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| 04:12.51 | its_jeremy_ | hello. anyone here? i have a quick question |
| 04:14.08 | its_jeremy_ | i want to have an exention be able to use the PITCH_SWITCH function on an outgoing call. I am not able to figure this out |
| 04:43.06 | ChannelZ | it's PITCH_SHIFT |
| 04:43.14 | ChannelZ | core show function PITCH_SHIFT |
| 04:43.45 | ChannelZ | Set it before you Dial and see what happens |
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| 04:59.26 | overyander | ChannelZ, will it make a man sound like a little girl? LOL |
| 05:00.36 | ChannelZ | Like a cartoon alien girl, yes |
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| 05:05.18 | overyander | awesome, i'll have to enable that for all inbound/outbound calls tomorrow. :) |
| 05:05.31 | overyander | that actually sounds like a great april fools pranks! |
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| 07:26.18 | Zogot | ahoyhoy |
| 07:33.55 | Zogot | when including contexts, it doesn't work if all the lines in that context are priority 'n'? |
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| 07:44.24 | ChannelZ | You have to have a 1 or something first in order to "next" |
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| 08:14.12 | Zogot | so i have this currently in the dialplan. as with the 'o' flag in dial, the incoming caller id is set to the CALLERID variable, but where i have Xfer: (line 14) i want the original recieving CallerID's number. https://gist.github.com/zogot/78064ef4f0fa2e03a78f i supplied an example in the comment |
| 08:15.52 | Zogot | is that even doable? |
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| 08:23.47 | JerJer | Zogot: you will want to copy CALLERID(num) too |
| 08:24.16 | Zogot | aye but that is 100(John), as the o flag sets it to that no? |
| 08:24.34 | JerJer | oh |
| 08:24.43 | JerJer | i haven't used o flag |
| 08:25.07 | JerJer | i'm old school, i guess |
| 08:25.29 | Zogot | im still a major rookie at this, just started with this stuff. i have another team member who is working on this but hes been pretty busy so far to take a look |
| 08:25.50 | Zogot | if i can get the original receivers phone, i can use agi to get the number |
| 08:25.52 | Zogot | i think lol |
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| 09:14.59 | jamesc|2 | what can cause this? app_directed_pickup.c:73 pickup_do: Unable to answer ,The context is I am trying to pickup a ringing phone |
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| 10:27.35 | Milos | Hi guys, I'm trying to retain original caller ID when transferring unconditionally using Dial(). |
| 10:27.42 | Milos | Apparently Dial(num,,o) is meant to do this, but it doesn't. |
| 10:27.44 | Milos | Tips? |
| 10:28.43 | eirirs | Redial. |
| 10:28.52 | Milos | Redial what sorry? |
| 10:28.53 | eirirs | </troll> |
| 10:28.57 | Milos | Okay. |
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| 11:22.06 | Zogot | Milos: it does, been working with it today |
| 11:22.09 | Zogot | its working for me |
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| 11:22.22 | Zogot | just setup that will display Xfer: on attended transfer |
| 11:22.32 | Zogot | Xfer: Original Caller ID Name |
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| 12:42.10 | Stefan27 | i have an active call SIP/A with SIP/B in a simple_bridge. SIP/A sends DTMF-sequence **1 which is mapped to a macro in dialplan which after a while executes: Executing [s@macro-monitor:14] MixMonitor("SIP/A", "auto___1408624701___55___53-in.wav,,/var/lib/asterisk/scripts/post-recording.sh auto___1408624701___55___53") in new stack. But this application seems to cause a hangup (SIP/A leaves |
| 12:42.11 | Stefan27 | the bridge) |
| 12:43.08 | Stefan27 | I googled 'MixMonitor cause hangup' but without any great answers am I missing a parameter to MixMonitor or something? |
| 12:43.20 | [TK]D-Fender | Show us |
| 12:43.46 | Stefan27 | The entire log with debug? it's full of other crap :) |
| 12:44.07 | [TK]D-Fender | Stefan27: no body = no autopsy |
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| 12:45.08 | Stefan27 | no body? no autospy? |
| 12:45.46 | [TK]D-Fender | Stefan27: Very clear and related analogy. |
| 12:46.03 | [TK]D-Fender | Stefan27: You want us to look at why the call DIED... give us teh BODY to examine |
| 12:46.28 | Stefan27 | sure, will do! |
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| 12:52.45 | Stefan27 | http://pastebin.com/0sKv8C8W line 1656 |
| 12:53.49 | file | this is Asterisk 12 or 13 |
| 12:53.54 | Stefan27 | 12.3.2 |
| 12:54.01 | Stefan27 | one version behind i guess |
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| 12:55.07 | file | https://issues.asterisk.org/jira/browse/ASTERISK-24027 |
| 12:55.21 | Stefan27 | aha autopsy is the word for examining dead bodies |
| 12:56.49 | Stefan27 | thanks file... didn't find that JIRA! |
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| 13:19.15 | Zogot | for the Dial command, option 'b', how you you write that? |
| 13:20.10 | Milos | Trying to transfer an incoming call and I get chan_sip.c:23941 handle_response_refer: SIP transfer to <sip:num@provider> declined, call miserably fails. |
| 13:20.15 | Milos | What would cause this? |
| 13:20.56 | Katty | lack of breakfast. |
| 13:20.59 | Katty | did you eat breakfast? |
| 13:21.07 | Stefan27 | Zogot, just Dial(SIP/blabla,20,b) ? |
| 13:21.23 | Milos | Katty, I did not :( |
| 13:21.33 | Katty | i recommend pancakes. |
| 13:21.39 | Milos | DELICIOUS |
| 13:21.41 | Milos | will you cook? |
| 13:21.51 | Zogot | Stefan27: sorry i mean all the options for b, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial |
| 13:22.06 | Zogot | Stefan27: b(context,exten,priority(arg1,argN))? |
| 13:22.07 | Katty | will i? hmm. possibly |
| 13:22.14 | Katty | i plan on cooking dinner tonight. |
| 13:22.34 | Milos | you sound pretty awesome |
| 13:22.44 | Milos | cooking dinner is srs bsns |
| 13:22.58 | Stefan27 | oh, sorry, b means something else in my asterisk |
| 13:22.58 | Katty | Milos: (= |
| 13:23.24 | Milos | / |
| 13:23.26 | Milos | | = |
| 13:23.28 | Milos | \ |
| 13:24.03 | [TK]D-Fender | Milos: "core show application dial" <- |
| 13:24.09 | [TK]D-Fender | Milos: It's there |
| 13:24.32 | Katty | [TK]D-Fender: core show application breakfast |
| 13:24.43 | Katty | [TK]D-Fender: did you ever get back to the gym? |
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| 13:25.18 | [TK]D-Fender | Katty: That assumes I was "fast"-ing before. Time to transition to a constant flow of bacon.... |
| 13:25.58 | Katty | invests in fish oil stock |
| 13:26.14 | Katty | [TK]D-Fender: would you like a side of cholesterol with your bacon? |
| 13:26.37 | Milos | MAYBE |
| 13:27.02 | [TK]D-Fender | Katty: Stress is the real cause of cholerstol vs dietary intake. Bacon is calming, therefor good. |
| 13:27.13 | Katty | pats [TK]D-Fender |
| 13:27.14 | [TK]D-Fender | Milos: b([[context^]exten^]priority[(arg1[^...][^argN])]): Before initiating an outgoing call, Gosub to the specified location using the newly created channel. The Gosub will be executed for each destination channel. |
| 13:27.16 | Katty | of course dear. of course. |
| 13:27.16 | file | did I hear bacon? |
| 13:27.27 | Katty | you did! |
| 13:27.40 | Milos | no |
| 13:27.42 | Milos | he read bacon |
| 13:27.45 | Milos | he didn't hear it |
| 13:27.48 | Milos | unless it was sizzling |
| 13:27.50 | Katty | or did he? |
| 13:27.53 | Katty | maybe file is deaf |
| 13:27.58 | Katty | and his irc client reads to him |
| 13:28.00 | Milos | [TK]D-Fender, I'm trying to find out where that explains anything to do with tranfers being denied |
| 13:28.06 | file | hi |
| 13:28.13 | Katty | hi file |
| 13:28.19 | Katty | how's zoe |
| 13:28.43 | Milos | Katty, if he were death, he would not be hearing his IRC client read to him |
| 13:28.47 | Milos | er, deaf |
| 13:29.04 | Katty | and by deaf i meant blind hehe |
| 13:29.11 | Milos | heeeeeeeeeheeeeeeeeee |
| 13:29.28 | file | Katty, she is good! she just took off to sleep in her cave in my office |
| 13:29.32 | [TK]D-Fender | Milos: Bad aim, sorry.. |
| 13:29.38 | [TK]D-Fender | Zogot: those were for you |
| 13:29.56 | [TK]D-Fender | Zogot: "core show application dial" <- |
| 13:29.56 | Milos | wow |
| 13:30.00 | Milos | that was severely bad aim |
| 13:30.08 | [TK]D-Fender | Zogot: b([[context^]exten^]priority[(arg1[^...][^argN])]): Before initiating an outgoing call, Gosub to the specified location using the newly created channel. The Gosub will be executed for each destination channel. |
| 13:30.24 | [TK]D-Fender | Milos: Not in terms of when your questions were asked. |
| 13:30.26 | Milos | [TK]D-Fender, that being said, core show application dial says that using the 'o' option will retain original caller id. if this does not happen, my provider must be overriding - yes? |
| 13:30.28 | Zogot | [TK]D-Fender: thanks man, i figured it was for me :) |
| 13:31.05 | Milos | Katty, last night, I had something for dinner that rhymes with your handle |
| 13:31.15 | [TK]D-Fender | Milos: I'm not sure what's thransfering what to where... |
| 13:31.35 | Milos | [TK]D-Fender, incoming call to asterisk that I want to divert to a mobile phone |
| 13:31.57 | [TK]D-Fender | Milos: how are you doing this? |
| 13:32.05 | [TK]D-Fender | Milos: Transfer()? |
| 13:32.09 | Milos | [TK]D-Fender, using Dial() it shows asterisk's outgoing number again, using Dial(,,o) shows the same, Transfer() fails entirely with above message. |
| 13:32.13 | Katty | Milos: oh? i hope this is safe for work |
| 13:32.22 | [TK]D-Fender | Milos: Some providers may just tell you to GTFO on those... |
| 13:32.24 | Milos | Katty, it is. I had patties. |
| 13:32.33 | [TK]D-Fender | Milos: And force you to keep the call passing through your system. |
| 13:32.35 | Katty | Milos: beef? |
| 13:32.41 | Milos | [TK]D-Fender, why the f would they do that those bastards |
| 13:32.43 | [TK]D-Fender | Milos: Which I suppose is what you want to alleviate. |
| 13:32.43 | Milos | Katty, beef indeed |
| 13:32.52 | Katty | Milos: and was there.... cheese involved? |
| 13:32.57 | Milos | there was |
| 13:33.01 | Milos | r u spying on me m8 |
| 13:33.11 | [TK]D-Fender | Milos: probably ebcause their SBC can't optimise it all the way back |
| 13:33.25 | Milos | [TK]D-Fender, that sounds terrible |
| 13:33.41 | Milos | how am I meant to forward calls and retain caller ID -__ |
| 13:33.49 | [TK]D-Fender | Milos: You're probably stuck in "deal with it" ter3ritory... |
| 13:34.13 | Milos | I'ma call them up and give tell them how I feel. |
| 13:34.17 | Milos | s/give// |
| 13:34.17 | [TK]D-Fender | Milos: if you are bridging the call out this has NO impact on your ability to set the callerID. |
| 13:34.38 | Katty | Milos: cheese is divine ^_^ |
| 13:34.43 | Katty | [TK]D-Fender: tell me something sweet in french. |
| 13:34.45 | [TK]D-Fender | Milos: If they let you set it... then you can set it. If they let you and you misconfigured something.. that's on you. |
| 13:34.45 | Milos | I didn't understand what you meant by that - bridging? |
| 13:34.50 | Milos | Katty, mmmmmmmmmm. yum. |
| 13:34.52 | Katty | [TK]D-Fender: and don't pull a literal something sweet |
| 13:35.01 | [TK]D-Fender | Milos: Have we loked at the actual attempt yet? |
| 13:35.17 | Milos | [TK]D-Fender, how do you mean? By attempt you mean... logs? |
| 13:35.28 | [TK]D-Fender | Milos: Yes, as in "looking". |
| 13:35.41 | Milos | Are we talking asterisk logs or SIP protocol logs? |
| 13:35.57 | [TK]D-Fender | Milos: the "yes my side actually does look sane" check. |
| 13:36.10 | [TK]D-Fender | Milos: Looking at LESS would not be doing the job. |
| 13:36.36 | Milos | I mean this with utmost respect but I have no idea what you mean and how I'm meant to be concluding my side is sane. |
| 13:36.44 | Milos | Can you give me a more direct pointer at what I should do exactly? |
| 13:37.27 | [TK]D-Fender | Milos: Prove what your INVITE out them looks like to see if includes the proper callerID's, etc. |
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| 13:37.50 | Milos | Ok, I'll make it super verbose and find those statements. |
| 13:39.19 | Stefan27 | I seem to be able to work around the mixmonitor problem by replacing the mixmonitor() app call with a System(asterisk -rx "mixmonitor start $chan $args") ? |
| 13:39.38 | Stefan27 | is there a better way perhaps? |
| 13:40.20 | [TK]D-Fender | Stefan27: It has been fixed.. you could UPGRADE to the point where it was fixed.... |
| 13:40.56 | [TK]D-Fender | Stefan27: Asterisk 12.5.0 (2014/08/19) |
| 13:41.07 | Stefan27 | I could but then i have to re-apply my own patches to chan_sip |
| 13:41.30 | [TK]D-Fender | Stefan27: https://issues.asterisk.org/jira/browse/ASTERISK-24027 <---- Resolved: |
| 13:41.32 | [TK]D-Fender | <PROTECTED> |
| 13:41.43 | Stefan27 | probably will upgrade at some point anyways though |
| 13:41.50 | [TK]D-Fender | Stefan27: What patches? |
| 13:43.31 | Milos | [TK]D-Fender, I can see REFER with Refer-To and Referred-By and they are correct? |
| 13:44.17 | [TK]D-Fender | Milos: Since you shouldn't be using Transfer() any more at all based on the refusal we saw.... you should be DIAL-ing this like normal and it's be an INVITE |
| 13:44.21 | Stefan27 | just some small custom changes, like setting extra dialplan variables |
| 13:45.24 | [TK]D-Fender | Stefan27: Seems like you'll be having to do that for a while... |
| 13:45.45 | Stefan27 | yeah, but they are not significant |
| 13:46.01 | Stefan27 | goal is to work with clean installations just testing |
| 13:46.46 | Milos | [TK]D-Fender, From: "original-caller" <sip:asterisk-outgoing@ip-addr>;tag=as2e8e2fb2 |
| 13:46.50 | Milos | [TK]D-Fender, this looks correct |
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| 13:47.03 | Milos | [TK]D-Fender, it is setting the original caller ID it seems |
| 13:47.53 | Milos | also has "original-caller" in Remote-Party-ID |
| 13:48.55 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-kxbwguifpsjubgvx) |
| 13:48.57 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:49.02 | [TK]D-Fender | Milos: Then it seems yor provider not be letting you set it at all. |
| 13:49.52 | Milos | [TK]D-Fender, I see. And, in terms of the 603 Decline, that is most likely them not wishing to handle Transfer() or their infrastructure doesn't support it - what's your best guess? |
| 13:50.18 | [TK]D-Fender | Milos: I don't really recall seeing any that do... |
| 13:50.30 | [TK]D-Fender | Milos: only failures. |
| 13:51.10 | Milos | wtf |
| 13:51.12 | Milos | why?????? |
| 13:51.18 | *** join/#asterisk aross42 (~aross@CPEbcc8100f769b-CMbcc8100f7698.cpe.net.cable.rogers.com) |
| 13:51.30 | *** join/#asterisk [Outcast] (~jjones@64.206.121.41) |
| 13:55.45 | Zogot | b([[context^]exten^]priority[(arg1[^...][^argN])]): are the brackets on this wrong or is priority without arguments required? |
| 13:55.47 | [TK]D-Fender | Milos: I've already answered that. |
| 13:56.10 | Milos | <[TK]D-Fender> Milos: probably ebcause their SBC can't optimise it all the way back |
| 13:56.11 | Milos | that? |
| 13:56.26 | [TK]D-Fender | Milos: Yes. |
| 13:56.35 | Milos | alright |
| 13:56.58 | [TK]D-Fender | Zogot: Arguments are OPTIONAL which is why they are in braces |
| 13:57.09 | [TK]D-Fender | Zogot: Priority is REQUIRED at a minimum |
| 13:57.16 | Zogot | ok |
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| 14:10.07 | Milos | [TK]D-Fender, heh looks like I was doing it wrong |
| 14:10.30 | Milos | [TK]D-Fender, Transfer() probably works but I was doing it after Answer() and then it's not a transfer anymore, hence 603 Decline |
| 14:10.47 | Milos | [TK]D-Fender, just getting loads of retransmission times out after sending 302 Redirect |
| 14:11.00 | Milos | s/times out/timesouts/ |
| 14:11.10 | Milos | timeouts... not timesouts |
| 14:11.22 | [TK]D-Fender | Transfer is not stopped by being answered. |
| 14:11.55 | Milos | no but otherwise it doesn't send sip302... |
| 14:11.58 | [TK]D-Fender | perhaps there is something else in there |
| 14:12.10 | Milos | if I answer it it's not a redirect |
| 14:12.41 | Milos | http://the-asterisk-book.com/1.6/applikationen-transfer.html |
| 14:12.49 | Milos | <PROTECTED> |
| 14:14.21 | Milos | any idea why I would be getting loads of these suddenly? http://bpaste.net/show/8d3581b48e43 |
| 14:14.45 | Milos | like an entire waterfall of them |
| 14:14.49 | *** join/#asterisk BitEvil (~quassel@tor/regular/SpeedEvil) |
| 14:15.55 | [TK]D-Fender | Milos: NO, especially because we aren't even loking at the packets |
| 14:16.12 | Milos | it sent like a million 302 Redirects |
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| 14:17.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:19.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:22.08 | Milos | oh |
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| 14:30.43 | Milos | [TK]D-Fender, it was my extensions.conf going in a loop. anyway, it ACKs my 302 Redirect but then seemigly drops the call. |
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| 15:31.43 | hexanol | I'm wondering |
| 15:31.58 | hexanol | I've sent a mail at security@asterisk.org on monday |
| 15:32.09 | hexanol | and I haven't received a response yet |
| 15:32.13 | hexanol | you know how much time |
| 15:32.16 | hexanol | it usually take |
| 15:32.19 | alami | what is the best way to the run cli command like confbridge list from web interface? exec or AGI? |
| 15:32.34 | hexanol | to get an acknowledgement for such a mail ? |
| 15:33.11 | rhamnett | Can someone please give me some clarity on whether any version of Asterisk can use DNS SRV to invite to the next record in the DNS SRV entry if the first server fails? I have read the source code comments which seem to state there is no failover, but I am wondering whether this is referring to failover mid call. Many thanks for any advice |
| 15:33.16 | [TK]D-Fender | alami: How does AGI offer this? |
| 15:33.37 | WIMPy | alami: You should check your options a little more. AGI is not one of them. |
| 15:33.44 | file | rhamnett, chan_sip can't... really... |
| 15:34.02 | alami | <PROTECTED> |
| 15:34.21 | alami | with exec, it work for me just nice |
| 15:34.28 | [TK]D-Fender | alami: AMI can issue it, AGI can't |
| 15:34.57 | rhamnett | file, ok thanks. Do you have any recommendations on how I can achieve something similar with dial plan logic (specifically incorporating somehow the DNS entries from the SRV) .. I guess some sort of custom AGI |
| 15:35.07 | [TK]D-Fender | (without calling an ouside shell.. which isn't AGI really anyway) |
| 15:35.18 | alami | [TK]D-Fender: what about use only exec, and catch the output and format it like i want? |
| 15:35.31 | [TK]D-Fender | alami: Sure |
| 15:35.40 | rhamnett | file, it's quite surprising that Asterisk won't handle DNS SRV .. I would see it as a pretty fundamental feature for a robust PBX. I might even have a look at implementing it |
| 15:36.01 | alami | [TK]D-Fender: all right :-) |
| 15:36.02 | file | there is a new SIP channel driver, chan_pjsip, in 12 and 13 which has support for it |
| 15:36.35 | rhamnett | file, that's useful to know. How ready is it do you know? We're doing millions of calls a month |
| 15:36.59 | file | the only way to know for your setup... is to put it in a test environment and see |
| 15:37.51 | rhamnett | file, sure I appreciate that was just getting an opinion before I even start, you might have said...barely works don't bother :) |
| 15:38.10 | file | oh it works |
| 15:38.39 | file | chan_sip has even been marked as extended support in 13+ - http://lists.digium.com/pipermail/asterisk-dev/2014-August/069709.html |
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| 15:39.54 | rhamnett | file, ok thanks for your time |
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| 15:44.13 | file | chan_pjsip: It will even make you lunch! (If you use it to call a place to order food) |
| 15:49.43 | rhamnett | file, :) |
| 15:52.52 | casdude | hey |
| 15:53.40 | casdude | just wondering if any one can help me out, reference a very strange issue that I have come across |
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| 15:54.25 | WIMPy | Not unless you tell us what it is. |
| 15:55.43 | casdude | we have a call reporting system that sits in between the telco and a customer existing phone system |
| 15:56.41 | casdude | hi, it has a two port TE220 digium card one is plugged into the phone system and the other is plugged into the phone line |
| 15:57.18 | casdude | it handles the call from one span to the other, adds some dialing rules then passes the call on |
| 15:57.26 | casdude | fine, |
| 15:57.46 | casdude | however i have seen a number of calls going through that have incomplete numbers |
| 15:57.47 | casdude | http://pastebin.com/CtLLpVUN |
| 15:57.52 | casdude | is an output |
| 15:58.35 | WIMPy | Tell the users to dial faster. |
| 15:58.36 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 15:59.03 | casdude | i am just wondering how a call can be made that is only 0845 |
| 15:59.32 | casdude | DIALNO=0845 |
| 15:59.39 | WIMPy | Because someone made too long of a puse after entering those digits. |
| 15:59.42 | casdude | i would expect this call to fail... but it works |
| 15:59.56 | WIMPy | A very common and very annoying thing. |
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| 16:01.25 | casdude | is there anything I could do to make asterisk wait |
| 16:01.35 | casdude | for the entire number |
| 16:02.09 | casdude | I'm still not 100% sure how it can actually make a successful call when it is only appears to be dialing 0845 |
| 16:02.51 | WIMPy | You're lucky if that works anyway. Normally anything dialed thereafter is discarded. |
| 16:03.22 | WIMPy | Do you live in a place where you know the length of phone numbers? |
| 16:03.30 | casdude | sure that is what I would expect it appears the 6% of the calls they make are short |
| 16:03.35 | casdude | yeh |
| 16:03.46 | casdude | they are either 6 or 11 digits |
| 16:03.51 | casdude | uk |
| 16:04.10 | WIMPy | Then increase the timeout and make you patters the known lengths. |
| 16:04.25 | WIMPy | Err. I thought the UK had more than one length? |
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| 16:10.49 | casdude | is it that the system receives 0845 starts dialing and then receives the remaining digits |
| 16:11.00 | casdude | shortly after |
| 16:11.41 | casdude | so in the console it appears as 0845 but in fact it is 0845 12345689 |
| 16:21.06 | WIMPy | yes |
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| 17:26.23 | Katty | hi tony |
| 17:26.25 | Katty | i mean anthm |
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| 17:31.33 | drmessano | Hi Katty |
| 17:32.29 | Katty | ! |
| 17:32.34 | Katty | hugs drmessano to bits! |
| 17:32.49 | drmessano | huggles Katty to bytes! |
| 17:33.06 | Katty | how's the gym been? |
| 17:33.24 | drmessano | Pretty awesome. I'm all buff and lady-killa like |
| 17:34.19 | drmessano | How the holy heckamabob have you been? |
| 17:34.23 | Katty | that's great! as long as you're not actually lady-killin |
| 17:34.36 | Katty | well. |
| 17:34.48 | Katty | i'm guessing you don't want the short answer? |
| 17:34.57 | drmessano | lol |
| 17:35.24 | drmessano | Give me whatever answer you like :) |
| 17:37.08 | Katty | 42. |
| 17:39.53 | drmessano | 99 |
| 17:40.14 | drmessano | Thats how many problems I have, and a bitch ain't one |
| 17:41.01 | drmessano | Ironically, is the number of days since we split.. So I am celebrating by eating falaffel and drinking TWO monsters |
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| 17:47.46 | [TK]D-Fender | drmessano: EEK |
| 17:48.03 | drmessano | lol |
| 17:52.19 | Katty | [TK]D-Fender: he's going to give himself a panic attack |
| 17:52.31 | Katty | [TK]D-Fender: either that or do twice as many squats later today... maybe both! |
| 17:53.41 | [TK]D-Fender | I alrady do squat... what's double nothing? :) |
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| 18:08.33 | drmessano | lol |
| 18:08.58 | drmessano | The Monsters are yummy |
| 18:09.10 | drmessano | Coffee on the other hand |
| 18:09.23 | drmessano | I think it's the traces of cocaine in the shipments, but it makes me wirey |
| 18:10.41 | Penguin | I'm trying to set up BLF on a Linksys SPA-942. How do I specify the hints' context on the phone? The blf string examples I have found do not talk about specifying the context. |
| 18:11.08 | Penguin | Although one article said I have to move my asterisk hints into the default context. |
| 18:11.22 | Penguin | I don't really like that idea very much. |
| 18:12.34 | Penguin | One example of the string is: fnc=blf+sd+cp;sub=600@$PROXY;ext=600@$PROXY |
| 18:13.49 | [TK]D-Fender | Penguin: There is no such thing as "context" on the phone |
| 18:13.50 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 18:13.56 | Penguin | sub would be used for the hint. ext would be used for the speed dial. |
| 18:14.13 | [TK]D-Fender | Penguin: It uses either the phone's peer's context, or subscribecontext if avaiable |
| 18:14.38 | Penguin | Ah, that's an answer I can deal with. I'll try the subscribe context. |
| 18:16.38 | Penguin | <PROTECTED> |
| 18:16.39 | Penguin | <PROTECTED> |
| 18:16.50 | Penguin | That's better! Let's see if the phone knows what to do now. |
| 18:17.02 | Penguin | Watchers 1 |
| 18:17.08 | Penguin | Very nice. Thanks! |
| 18:17.20 | [TK]D-Fender | You're welcome |
| 18:17.37 | Penguin | I was looking on the wrong side of things. I felt like I could define it on the phone. |
| 18:17.46 | Penguin | s/could/should be able to/ |
| 18:19.33 | Katty | hi Penguin |
| 18:20.54 | Penguin | Hello. |
| 18:21.01 | *** join/#asterisk riess82 (~riessma@62-46-213-245.adsl.highway.telekom.at) |
| 18:21.53 | Penguin | One more thing. Some howtos say to set the Shared Line Appearance to "private" while others say to set it to "shared." It looks like both values allow the subscription. Which is right? |
| 18:22.22 | [TK]D-Fender | Private |
| 18:22.28 | [TK]D-Fender | * != SLA |
| 18:28.51 | Penguin | Every time I save changes on the phone, the Watchers count goes up 1. dialplan reload and sip reload don't reset the count. |
| 18:29.30 | WIMPy | It will time out. |
| 18:30.03 | WIMPy | (or "they") |
| 18:30.09 | Penguin | Oh, is that controlled by the timeout value on the phone or one built into asterisk? |
| 18:30.35 | WIMPy | Hopefully they negotiate it. |
| 18:30.36 | Penguin | I wouldn't think one phone would subscribe 4 times and have to time out on its own. |
| 18:30.38 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 18:31.22 | WIMPy | The phone will have forgotten about the old subscriptions. |
| 18:31.34 | Penguin | But I could see the phone subscribe and not desubscribe when restarting, causing the count to keep climbing. |
| 18:39.19 | Katty | Penguin: you doing ok these days? |
| 18:39.37 | Penguin | Pretty much. Just been really busy lately. |
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| 19:44.45 | gugaua | Hello, how can I change an incomming callernumber? for example if someones number is 12341234567 it will be shown as 012341234567 |
| 19:44.53 | gugaua | I am using asterisknow 11 |
| 19:49.59 | gugaua | How to delete a 0 with callerid plugin? |
| 19:49.59 | gugaua | ${CALLERID(num)} |
| 19:51.48 | pabelanger | ~book |
| 19:51.48 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:52.01 | pabelanger | gugaua, ^ explains the basics about dialplans |
| 19:52.35 | gugaua | okay thanks :) |
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| 20:56.31 | ggayan | Hi everyone |
| 20:56.46 | ggayan | is there an explanation of what a SIP zombie channel is? |
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| 21:00.44 | Nugget | Just be sure not to compile asterisk with app_brains and you'll be safe |
| 21:03.34 | newtonr | ggayan, i think there is on the wiki. I'll see if I can find it real quick. |
| 21:04.13 | *** join/#asterisk areski (~areski@80.174.128.86.dyn.user.ono.com) |
| 21:04.35 | newtonr | ggayan, https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+Project?src=search#Asterisk12BridgingProject-WhatisaMasquerade |
| 21:05.05 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 21:05.16 | ggayan | newtonr: thanks |
| 21:05.19 | newtonr | prepare to be confused |
| 21:05.20 | newtonr | you are welcome |
| 21:05.28 | ggayan | I'm seeing a lot of zombie channels when trying to bridge a call via AMI |
| 21:05.36 | ggayan | I'm using adhearsion |
| 21:06.07 | gugaua | It´s me again, where do I need to change settings for dahdi inbound caller id manipulation? it doesn´t work with set caller id |
| 21:07.06 | [TK]D-Fender | gugaua: "core show function CALLERID" |
| 21:09.02 | gugaua | [TK]D-Fender, do I need to do this via CLI? |
| 21:09.10 | [TK]D-Fender | yes |
| 21:09.32 | gugaua | okay then I need to read a book, I have no clue how asterisk CLI works |
| 21:10.24 | [TK]D-Fender | gugaua: Doesn't take a book to conenct to CLI and read a function's instructions.... |
| 21:10.33 | [TK]D-Fender | "asterisk -rvvvvvvvvvvvvv" |
| 21:11.12 | gugaua | I am connected but I need to change a sytax in asterisk to change all numbers that are longer than 3 digits to not show the first '0' |
| 21:12.25 | *** join/#asterisk mirela666 (~mirko.bra@95.180.116.173) |
| 21:14.23 | [TK]D-Fender | gugaua: So go check the CID, then change it accordingly |
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| 21:16.49 | gugaua | okay It´s not a problem with the CId, that I can do with the gui but an incomming call will always have an additional 0 with I want to have removed... |
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| 21:21.57 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 21:23.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:24.13 | [TK]D-Fender | hates his randmly rebooting computer... |
| 21:24.45 | gugaua | [TK]D-Fender, what happend? |
| 21:24.54 | [TK]D-Fender | it dies randomly. |
| 21:25.07 | [TK]D-Fender | Freeze & reboot. |
| 21:25.09 | [TK]D-Fender | Oh well. |
| 21:25.18 | [TK]D-Fender | Anyway... go check your CID and change it accordingly |
| 21:27.13 | gugaua | I don´t want to specify a cid for external calld but I can set it for internal calls |
| 21:27.30 | gugaua | I want everyone to be able to call me |
| 21:29.21 | gugaua | Even If i change the CID accordingly I will stil have a 0 too much shown |
| 21:32.00 | [TK]D-Fender | Go change the callerID |
| 21:32.01 | gugaua | are you here? |
| 21:32.08 | gugaua | ahh |
| 21:32.22 | *** part/#asterisk ryan_turner|MTW (Ryan@2600:3c01::f03c:91ff:fe69:f4ab) |
| 21:32.26 | gugaua | the callerID at incoming route? |
| 21:32.34 | gugaua | at asterisk gui? |
| 21:32.37 | [TK]D-Fender | [17:29]gugauaEven If i change the CID accordingly I will stil have a 0 too much shown <- How will it have a 0 too many.,.. IF YOU REMOVE IT YOURSELF? |
| 21:32.50 | [TK]D-Fender | gugaua: This is not a GUI support channel |
| 21:33.16 | gugaua | okay I am not sure If you know what I want... |
| 21:33.24 | [TK]D-Fender | gugaua: In asterisk dialplan you can do whatever you want with it. If you expect your GUI to offer you some feautre for this.. then that's up to them |
| 21:33.40 | [TK]D-Fender | gugYou want to adjust the callerID of the calls coming into your system |
| 21:33.54 | [TK]D-Fender | ASTERISK can do this if you change your dialplan accordingly. |
| 21:34.02 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 21:34.25 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 21:34.28 | gugaua | okay, then I need to change the dialplan from-pstn |
| 21:35.14 | gugaua | is that right? |
| 21:35.47 | [TK]D-Fender | You would have to make your own context in extensions_custom.conf and point your channels there |
| 21:36.11 | gugaua | okay thats the first line.... |
| 21:36.12 | gugaua | [from-pstn] |
| 21:36.12 | gugaua | include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations |
| 21:37.14 | [TK]D-Fender | gugaua: No. |
| 21:37.25 | [TK]D-Fender | [17:35][TK]D-FenderYou would have to make your own context in extensions_custom.conf and point your channels there |
| 21:38.01 | gugaua | okay, but I can use the original one as a template? |
| 21:38.14 | [TK]D-Fender | no |
| 21:40.44 | gugaua | okay, so I will create a new context I call it for example [test-custom] |
| 21:41.35 | gugaua | [TK]D-Fender and there I have to put my own context in it with the setcallerid? |
| 21:45.16 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 21:53.01 | [TK]D-Fender | gugaua: You point your other devices there, match the calls yourself, and then jump back to from-pstn targeting the same exten they came in on. |
| 21:57.51 | *** join/#asterisk niekniek (~niek@77-172-219-210.ip.telfort.nl) |
| 22:04.47 | gugaua | [TK]D-Fender, Thanks for you help but I just don´t get it I pretty new to asterisk and don´t understand what you want from me :) I think I will let it as it is and try another time |
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| 23:23.24 | Milos | [TK]D-Fender, changed providers, Transfer() works fine now, before and after Answer()ing. |
| 23:23.28 | Milos | :D |
| 23:23.41 | Milos | Dial(,,o) still doesn't seem to set it though. |
| 23:29.34 | [TK]D-Fender | I continue to see nothing... |
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