NO |
| 05:28.29 | irishpilot | hehe |
| 05:28.41 | [TK]D-Fender | AGI is a dumb concept for hooking stdin/out for live interaction. |
| 05:28.47 | [TK]D-Fender | You aren't supposed to be doing this |
| 05:29.08 | [TK]D-Fender | that blocks the call from really bridging. |
| 05:29.12 | irishpilot | ok |
| 05:29.21 | irishpilot | so I am forced to use Perl |
| 05:29.22 | [TK]D-Fender | I said launch and external scripst. as in not tied to anything. |
| 05:29.26 | [TK]D-Fender | NO. |
| 05:29.28 | irishpilot | or Php |
| 05:29.28 | irishpilot | or something from the shell |
| 05:29.33 | irishpilot | and can't be java? |
| 05:29.37 | [TK]D-Fender | asdjhajkslhdalsd |
| 05:29.41 | irishpilot | or I guess could be a java command |
| 05:32.56 | *** join/#asterisk jeffgus (~jeffgus@2001:470:f2eb:1::4) |
| 05:34.42 | ChrisInSydney | p3nguin: What handsets are you using ?? |
| 05:35.09 | p3nguin | Cisco 7900 series |
| 05:35.19 | ChrisInSydney | I have 2xSnom w v7.3.30 / 8.4.32 + an Aastra + a Cisco SPA524G2 |
| 05:35.23 | ChrisInSydney | all the same |
| 05:36.00 | ChrisInSydney | Blind xfer works. Attended doesnt |
| 05:36.02 | p3nguin | I also use SCCP, not SIP. |
| 05:36.12 | ChrisInSydney | This is all SIP |
| 05:36.41 | ChrisInSydney | gotta go. Family are draging me off to get a life |
| 05:36.51 | ChrisInSydney | Thanks. Type soon |
| 05:51.08 | irishpilot | Ok I've good news Fender |
| 05:51.10 | irishpilot | <PROTECTED> |
| 05:51.22 | irishpilot | and it works |
| 05:51.50 | irishpilot | I used a GoSub U option on the Dial as Macro's don't work in AEL if you follow the manual so I'll lodge a bug on that later |
| 05:52.07 | irishpilot | the U option doesn't block as it comes back and calls as we wanted |
| 05:52.13 | irishpilot | call the script that is |
| 05:52.20 | [TK]D-Fender | it isn't the U that is not blocking |
| 05:52.26 | [TK]D-Fender | it was your use of AGI |
| 05:52.27 | irishpilot | I know |
| 05:52.29 | irishpilot | yep |
| 05:52.36 | irishpilot | so you're ok with the U option instead of M |
| 05:52.38 | irishpilot | ? |
| 05:52.45 | irishpilot | as M just won't work |
| 05:52.47 | irishpilot | with AEL |
| 05:53.18 | [TK]D-Fender | It was done wrong somehow |
| 05:53.39 | irishpilot | Ill work on that another time as once the U works then great |
| 05:53.47 | irishpilot | will see how the originate works now |
| 05:53.50 | irishpilot | brb |
| 05:54.28 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 06:21.12 | irishpilot | hey Fender I am close now, almost done! Just the Originate doesn't like my request with the "playhalf" and "spyhalf", here is what I mean: http://pastebin.com/DXHveYQf |
| 06:21.54 | irishpilot | To test it is working I tried putting in an extension so SIP/556 and that worked and it connected me to the playhalf message as expected. |
| 06:22.28 | irishpilot | I just need to get actual channel in there somewhere, which is where I am confused as this will be a 3rd channe; |
| 06:23.43 | irishpilot | Its the originateAction.setChannel("spyhalf"); that I must be wrong |
| 06:27.34 | [TK]D-Fender | irishpilot: I handed you the channel format |
| 06:29.57 | irishpilot | I don't understand then |
| 06:30.03 | irishpilot | because I did what I thought you said |
| 06:30.07 | p3nguin | SIP/556 <-- put in the channel parameter |
| 06:30.07 | irishpilot | left leg right leg |
| 06:30.13 | p3nguin | SIP/556 <-- not an extension |
| 06:30.25 | irishpilot | ok P but I can't call an extension |
| 06:30.32 | irishpilot | as the call is already in progress |
| 06:30.34 | p3nguin | You could if you wanted. |
| 06:30.52 | irishpilot | no I can't, let me explain |
| 06:30.57 | p3nguin | You can. |
| 06:31.01 | p3nguin | Well, we can... |
| 06:31.04 | p3nguin | Maybe you can't. |
| 06:31.23 | irishpilot | just before the Dial() takes place we goSub to the Java programme |
| 06:31.38 | irishpilot | then we immediately come back and bridge the channels in the dial |
| 06:31.53 | irishpilot | meanwhile the java programme is running in the back |
| 06:32.03 | [TK]D-Fender | [22:45][TK]D-Fenderchannel: Local/spy@spycontext/n |
| 06:32.10 | [TK]D-Fender | 3 hours ago |
| 06:32.25 | p3nguin | If there is an active channel, you can spy it. |
| 06:32.31 | [TK]D-Fender | I gave you the sample for the whole thing |
| 06:32.42 | p3nguin | And I gave you the originate command. |
| 06:32.46 | irishpilot | hang on |
| 06:32.49 | irishpilot | no you didn't' |
| 06:32.54 | p3nguin | Yeah, we did. |
| 06:33.06 | irishpilot | ok just a sec |
| 06:33.12 | irishpilot | <PROTECTED> |
| 06:33.18 | irishpilot | what goes in there |
| 06:33.20 | irishpilot | as I don't know |
| 06:33.33 | p3nguin | a channel's name |
| 06:33.36 | irishpilot | originateAction.setChannel(Fenderchannel: Local/spy@spycontext/n"); |
| 06:33.39 | irishpilot | ? |
| 06:34.11 | irishpilot | guys I know this seems totally easy to you but to me its completely new |
| 06:34.25 | irishpilot | I am doing my best and reading everything you are saying respectfully |
| 06:34.42 | irishpilot | I am also very grateful for your help |
| 06:35.12 | p3nguin | If you want to originate to a part of dial plan, use Local channels. Local/spy@spycontext/n is a valid Local channel name to originate to. |
| 06:35.12 | irishpilot | so please don't think I intentionally not getting you |
| 06:35.24 | [TK]D-Fender | originateAction.setChannel("Local/spyhalf@spycontext/n") |
| 06:35.35 | p3nguin | That's the "left" side. |
| 06:35.43 | [TK]D-Fender | Indeed |
| 06:35.50 | p3nguin | Then the right side can be either an extension or an application. |
| 06:35.53 | p3nguin | Your choice. |
| 06:35.56 | [TK]D-Fender | The rest looked about proper |
| 06:36.02 | irishpilot | ok |
| 06:36.03 | irishpilot | thanks |
| 06:36.10 | irishpilot | I appreciate that |
| 06:36.19 | irishpilot | the word Local |
| 06:36.38 | irishpilot | is that to be replaced with the channel that called the script |
| 06:36.41 | p3nguin | Local channels are used to turn a point in the dial plan into a device. |
| 06:36.42 | [TK]D-Fender | NO |
| 06:36.45 | irishpilot | ok |
| 06:36.46 | [TK]D-Fender | that is a literal word |
| 06:36.51 | [TK]D-Fender | letter for letter |
| 06:36.53 | irishpilot | ok so I use it as you put |
| 06:36.58 | irishpilot | 2 secs will run it |
| 06:36.59 | irishpilot | :) |
| 06:37.15 | [TK]D-Fender | You will have to use SetVar's in your originate to apss the channel name as a var that you can call up in your ChanSpy |
| 06:37.27 | [TK]D-Fender | No sense in running. You didn't give it the target |
| 06:37.41 | irishpilot | I thought I did with originateAction.setExten("playhalf"); |
| 06:37.54 | [TK]D-Fender | With Originate you can preset var values so you can use them immediately |
| 06:38.08 | [TK]D-Fender | no, we need the actualy CALL name to hook into |
| 06:38.08 | irishpilot | ok let me see how to do that brb |
| 06:38.10 | [TK]D-Fender | that is run-time |
| 06:38.40 | [TK]D-Fender | [TK]D-Fender23:48] irishpilot and then [macro-payupbastard] exten => s,1,System(/somewhere/watchhimlikeahawk.pl ${ARG1} &) |
| 06:38.48 | [TK]D-Fender | Which is why you see me pass it to the script |
| 06:38.58 | irishpilot | yep ARG1 is the channel name right |
| 06:39.06 | p3nguin | If you're just going to be playing a sound byte, why would you use Exten? App[lication] would be more appropriate. |
| 06:39.07 | irishpilot | I just need to change that to ${CHANNEL} |
| 06:39.16 | [TK]D-Fender | Now ARG1 shold have been from a macro where you actually passed it ${CHANNEL}} |
| 06:39.27 | [TK]D-Fender | So you could probably just do it direct from there |
| 06:39.47 | irishpilot | 2 secs p3 let me get this running first |
| 06:39.49 | [TK]D-Fender | Or not.... I think you needed the first leg, not the spawned Macro/gosub one. |
| 06:40.10 | irishpilot | the reason we used an extension is to use playback from there but I guess it could also be the app playback |
| 06:40.13 | [TK]D-Fender | Indeed this has to be from the callling side, so m() is the best way unless Gosub also supports args |
| 06:40.14 | p3nguin | There's a word I'm thinking of to describe this. |
| 06:40.32 | [TK]D-Fender | bed time here, later all... |
| 06:40.41 | irishpilot | yikes nooo :) |
| 06:40.44 | irishpilot | don't go to bed on me :) |
| 06:40.50 | irishpilot | Im almost there :) |
| 06:40.57 | [TK]D-Fender | That's what SHE saud |
| 06:40.59 | [TK]D-Fender | said* |
| 06:41.03 | irishpilot | haha |
| 06:41.10 | irishpilot | I thought of it after I said it! |
| 06:41.11 | irishpilot | haha |
| 06:48.31 | *** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net) |
| 06:50.55 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 06:51.23 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 06:53.42 | irishpilot | hey P3 I think I have it working, only I am not hearing the "warning" message played through Spy, I have the output of it, can show you if you like on pastebin? |
| 06:54.12 | irishpilot | seems to be playing to the wrong channel: -- Executing [playhalf@spycontext:2] Playback("Local/spyhalf@spycontext-644a;1", "somewarningmessage") in new stack |
| 06:55.00 | *** join/#asterisk mjordan (~mjordan@28.254-240-81.adsl-static.isp.belgacom.be) |
| 06:55.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 06:57.47 | irishpilot | Maybe ChanSpy needs another argument to break into the channel or something |
| 07:04.51 | irishpilot | that worked! needed whisper mode on |
| 07:04.58 | irishpilot | ChanSpy(${callerCh},wq); |
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| 09:59.53 | saxa | p3nguin: yes, i pastedbinned my sip.conf 2 days ago, and Fender confirmed it was ok. It is here: http://pastebin.com/ZTL5zjQZ |
| 10:00.34 | saxa | p3nguin: any corrections are welcome, but I used that sip.conf file when it was working. |
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| 12:03.22 | ChrisInSydney | Hi all. I have a loaded question: |
| 12:04.41 | ChrisInSydney | Can Ast 1.8.current manage the caller ID of attended transfers. For example A calls B, B places A on hold. B calls C, talks. B then presses Transfer to transfer A to C. C should now see A on their caller display |
| 12:04.47 | ChrisInSydney | Currently sees B |
| 12:05.21 | ChrisInSydney | The handsets are Snom. I also have some Aastra and Cisco SPA524G2s to try against |
| 12:11.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 12:22.33 | ChrisInSydney | sendrpid=pai |
| 12:22.41 | ChrisInSydney | fixes it |
| 12:26.21 | ChrisInSydney | Thats half the story. The display for A still says B even though they are talking to C |
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| 14:04.42 | kaldemar | ChrisInSydney: try rpid_update=yes |
| 14:05.19 | ChrisInSydney | in sip.conf under the peer defiition ? |
| 14:08.11 | ChrisInSydney | Got caller ID on call pickup working. Needed a firmware upgrade to 8.4.32 on the handsets. The 7.3.30 doesnt do it :-( |
| 14:09.13 | kaldemar | ChrisInSydney: under [general] |
| 14:10.28 | ChrisInSydney | <kaldemar> Ill give it a go |
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| 14:21.32 | ChrisInSydney | nup |
| 14:21.38 | ChrisInSydney | I'll do a sip trace |
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| 14:35.55 | ChrisInSydney | Firmware |
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| 15:13.03 | ChrisInSydney | <kaldemar> Looks like you need to set CONNECTEDLINE(number,i) and CONNECTEDLINE(name,i) for it to work |
| 15:13.28 | ChrisInSydney | You also need the latest SNOM firmware |
| 15:14.24 | ChrisInSydney | Moreover sendrpid=yes and rpid_update=yes have to be explicitly turned off otherwise I get a 604 error from the ITSP |
| 15:15.13 | ChrisInSydney | So, back to the drawing board as to getting this migration completed. More planning, dial plan rewriting and testing |
| 15:15.26 | ChrisInSydney | off to bed for me. its 2:15am |
| 15:15.38 | ChrisInSydney | night all. and thanks for the help :-) |
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| 16:13.38 | volga629 | AstriCon 2012 will be held October 23-25 in Atlanta, Georgia Is this real info ? How place reservation ? |
| 16:14.30 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:14.54 | ujjain | What is more respected here? Elastix or Trixbox? Or is this question evil? |
| 16:17.38 | pabelanger | ujjain: neither to be honest, each have their separate rooms |
| 16:17.56 | pabelanger | volga629: yes, I thought I seen info on the astricon.net site |
| 16:18.20 | ujjain | pabelanger, thanks :) |
| 16:19.08 | pabelanger | ujjain: actually, its not. Just, http://blogs.digium.com/2012/01/03/astricon-2012-save-the-date-and-make-a-resolution-to-attend/ |
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| 16:19.48 | ujjain | pabelanger, I can't visit. I live in another world. |
| 16:20.03 | pabelanger | ujjain: Mars? |
| 16:20.07 | pabelanger | way cool |
| 16:20.09 | ujjain | Holland. |
| 16:20.11 | volga629 | excellent I was unable make last year, but in 2012 I will thank you |
| 16:20.13 | pabelanger | :) |
| 16:20.14 | ujjain | I used Asterisknow before, but it's pretty horrible and insecure by default, too many standard passwords and requires much knowledge to change passwords. |
| 16:20.53 | ujjain | there are like 5 default passwords that need changing, it's insanely insecure, even if you edit the amportal.conf passes and mysql passwords, no good guides, documentation |
| 16:21.56 | pabelanger | never used it to be honest |
| 16:22.02 | pabelanger | but that seems bad |
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| 16:49.51 | [ProB]CrazyMan | hi, I connected for testing one asterisk box 1.4.26.3 Bristuffed 0.4.0 with one BRI Channel 13/15 with signaling bri_net to an other asterisk box (new install) 1.8.7.1 and jnet dahdi driver 1.0.13 on Channel 1/2 with signaling bri_cpe now I get on first box |
| 16:50.34 | [ProB]CrazyMan | now i get on the first box following error No D-Channel availible! using Primary channel 16 as D-Channel anyway! |
| 16:51.13 | [ProB]CrazyMan | on the newer box I get: Detected alarm on channel 1 and 2 : Red Alarm |
| 16:51.49 | [ProB]CrazyMan | do I have to enable somethin if i connect to Bri cards ? |
| 16:59.18 | p3nguin | saxa: Which end point is the one giving you trouble? |
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| 18:22.58 | WIMPy | [ProB]CrazyMan: Those are pretty ancient versions. You should use at least 1.8 for that kind of experiment. |
| 18:23.51 | WIMPy | Or one version. |
| 18:24.42 | [ProB]CrazyMan | I know but the olde one is the production server, and I want to migrate to 1.8, therefor I want to test if dialplan and every thing is working bevore I change to the new version |
| 18:24.57 | [ProB]CrazyMan | therefor I want to make test with one line |
| 18:25.47 | [ProB]CrazyMan | if I connect to bri cards, do I have to enable on one the terminator ? |
| 18:26.00 | WIMPy | Is the link jumpered or cabled for crossover? |
| 18:26.08 | WIMPy | You need termination on both ends. |
| 18:26.33 | [ProB]CrazyMan | ok its a normal ethernet cable between, do I need a crossed cable? |
| 18:26.53 | [ProB]CrazyMan | do the cards the crossing not by their own? |
| 18:27.39 | WIMPy | If you have jumpers for NT mode, that's for doing the crossover part. |
| 18:28.11 | [ProB]CrazyMan | ok so one side ist jumpered to NT mode |
| 18:28.41 | WIMPy | Then the ethernet cabel is right. |
| 18:29.04 | [ProB]CrazyMan | so what does this red alarm mean? |
| 18:29.17 | WIMPy | No link |
| 18:29.21 | [ProB]CrazyMan | the alarm comes every 5 seconds |
| 18:29.58 | [ProB]CrazyMan | currios it detect the alarm but in the same second it cleared the alarm |
| 18:30.15 | WIMPy | I've never used bristuff, so I don;t know if that supports powersave, which dahdi doesn;t seem to like very much. |
| 18:30.25 | WIMPy | Huh? |
| 18:31.04 | [ProB]CrazyMan | http://pastebin.com/gyM1G4Jt |
| 18:31.08 | WIMPy | So maybe it does. |
| 18:31.53 | [TK]D-Fender | D-chan on 16 is an E1 PRI stndard number for the 1st PRI port. |
| 18:31.58 | WIMPy | So can you make calls? Does it stop then? |
| 18:32.07 | [ProB]CrazyMan | if I try to do a call threw this channel it tells me that everybody is congested |
| 18:32.09 | [TK]D-Fender | Not something I would expect to see off anything BRI related. |
| 18:33.42 | [ProB]CrazyMan | I'm not shure if it has something to do with the older asterisk because there I alsways get the no D-Channels availible on that channel in also 5sec interval |
| 18:39.07 | kaldemar | [ProB]CrazyMan: older chan_dahdi used to do that for all configured spans that were not up. |
| 18:41.36 | saxa | p3nguin: i think is the nat where * is behind |
| 18:42.24 | saxa | p3nguin: at least the thing stopped working after i opened some ports on that nat. |
| 18:42.46 | p3nguin | (1059.18) <p3nguin> saxa: Which end point is the one giving you trouble? |
| 18:44.27 | saxa | thats it |
| 18:44.42 | saxa | the end point where * is behind imho |
| 18:45.04 | saxa | but to be honest, it does not work the client connected to the * server |
| 18:45.06 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 18:45.35 | p3nguin | Do you know what an end point is within the scope of this discussion? |
| 18:46.40 | p3nguin | It's going to be a phone or phone adapter. We're not talking about Asterisk; Asterisk is not an end point. |
| 18:53.45 | saxa | p3nguin: i have an grandstream gxp285 phone behind a nat and it connects to a server in my office behind another nat. |
| 18:54.06 | saxa | so that phone does not work anymore, does not work means i have no audio at all. |
| 18:54.27 | saxa | this is one of the 5 phones i have connected on that * box. |
| 18:54.34 | saxa | all others work ok |
| 18:54.37 | p3nguin | I'm going to ask only one more time and then I'm moving on to something else. |
| 18:55.09 | saxa | i do not understand you what you mean with an end point if its not the client. |
| 18:55.44 | p3nguin | Of the six devices that you have configured in your sip.conf, WHICH ONE is giving you trouble? |
| 18:59.03 | bbourdage | <PROTECTED> |
| 19:01.28 | saxa | p3nguin: maybe this explains better my setup http://pastebin.ca/2109741 |
| 19:01.44 | saxa | p3nguin: casasip |
| 19:02.01 | saxa | p3nguin: in the pastebin client == casasip |
| 19:02.46 | p3nguin | The probably could be that double NAT that you have your phone behind. |
| 19:03.00 | *** join/#asterisk P-NuT (~P-NuT@5ad48b0d.bb.sky.com) |
| 19:03.01 | p3nguin | You have two choices. |
| 19:03.09 | saxa | i thougt that |
| 19:03.30 | p3nguin | Put the modem in bridge mode to disable NAT on it... (my choice) |
| 19:03.30 | saxa | but it was working before in the same setup |
| 19:03.44 | p3nguin | Or don't use the NAT in the wireless router. |
| 19:03.49 | saxa | i have disabled everything possible in the modem |
| 19:04.21 | saxa | ok thx for the advise, I will try to see if there is way to put the modem in the bridge mode |
| 19:04.21 | p3nguin | You missed part of the statement: Put the modem in bridge mode. <--------- |
| 19:04.24 | p3nguin | bridge mode |
| 19:04.27 | p3nguin | not NAT mode |
| 19:04.33 | saxa | i got it yes |
| 19:04.52 | saxa | will try to connect firstly my phone directly to the modem |
| 19:04.57 | p3nguin | If you can bridge the modem, the router will be the first node on the customer premises. |
| 19:05.05 | saxa | and disconnect all other stuff. |
| 19:05.11 | P-NuT | Hi all, I know IAX2 supports RSA authentication, but does it support RTP encryption? |
| 19:05.11 | P-NuT | If not, what are my options for that? And, what is the strongest encryption I can get with asterisk? |
| 19:05.19 | saxa | ok let me try this. |
| 19:05.23 | saxa | brb |
| 19:05.25 | p3nguin | If you cannot bridge the modem, I can help you configure the router to not use NAT. |
| 19:05.27 | kaldemar | P-NuT: IAX2 doesn't use RTP. |
| 19:05.46 | saxa | thx p3nguin |
| 19:06.04 | saxa | let me try first to see if there is a way to put the modem in the bridge mode. |
| 19:06.11 | P-NuT | kaldemar: Ok sorry, my point is that I reeealy want to encrypt the audio with the strongest encryption. Suggestions? |
| 19:07.35 | kaldemar | P-NuT: IAX2 does support encryption though, see http://www.rfc-editor.org/rfc/rfc5456.txt for more information. other than that, you could also use SIP and SRTP. |
| 19:07.59 | P-NuT | kaldemar: what about zrtp? |
| 19:08.33 | kaldemar | P-NuT: zrtp is not supported by vanilla asterisk, yet. https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 19:09.09 | kaldemar | P-NuT: i recall running into a 3rd party patch/addon that enables zrtp. |
| 19:13.13 | P-NuT | kaldemar: If IAX does not support encryption, then what does the line encryption=aes128 in users.conf do? |
| 19:14.31 | kaldemar | P-NuT: i said it does support encryption. i told you it doesn't use RTP. |
| 19:16.00 | saxa | p3nguin: i did a test, conncetion the phone directly into the modem, it got one lan ip and it registered to the server, calling the voicemail i have not been able to hear anything, so same thing as with previous connection setup i have. |
| 19:17.24 | saxa | the phone got the 192.168.1.63 ip on the nat from thomson modem. |
| 19:17.46 | saxa | s/nat/lan side |
| 19:18.57 | p3nguin | Do you have any port forwarding enabled on the Thomson modem/router? |
| 19:20.29 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 19:23.03 | P-NuT | kaldemar: Oh sorry I am blind! |
| 19:24.34 | saxa | p3nguin: i have disabled everything. |
| 19:24.56 | p3nguin | How about bridge mode? Did you find that setting yet? |
| 19:25.22 | saxa | no, there is no way to put the modem in the bridge mode, at least i have not find it |
| 19:29.30 | bbourdage | Anyone have an idea on the voicemail directory question I posed previously ? |
| 19:33.53 | saxa | bbourdage: i have no idea, but i know that you can separate any field with the comma. |
| 19:34.23 | saxa | so in the field you can have as many spaces as you need |
| 19:34.35 | saxa | i'm not sure if this is what you want. |
| 19:35.35 | bbourdage | Thanks, I will try that. |
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| 19:37.04 | saxa | anybody knows if i need to enable TR069 ? |
| 19:37.17 | saxa | or should i disable that ? |
| 19:38.10 | jpsharp | On a DSL modem? No, don't disable that. |
| 19:38.22 | WIMPy | That is for auto-provisioning. |
| 19:38.34 | jpsharp | What he said. |
| 19:38.58 | WIMPy | So if you want to change things, better turn that off. |
| 19:41.32 | p3nguin | What he really needs to figure out is how to put it in bridge mode so he can get rid of double NAT. |
| 19:49.10 | WIMPy | Just remove the account settings? |
| 19:49.24 | WIMPy | Or is it bridged? |
| 19:51.07 | saxa | jpsharp: yes, on a dsl modem |
| 19:52.12 | saxa | one problem is that the modem has a very poor ui, so i have not seen any option to put it in a bridged mode. |
| 19:52.54 | saxa | i'm trying to find that , but there is only few options wia web, will try to acces the modem by telnet/ssh, if it has access. |
| 19:52.55 | Nugget | telnet is eeeeeeevil! |
| 19:53.04 | WIMPy | How is the modem side operating? bridged? pppoe? pppoatm? whatever? |
| 19:53.04 | saxa | lol |
| 19:53.11 | saxa | pppoe |
| 19:53.14 | jpsharp | If its anything like the Cisco DSL modem I have, its in a really obscure place. |
| 19:53.35 | WIMPy | Ok, just remove the pppoe settings and you should be able to use it as a bridge. |
| 19:53.35 | saxa | it is a Thomson TG508 |
| 19:54.02 | saxa | ok let me try |
| 19:54.12 | WIMPy | You might even be able to do both at the same time, but as you probably only have one account... |
| 19:54.43 | WIMPy | You plan to do the pppoe on a pc then? |
| 19:56.42 | jpsharp | According to the manual, the configuration should be in the settings for PVC0. It should have the selection of bridge, pppoe, and pppoa. |
| 19:57.24 | jpsharp | But yeah, you put it in bridge mode and you'll have to have something else that runs PPPoE. |
| 19:58.11 | jpsharp | Unless your provider is just doing DHCP instead of the sucktasticness that is PPPoE. |
| 20:01.10 | p3nguin | The WRT54G should be able to handle the PPPoE settings if they are needed after putting the modem in bridge mode. |
| 20:02.54 | saxa | WIMPy: i setup also a bridged conncetion |
| 20:03.08 | saxa | now i have pppoe on pvc0 |
| 20:03.14 | saxa | and bridge on pvc1 |
| 20:03.39 | saxa | i need to play a bit with those settings to see if i can reconfigure the whole thingy |
| 20:04.05 | p3nguin | I've never even heard of pvc0 and pvc1. |
| 20:04.36 | saxa | this is how is named here in the thomson |
| 20:05.01 | p3nguin | Right, but I don't know what those devices represent. |
| 20:05.06 | saxa | but yes, anyway i need to find now if i can do pppoe on the linksys |
| 20:05.17 | p3nguin | You can. |
| 20:06.25 | saxa | now there is a problem , i need to find out the password from my isp |
| 20:06.35 | saxa | to set up the wrt54g |
| 20:06.52 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 20:10.11 | p3nguin | That part may be more difficult, depending on how competent they are. |
| 20:11.01 | p3nguin | Tell them you need to configure your modem in bridge mode and that you need your credentials to put into your router connected to the modem. Maybe they'll be able to figure it out and tell you what you need. |
| 20:11.02 | ketas | can't dump that out? |
| 20:12.32 | ketas | i luckily got rid of that pppoe on my adsl |
| 20:12.38 | ketas | thank god |
| 20:14.24 | P-NuT | kaldemar: Does IAX2 encryption only encrypt the audio, or does it encrypt the setup info and everything? |
| 20:15.53 | p3nguin | p-nut: In IAX2, there is no separate media stream. |
| 20:16.12 | p3nguin | It's not like SIP and RTP. |
| 20:17.16 | kaldemar | P-NuT: it also encrypts the messages. |
| 20:32.01 | P-NuT | kaldemar: and aes128 is the highest encryption available? |
| 20:33.56 | saxa | p3nguin: i got the pass from my isp, they have it in automatic answers. |
| 20:34.42 | saxa | anyway i was unable to start the modem in bridge mode |
| 20:34.43 | p3nguin | Nice. |
| 20:34.59 | saxa | i mean, I have not been able to configure it to work |
| 20:35.00 | p3nguin | What do you mean unable to start it? |
| 20:35.06 | saxa | will need some time to play with it |
| 20:35.55 | saxa | but to me this seem not a big problem, because connecting the client directly into the modem, and having only one nat, it connected to asterisk server, but no audio at all. |
| 20:36.24 | saxa | also because before it was working with this same setup. |
| 20:36.39 | saxa | so if it worked before why it should not work now anymore. |
| 20:36.46 | saxa | ? |
| 20:37.11 | saxa | i understand that having 2 nats in the middle can be of a problem |
| 20:37.18 | p3nguin | Let me explain one thing. |
| 20:37.27 | saxa | go ahead |
| 20:37.39 | p3nguin | The NAT of the modem/router is not the same as the NAT in the Linksys router. |
| 20:38.05 | p3nguin | Until you eliminate the NAT in the modem/router, we don't know if that is causing the problem. |
| 20:38.31 | p3nguin | For example, a Belkin wireless router connected to a cable modem in bridge mode does not work with SIP phones. |
| 20:38.57 | p3nguin | But a Linksys wireless router connected to the same cable modem in bridge mode might work with SIP phones. |
| 20:39.14 | p3nguin | Not all companies make their devices equal. |
| 20:39.30 | saxa | ok I got it, but this one was working. |
| 20:39.42 | p3nguin | You keep saying that, but it isn't working now. |
| 20:39.56 | saxa | actually the first time you have helped me to configure up the whole thing. |
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| 20:40.23 | saxa | p3nguin: yes, unfortunately I was not messing anything on my client side when it stoped to work |
| 20:40.30 | saxa | but yes, i agree with you. |
| 20:44.17 | saxa | i'm trying to find out why it does not connect in bridge mode |
| 20:44.55 | saxa | i have put the password, but i have no way to configure the static ip in the pppoe in my linksys wrt54g |
| 20:45.20 | saxa | in the thomson i have the static ip in my pppoe config. |
| 20:45.31 | saxa | any idea how can i do that ? |
| 20:46.38 | saxa | ok probably i can set this up in advanced routing |
| 20:46.47 | WIMPy | I wouldn;t expect that to be neccessary. |
| 20:52.16 | saxa | i have an option to leave nat enabled in bridged mode |
| 20:52.24 | saxa | do i leave it on ? probably not |
| 20:53.30 | WIMPy | no |
| 20:57.45 | saxa | ok I should be in bridged mode now |
| 20:57.53 | saxa | let me check few things |
| 20:57.59 | saxa | and i can confirm it |
| 20:58.12 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 21:01.37 | saxa | ok yes, I'm in bridged mode right now |
| 21:01.45 | saxa | no nat, no dhcp on the modem |
| 21:01.58 | saxa | so only nat i have is the one from wrt54g |
| 21:02.26 | saxa | but yes, no audio at all on my phone |
| 21:03.08 | saxa | http://pastebin.ca/2109762 |
| 21:03.26 | saxa | good, bridged mode seems faster. |
| 21:03.44 | saxa | it is faster, since there is one nat less in the path |
| 21:04.01 | saxa | the above is what i get from my server |
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| 21:18.51 | saxa | hmm, still no audio |
| 21:19.18 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
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| 21:20.55 | saxa | Name/username Host Dyn Forcerport ACL Port Status |
| 21:20.59 | saxa | casasip/casasip 189.26.255.43 D N 7741 UNREACHABLE |
| 21:21.07 | saxa | heh, but i canplace a call |
| 21:21.33 | saxa | of course with no audio, but i see it starts the voicemail on * , i can see that on CLI |
| 21:22.42 | saxa | ok I got it now |
| 21:22.45 | saxa | yeayyyy |
| 21:23.05 | saxa | i just set up to use the external ip on my phone |
| 21:23.31 | saxa | great |
| 21:23.33 | saxa | finally |
| 21:23.40 | saxa | thanks to all |
| 21:30.16 | saxa | ok, now is still something wrong as i can not place calls outside over DAHDI |
| 21:30.53 | saxa | if I call my mobile phone, I see that * places a call, but i do not hear anything back. |
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| 21:42.25 | p3nguin | saxa: You enabled nat traversal on the phone? |
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| 22:50.48 | titou | hi all |
| 22:51.27 | titou | I would like to give SIP calls from a low badnwidth |
| 22:51.39 | titou | for this reason I guess the LPC10 codec would help me |
| 22:52.20 | titou | unfortunately my SIP provider doesn't accept LPC10 codec but only alaw and ulaw |
| 22:53.13 | titou | thus I would like to know if asterisk is the right software to create a gateway between my SIP client which would use the LPC10 codec and my SIP provider which uses alaw/ulaw |
| 22:53.32 | titou | (ie. install asterisk on my server located on a high wandwidth internet connection) |
| 22:53.36 | titou | ? |
| 22:53.47 | WIMPy | It can do so, yes. |
| 22:53.58 | titou | is it easy to implement? |
| 22:54.35 | WIMPy | Depends on what you want. |
| 22:54.52 | WIMPy | If you're easily pleasy, yes. Otherwise, no. |
| 22:56.36 | titou | pleasy? |
| 22:56.50 | WIMPy | oops |
| 22:56.52 | WIMPy | pleased |
| 22:57.24 | titou | ok I will see |
| 22:57.52 | titou | and are we still able to hear something with the LPC10 codec ? :) |
| 22:57.58 | titou | (I mean the quality is not definitly so bad?) |
| 22:58.40 | WIMPy | "something" |
| 22:59.04 | WIMPy | You may want to try different codecs with different settings. |
| 22:59.13 | WIMPy | There's plenty to choose from. |
| 22:59.51 | titou | y connection is around 4ko/s I don't think I have a lot of choice in the codec I can use :( |
| 22:59.56 | titou | my* |
| 23:00.23 | WIMPy | No go |
| 23:01.01 | WIMPy | The RTP headers will hardly fit, let alone any voice. |
| 23:01.45 | titou | you mean it's unthinkable to give a SIP call with that kind of connection? |
| 23:02.11 | WIMPy | yes |
| 23:02.18 | titou | :'( |
| 23:03.21 | WIMPy | Err, wait. You mean octetts? |
| 23:03.31 | titou | yup |
| 23:03.40 | WIMPy | Ah, ok. |
| 23:03.52 | titou | sorry ^^ |
| 23:04.04 | WIMPy | Add some delay and that should be ok. |
| 23:04.38 | titou | my connection comes from a 3G usb key |
| 23:04.59 | WIMPy | Without 3g? |
| 23:05.00 | titou | but here I only have EDGE connection |
| 23:05.21 | WIMPy | Not a good situation. |
| 23:05.29 | titou | that's why the bandwidth is so small :s |
| 23:05.57 | WIMPy | That probably means lots of jitter as well. |
| 23:05.58 | titou | ok then I'll try and I will see :) |
| 23:06.10 | titou | thank you for all these information WIMPy |
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| 23:08.51 | fornax | Hi, I'm running dahdi-2.5.0.2-r2 on a gentoo machine with the recent 3.2.1 kernel and get the following BUG message: http://pastebin.com/2T4fq79W. Can someone give me a hint what went wrong and where to file a bug? |
| 23:09.40 | WIMPy | Using punctuation after an URL is evil. |
| 23:10.23 | WIMPy | vzaphfc? What version is that? |
| 23:10.33 | WIMPy | You should try the dahdi-hfcs. |
| 23:13.09 | fornax | how can I get the vernon of vzaphfc? |
| 23:13.31 | fornax | it is the last stable built published for gentoo |
| 23:14.36 | WIMPy | Actually I'm asking about vzaphfc itself. It's not a version I know about. |
| 23:14.59 | WIMPy | The old one I know is just zaphfc and that has been replaced by dahdi-hfcs. |
| 23:15.58 | fornax | hm, okay, interesting, I'm running the gentoo builds for many years now and the kernel update and the switch to the new package resulted in this problem. I have a cheap hfc-s card. |
| 23:17.13 | WIMPy | Did you make sure the kernel drivers aren't loaded? |
| 23:18.11 | WIMPy | Oh, that was in the pb. Don't seem to be loaded. ok. |
| 23:18.12 | fornax | are there new kernel drivers that overlay the dahdi drivers? |
| 23:18.28 | WIMPy | Not new. |
| 23:18.42 | WIMPy | You have several drivers to choose from. |
| 23:18.52 | fornax | hm, okay, so I made an oldconfig and did no enable something new that related to dahdi |
| 23:19.25 | WIMPy | dahdi is not in the kernel. |
| 23:19.43 | fornax | I know that the original dahdi drivers need to be patched to be used with hfc-s cards. Maybe something went wring |
| 23:19.54 | WIMPy | But mISDN is. But at least that wasn;t loaded according to your PB. |
| 23:20.28 | WIMPy | The zaphfc (and related modules) aren't part of dahdi. |
| 23:20.37 | WIMPy | They are 3rd party add ons. |
| 23:22.29 | fornax | okay, so maybe they made something wrong when they patched the recent dahdi sources and I'm one of the rare people that have such a plain hfc-s :-( Now, I seem to have to downgrade the kernel or have to look for a card that works out of the box with asterisk |
| 23:22.34 | fornax | but they are so expansive |
| 23:23.18 | WIMPy | That should be the most used card there is. |
| 23:23.53 | WIMPy | I prefer to use the kernel drivers, but if you want to use dahdi, use dahdi-hfcs. No the older stuff. |
| 23:24.00 | fornax | okay, but why is its such a problem to apply correct patches for it or make dahdi support it out of the box? |
| 23:24.28 | fornax | how to do it with the kernel drivers? I never thought that this is possible? Using a bridge over misdn? |
| 23:24.38 | WIMPy | dahdi is a Digium thing. That may already be the answer. |
| 23:24.52 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 23:25.09 | WIMPy | There you've got a list of the possibilities. |
| 23:25.12 | fornax | okay, sure, this is the answer. But I always thought that dahdi is the best and clean solution |
| 23:25.59 | WIMPy | There is no good solution and which is the best, depends on your needs. |
| 23:26.22 | fornax | okay, I see dahdi-hfcs is unrelated to dahdi, interesting, so I could probably uninstall dahdi and try dahdi-hfcs? |
| 23:26.27 | fornax | ok |
| 23:26.47 | WIMPy | No, you need dahdi as well. |
| 23:27.16 | WIMPy | It's a framework with sub-drivers, just like many other things. |
| 23:28.18 | fornax | okay, so i worked very much with dahdi and the integratio of zaphfc and just want to find a solution that is stable and survives upgrades |
| 23:29.15 | WIMPy | The hfc support had to be updated regularly to work with current dahdi versions. |
| 23:29.58 | fornax | ok |
| 23:30.06 | fornax | so maybe misdn is an option |
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