00:06.37 | *** join/#asterisk Ionic (~ionic@2001:41d0:1:6276:fa1e:dfff:fedd:1f7) |
00:07.39 | Ionic | how does asterisk actually select the codec to be used for a call? |
00:07.58 | Ionic | is it trying to determine the best codec of all involved peers/channels? |
00:20.33 | pabelanger | Ionic, it is a negotiation process between asterisk and the end point, each will have a list of codecs and order. Match is made |
00:21.58 | Ionic | pabelanger: yes, kinda, I guess |
00:22.41 | Ionic | pabelanger: only in my case, I'm going through a middle point (ISDN), which is only capable of ulaw 8KHz PCM |
00:22.57 | pabelanger | then that is all you will use |
00:22.58 | Ionic | so I guess I'm stuck with that |
00:23.24 | Ionic | yeah, makes sense |
00:24.54 | Ionic | I thought maybe asterisk was capable of receiving speex32 from my SIP client and converting it back and forth, but that doesn't make a lot of sense, better use ulaw straight through |
00:26.23 | WIMPy | You could use G.722 if you still have a real phone line and someone had implemented it in any of the Asterisk channels. |
00:29.57 | Ionic | 100012 audio g722 (G722) |
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00:31.38 | volga629 | is possible hash out body message like Fee Set("Message/ast_msg_queue", "BASE64MSG=BASE64_ENCODE(Fee)") in new stack |
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00:37.01 | Ionic | hmm, but if I knew whether that's supported by the channel |
00:39.23 | Ionic | should be, though |
00:41.01 | floren | is format_mp3 still dependent on mpg123 for asterisk 11? i thought is not, i geta compile error |
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00:46.26 | Ionic | (gsm|ulaw|alaw|speex|speex16|g722|h264|speex32) hmmm... not exactly the right order |
00:46.27 | ChannelZ | yes |
00:46.34 | Ionic | and stuff like opus is missing too |
00:46.44 | ChannelZ | Who uses it? |
00:47.13 | Ionic | I defined it as the top/preferred codec in sip.conf/users.conf |
00:47.27 | Ionic | so, even while not using it, I'd expect asterisk to put it in front of the list |
00:47.28 | Imposs | Strange issue: I can call setup my voicemail box, leave messages in my voicemail box, but when I hit the voicemail button enter my password "You have no messages" |
00:48.06 | Imposs | System is Asterisk 11, FreePBX latest version |
00:48.09 | ChannelZ | have they even finished opus for asterisk? |
00:48.56 | Ionic | nevermind me, core show codecs does not list opus |
00:49.01 | ChannelZ | there's silk, if you go download it, but not opus AFAIK |
00:49.04 | Ionic | still... the list is pretty much in the wrong order |
00:51.19 | Ionic | allow=opus,speex32,speex16,speex,g722,ulaw,alaw |
00:51.20 | Ionic | hmhm |
00:52.30 | Ionic | well, I could try disallow=all first |
00:53.44 | Ionic | meh, now GSM is disabled, but the order is still messed up |
00:53.59 | Ionic | chan_sip.c:13084 add_sdp: ** Our capability: (ulaw|alaw|speex|speex16|g722|h264|speex32) |
00:57.58 | Ionic | oh, great |
00:58.07 | Ionic | disallow = all in users.conf is giving me speex32 |
00:58.12 | Ionic | wait.. what? |
00:58.34 | volga629 | yes, got fixed exten => _X.,n,Set(BASE64MSG=${BASE64_ENCODE(${MESSAGE(body)})}) |
00:58.57 | volga629 | thank you Everyone for hints and help |
00:59.29 | Ionic | Our capability: (ulaw|speex|speex16|g722|speex32) |
00:59.49 | Ionic | and then speex32 gets selected... that's not making any sense, UNLESS the last entry is the most significant one |
01:00.14 | Ionic | but then again, why was ulaw selected before |
01:01.18 | Ionic | chan_sip.c:7911 sip_new: *** Our native formats are (speex32) < hum... |
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01:03.07 | Ionic | probably because my client prefers speex32... but... oh well |
01:03.07 | Ionic | *moving g722 up* |
01:07.14 | Ionic | well, that didn't help either... |
01:07.26 | Ionic | I'm confused, but at least it's using speex32 now |
01:15.06 | WIMPy | IIRC it makes a difference if you use multiple allow lines instead of one with a list. |
01:16.36 | Ionic | I'm now using multiple disallow lines as well |
01:17.03 | Ionic | Confused, but stopping right here, I've found some worse problem than encodings |
01:17.16 | WIMPy | The usual story. |
01:19.49 | Ionic | calls keep hanging around in the capi channel list even after they are terminated/disconnected and a hangup request won't work either, gotta sort this out first |
01:20.11 | Ionic | reloading the channel helps, but that's merely a workaround |
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02:30.37 | floren | i have a quick question related to compiling asterisk with a custom /var/lib location |
02:31.43 | floren | make ASTVARLIBDIR=/usr/share/asterisk will not work for example |
02:32.21 | floren | what is the proper variable to use? I'm looking at the source code and I see a lit of references in this format: AST_VAR_DIR |
02:33.08 | Ionic | why would it? |
02:33.18 | Ionic | use the correct configure switch, --libdir |
02:33.25 | Ionic | check ./configure --help |
02:33.40 | floren | ya |
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02:55.24 | floren | is not working, i set %configure --sharedstatedir=/usr/share instead of /var/lib and it still compiles into /var/lib |
03:19.58 | Ionic | wth |
03:20.09 | Ionic | --sharedstatedir isn't even used by asterisk I guess |
03:20.33 | Ionic | why don't you just set --libdir=/usr/share? |
03:20.44 | Ionic | and why would you have libraries in /usr/share anyway? |
03:20.59 | Ionic | that's not sane and against any established FSH |
03:22.43 | Ionic | sharedstatedir is not used by *any* makefile as far as I can see |
03:23.53 | Ionic | please, I beg you, don't try to change options you don't understand, this will only lead you to broken system... |
03:24.31 | Ionic | s/lead you to/lead to a/ |
03:29.22 | WIMPy | huh? |
03:30.00 | WIMPy | The great things about unixoid OSs is that you can change almost everything. What's your issue with that? |
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03:47.42 | Ionic | WIMPy: in theory, yes, but first and foremost /usr is vendor land and then there are file system hierarchy rules to make life less of a pain |
03:51.59 | Ionic | anyway, I'm stepping to bed, night and good luck |
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05:53.06 | hrolf | Hi #asterisk, how can we do a supervised (2 step) transfer in Asterisk? |
05:54.47 | WIMPy | Check your terminals manual or features.conf. |
05:55.05 | hrolf | WIMPy: What is terminals manual? |
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05:56.06 | WIMPy | Terminal= Phone or ATA ore whatever you might be using. |
05:56.34 | hrolf | WIMPy: I'm using SIP |
05:56.48 | hrolf | WIMPy: SIP trunk between Avaya and Asterisk |
05:57.07 | WIMPy | Then check the Avayas manual. |
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06:04.20 | hrolf | WIMPy: But I want to do it from Asterisk's side not Avaya. |
06:04.30 | WIMPy | With what? |
06:05.45 | hrolf | WIMPy: With the SIP trunk established with Avaya. That is, call is received on the SIP trunk from Avaya on Asterisk. Now do a 2-step xfer from Asterisk to an extension in Avaya. |
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06:06.52 | WIMPy | I'd use a phone. |
06:07.33 | WIMPy | Check your terminals manual or features.conf. <= two possibilites to do transfers on Asterisk. |
06:08.38 | hrolf | WIMPy: One question, how does Asterisk do the transfer internally? Does it uses Transfer()? |
06:08.56 | hrolf | .. in case of 2-step xfer. |
06:09.28 | WIMPy | If you want to send a call to another URI, you can use Transfer(). |
06:10.16 | WIMPy | If someone wants to do it manually, he will use his phone or DTMF features. That's not dialplan. |
06:12.21 | hrolf | WIMPy: I want to do it through the dialplan, so in that case I believe features.conf doesn't have anything to do, right? |
06:13.17 | WIMPy | Correct, but doesn't fit your original question. |
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06:14.31 | hrolf | WIMPy: Means that 2-step transfer is only for phones, not dialplan? |
06:14.41 | WIMPy | Transferring autonatically and attended don;t go together. |
06:15.08 | WIMPy | How are you going to do that if you only have a caller and no callee? |
06:17.19 | hrolf | WIMPy: I was under the impression that, we can do 2step transfer in dialplan like, Dial(), then after the called party answers, use function XXXXX() to complete the 2-step xfer? |
06:17.57 | WIMPy | The dialplan won't continue before one party hangs up. |
06:18.10 | hrolf | WIMPy: I see. |
06:18.52 | WIMPy | And even that will only happen if you pass the right parameter to Dial(). |
06:20.02 | hrolf | WIMPy: the 'g' paramtere? |
06:21.10 | WIMPy | yes |
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06:21.51 | hrolf | WIMPy: Okay so what will be that XXXXX() application/function which will then complete the 2-step xfer? |
06:22.05 | WIMPy | Your phone. |
06:22.27 | hrolf | WIMPy: There is no phone, we are in dialplan. |
06:22.28 | WIMPy | It starts as a 2nd call. |
06:22.50 | WIMPy | Then you're not talking to someone. |
06:23.07 | WIMPy | What do you want to do? |
06:24.23 | hrolf | WIMPy: Call received from Avaya on Asterisk (through the SIP trunk), play IVR, then do a 2-step xfer to an extension on Avaya. |
06:24.44 | hrolf | WIMPy: We tried Transfer() (blind transfer) but it is not working. |
06:24.58 | WIMPy | Where is the 2nd step when there is only one person involved in the call? |
06:26.06 | WIMPy | Transfer() tries to redirect the user away from Asterisk to anothr URI. If you just want to send the caller to some other device, use Dial(). |
06:27.01 | hrolf | WIMPy: Please some clarification, you mean to say 2-step transfer only happens when 2 persons are involved? |
06:27.29 | WIMPy | Doesn;t make any sense to me otherwise. |
06:27.52 | WIMPy | But maybe you should explain what "2-step transfer" means to you. |
06:30.14 | hrolf | WIMPy: (Sorry if I'm using the wrong terminology), but what I mean by 2-step xfer, is that a call is received on Asterisk, we do Dial(), the dialed party picks up, then the call is handed over to the dialed party, the one who intiaited the dial drops. |
06:30.46 | WIMPy | But there's no call left then. |
06:31.16 | hrolf | WIMPy: There is, the original caller. |
06:31.36 | WIMPy | And who else? |
06:31.57 | hrolf | WIMPy: And the one to whom we just 2-step xfered. |
06:32.37 | WIMPy | Maybe you should start all over again. That doesn't make any sense at all. |
06:32.49 | WIMPy | You said the caller is in an IVR. |
06:33.29 | hrolf | Call received from Avaya to Asterisk on SIP trunk. We played the IVR, now IVR wants to transfer to an extension on Avaya. So we can do Transfer(), or Dial(). |
06:33.33 | hrolf | Transfer() is not working. |
06:33.48 | hrolf | Dial() uses another channel (we don't much licenses.) |
06:34.13 | hrolf | So I'm saying we can try the other method 2-step transfer to give the call back to an extension on Avaya. |
06:35.01 | WIMPy | Then Transfer() is your only option. |
06:35.19 | WIMPy | But maybe the Avaya won't accept that. |
06:35.26 | hrolf | WIMPy: Yes. |
06:35.36 | hrolf | WIMPy: Do you know why? |
06:36.05 | WIMPy | Security configuration? |
06:36.11 | WIMPy | Just not implemented? |
06:36.46 | WIMPy | Maybe it's cheaper if you connect them via PRI instead of SIP? |
06:37.32 | hrolf | WIMPy: Okay. |
06:37.41 | hrolf | WIMPy: The customer will have to buy a card then. |
06:38.07 | hrolf | WIMPy: One thing, can you explain what is a 2-step xfer. Is it even possible in the scenario I described? |
06:38.43 | WIMPy | I just set up an Asterisk with PRI card for someone who wanted to use SIP with his PBX. Two PRI cards cost less thatn the SIP licenses. |
06:39.08 | WIMPy | I still don;t know what you mean by 2-step transfer. |
06:40.44 | hrolf | WIMPy: Like from in the dialplan, after playing IVR, you do another Dial(,'g'), the dialed party answers, the caller (who called the IVR) is transfered to that dialed party. |
06:41.49 | WIMPy | Usually the underlying channel will do so automatically if possible and configured. |
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06:42.34 | WIMPy | But I'm not sure if that would be media only for SIP. I never dared to try. |
06:44.47 | hrolf | WIMPy: Okay. Can you tell what is the propery format for Transfer()? |
06:45.37 | WIMPy | A full URI. |
06:46.05 | hrolf | WIMPy: Is it Transfer(sip/avaya/252) where sip/avaya is the SIP user I have defined in my Asterisk (with it's host=<IP Addr of Avaya) and 252 is the extension on Avaya? |
06:46.15 | hrolf | WIMPy: On version 1.6 |
06:46.37 | WIMPy | But if I read the info correctly, that would only do what you want if the call hasn't been answered yet. |
06:47.18 | WIMPy | AFAIK it's a URI, not a peer. |
06:47.52 | WIMPy | sip/destination@avaya.local or something, but see above. |
06:48.07 | hrolf | WIMPy: The documentation for Asterisk 11, says Transfer(Tech/destination), isn't that Transfer(SIP/avaya/252) ? |
06:48.53 | WIMPy | I'm pretty sure I read that that does not work. |
06:49.20 | hrolf | WIMPy: I have tried Transfer(sip/avaya/252) between two Asterisk servers (assuming avaya's host=<IP Addr of the other Asterisk>) and it worked. |
06:49.30 | hrolf | WIMPy: But a full URI didn't |
06:49.35 | WIMPy | But maybe it will work if the call was answered, but then it will use two channels as I understand that. |
06:49.48 | hrolf | WIMPy: and no two channels were used. |
06:50.16 | WIMPy | Well, ok, then you know how to do it. |
06:50.52 | hrolf | WIMPy: Yes, but I thought may be due to the wrong format it is not working with Avaya etc. |
06:51.18 | WIMPy | You can always go and read the sip debug. |
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07:02.00 | hrolf | WIMPy: Can you take a look at the logs? I have both from avaya's side and Asterisk's side. (SIP debug) |
07:02.07 | hrolf | http://pastebin.com/Z62QMm5G |
07:02.11 | hrolf | http://pastebin.com/27PksdbT |
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08:38.45 | tparcina | "voicemail show users for default" shows that users have messages, but there are no dirs in /var/spool/asterisk/voicemail/default/ |
08:38.50 | tparcina | How can this be? |
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08:42.55 | Schreda | HEy everybody |
08:43.17 | Schreda | I would need some help and it would be nice to get an feedback :D |
08:44.17 | Schreda | Is it possible to have 2 different dialplans but with same extensions? |
08:44.18 | Schreda | e.g |
08:44.56 | Schreda | I have one phone which can be called by using an extension e.g 1 |
08:45.23 | Schreda | after some time and nobody answered this call is forwarded to other phones ... This works fine |
08:47.28 | Schreda | The problem now is that I get a call on a other phone and if this phone isnt answered I call all phones incl. extension 1... but now an dial circle exist because if the extension 1 is not reply after a certain time... So I want to use 2 different extension configuration depending on which extensions got called... |
08:47.32 | Schreda | is it possible? |
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10:33.13 | Mads | Hello, i'm having a problem with the Asterisk cdr csv file. As far as i can see the csv file contains 16 fields my problem is within field 3 (destination). When call forwarding is enabled the destination is not showing the number that is forwarded to but the number that is called in first place, is this the normal way it works or some bug? |
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11:16.28 | LooserOuting | does asterisk support ilbc mode 20 |
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12:00.00 | kchehab | hi i am using 1.8.16 , trying to make the rtp of calls pass by my asterisk i set in sip.conf directmedia=yes directmediadeny=0.0.0.0/0 canreinvite = no |
12:00.07 | kchehab | but still not working please advice |
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12:15.23 | kchehab | please any one can help ? |
12:16.50 | Gugge | kchehab: if you dont want directmedia, why do you set it to yes? |
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12:33.00 | Mads | Hello, i'm having a problem with the Asterisk cdr csv file. As far as i can see the csv file contains 16 fields my problem is within field 3 (destination). When call forwarding is enabled the destination is not showing the number that is forwarded to but the number that is called in first place, is this the normal way it should be? |
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12:44.47 | Greenlight | kchehab: You want RTP to bypass your Asterisk box? For this to happen the codecs etc need to match exactly |
12:46.25 | Greenlight | Mads: Yes |
12:49.42 | kchehab | GameGamer43 yes i want that |
12:50.07 | kchehab | Gugge soeey i set it to no in my config ,this was a mistake i fix it and still have the problem |
12:50.29 | kchehab | Gugge RTp is still oassing directly ,and i am debugging using rtp set debug on |
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12:50.44 | Greenlight | So there are a number of things that will prevent direct RTP |
12:51.00 | Greenlight | Are you call recording? Are you detecting DTMF? Are the codecs different? |
12:51.48 | kchehab | Greenlight no |
12:52.16 | kchehab | Greenlight is my config has a right syntax |
12:52.40 | Greenlight | directmedia=yes on the peer(s) is all that's needed |
12:53.00 | jmetro | or directmedia=no. some people prefer that. |
12:53.04 | kchehab | Greenlight using sip show setting i can see Direct RTP setup: No |
12:53.26 | Greenlight | jmetro: If the goal is to enable direct media, I don't see why you'd set that .... |
12:53.41 | kchehab | Greenlight i will set directmedia=no on user and trunk |
12:53.44 | kchehab | and try |
12:53.51 | Greenlight | Umm |
12:53.57 | Greenlight | Sure... good luck with that. |
12:53.59 | *** join/#asterisk Draecos (~Draecos@101.112.150.68) |
12:54.56 | jmetro | if you WANT it, then directmedia=yes indeed |
12:55.03 | jmetro | it seems like it causes issues though |
12:55.12 | kchehab | Greenlight i think it works yes |
12:55.18 | kchehab | thanks alot |
12:55.28 | Greenlight | shurgs |
12:57.42 | Mads | My problem is that the guy who is forwarding his phone needs to be billed for the forwarding call but the CSV file is just logging the call as a standard incomming call that should not be billed. |
12:58.02 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
12:58.17 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-sffwicygczovyoda) |
12:58.56 | *** join/#asterisk ThothCastel (~chatzilla@37-110-251-66.g3ns.net) |
12:59.31 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
13:00.03 | Greenlight | Mads: Are you sure you don't get a *second* CDR for the forwareded call ? |
13:00.14 | ThothCastel | why do I keep on getting 'failed for 192.168.100.46:33992' - wrong pasword' error message when trying to register a sip account user (Zoiper Communication' softphone |
13:00.46 | eirirs_ | network problems |
13:00.47 | Greenlight | ThothCastel: Going to take a wild stab in the dark guess here - are you using the wrong password ? |
13:00.55 | eirirs_ | haha |
13:00.59 | Mads | Greenlight there is no second CDR for the forwarded call |
13:01.00 | ThothCastel | Greenlight: NO |
13:01.13 | ThothCastel | sorry, meant no |
13:02.01 | Greenlight | Does the account exists and you've reloaded if neccissary ? |
13:02.02 | ThothCastel | Greenlight: I added [1001] |
13:02.09 | ThothCastel | username=1001 |
13:02.18 | ThothCastel | secret=QwEr1QwEr2q |
13:02.31 | ThothCastel | all these to the end of the sip.conf file |
13:02.48 | Greenlight | You're *sure* you've reloaded ? |
13:02.52 | ThothCastel | reloaded? well, I reboot the OS |
13:02.58 | Greenlight | Okay |
13:03.17 | Greenlight | And you've not mistyped the password ? |
13:03.59 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
13:04.56 | Greenlight | Mads: I would have expected a 2nd CDR. Perhaps if you enable unanswered calls CDR's it will show |
13:05.10 | Greenlight | Unless - how are you doing the redirect ? |
13:08.38 | ThothCastel | Greenlight: do I need to add the 1001 extension to the extensions.conf file as well? |
13:09.16 | Greenlight | Not for it to register, no. |
13:09.35 | Greenlight | Lets see your sip.conf |
13:09.38 | Greenlight | ~pb |
13:09.38 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
13:09.41 | Greenlight | ^^^ |
13:10.42 | Mads | Greenlight the call forwarding is done through an extension, if a forwarded call is unanswered it does show the unanswered call as expected :) |
13:11.20 | ThothCastel | Greenlight: http://pastebin.com/8XaUEXGN |
13:12.26 | Greenlight | tupe=friend |
13:12.30 | Greenlight | That should be type |
13:12.35 | Greenlight | For starters |
13:12.41 | jmetro | s/tupe/type |
13:13.45 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
13:14.11 | Greenlight | Fix that, and I bet it'll work |
13:14.55 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
13:15.25 | ThothCastel | Greenlight: thank you, I updated and did a 'core restart now' and then 'asterisk -r' |
13:15.41 | ThothCastel | I tried registering the softphone now and still the same error msg |
13:16.32 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
13:17.35 | ThothCastel | Greenlight: this is the error msg http://pastebin.com/iAbHyesK |
13:17.44 | Greenlight | Oh. Add type=friend and context=LocalSets, host=dynamic |
13:17.58 | Greenlight | DIdn't spot that it wasn't inheriting |
13:19.47 | ThothCastel | Greenlight: these are already added lines 11, 13 and 15 respectivelly |
13:20.06 | Greenlight | You're trying to register 1001 though |
13:20.49 | ThothCastel | right, I see so I should add those to each of the [device] ? |
13:21.05 | Greenlight | Or add (office-phone) after the [1001] |
13:21.12 | Greenlight | To get it to inherit the template |
13:21.14 | ThothCastel | won't [1001] get the settings from the (!) template? |
13:21.51 | Greenlight | If you tell it to, yes |
13:22.10 | Greenlight | As you have with your other devices |
13:22.21 | ThothCastel | yey! Thanks Greenlight. it worked now that I added the (office-pone) to it |
13:22.29 | ThothCastel | It registered no |
13:22.32 | ThothCastel | now |
13:22.35 | Greenlight | Excellent |
13:22.52 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
13:22.57 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
13:23.13 | ThothCastel | Greenlight, so on the way it is now, if I register a second softphone with a different extension number, then I will be able to place a call from ext 1001 to ext 1002 (eg) |
13:23.15 | ThothCastel | ? |
13:23.56 | ThothCastel | Or I still need to configure some other stuff? this is my first time setting up asterisk |
13:24.22 | Greenlight | You would need to configure that in your extensions.conf |
13:24.46 | Greenlight | Specifically in the LocalSets context that you've set them to go to |
13:25.24 | Greenlight | At a very basic level, you could define exten => _XXXX,1,Dial(SIP/${EXTEN}) |
13:25.31 | Greenlight | same => n,Hangup |
13:25.40 | *** join/#asterisk Rumbles (~Rumbles@77.107.183.230) |
13:25.51 | Greenlight | So when a 4 digit number was dialled it'll try and find a matching SIP peer |
13:27.21 | *** join/#asterisk Rumbles (~Rumbles@77.107.183.230) |
13:27.40 | ThothCastel | Greenlight: right, where EXTEN = 1001? or where XXXX = 1001 |
13:28.56 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nynxbbzttjobnfeh) |
13:28.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
13:29.12 | Greenlight | The _XXXX will act as a "catch" for any 4-digit number dialled. ${EXTEN} will be subsituted with that 4 digit number. |
13:32.12 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
13:33.10 | bulkorok | hi... how can I configure Asterisk that SendFax() send RTP after receiving 183 SDP and not only after 200 OK ?! |
13:34.41 | jmetro | http://lmgtfy.com/?q=asterisk+send+rtp+after+receiving+183&l=1 |
13:35.21 | jmetro | i dont know anything about fax. |
13:35.28 | jmetro | but i found your answer. |
13:36.01 | ThothCastel | Greenlight: thank you :) |
13:41.56 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
13:43.32 | vlad_starkov | Question: Anyone know is it possible to retrieve callerid information from parked call in exact parking lot? |
13:44.01 | jmetro | vlad_starkov: yes. |
13:44.11 | vlad_starkov | jmetro: could you point me please |
13:44.12 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-yjsvvlhfrvfgirrd) |
13:44.43 | jmetro | vlad_starkov: i know Aastras can do it, via XML apps. |
13:44.45 | vlad_starkov | jmetro: what I need is to retrieve callerid of parked call before picking up that call |
13:45.11 | vlad_starkov | jmetro: I mean to do it in * dialplan |
13:45.13 | jmetro | vlad_starkov: i actually have a button on my phone that shows the first 9 parked spots, caller ID's |
13:46.50 | *** join/#asterisk Rumbles (~Rumbles@host-92-27-117-96.static.as13285.net) |
13:49.05 | jmetro | I dont know how to do it in dialplan, but on the phone is kind amore useful isnt it? |
13:52.48 | vlad_starkov | jmetro: it is useful to make it in dialplan when you write some logic |
13:53.32 | bulkorok | jmetro: the r is not set in my Dial |
13:55.27 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
13:55.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
13:55.55 | *** join/#asterisk btracht (~btracht@70.89.37.217) |
13:56.34 | *** join/#asterisk Caplain (~shayne@2604:8800:123:0:a8dc:ede4:af2a:eca6) |
14:03.11 | bulkorok | so... anither idea why sendfax does not begin rtp after 183 but after 200 ?! |
14:04.59 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
14:05.11 | Katty | hello my asterisk does not work at all how to fix plz is urgent thx |
14:05.40 | jmetro | Katty: go to www. |
14:05.51 | Katty | jmetro: what is www. plz |
14:06.00 | jmetro | www.girlsgogames.com/ they have manuals for fix urgent plz |
14:06.07 | btracht | i am running trixbox v 2.8.0.4 and trying to decipher the CDR report. what does it mean when the channel is SIP/the servers own ip? |
14:07.14 | *** join/#asterisk Korrosion (~nate@70.89.37.217) |
14:08.03 | Katty | hugs on jmetro |
14:08.08 | jmetro | :3 |
14:08.14 | jmetro | Mornin lass |
14:08.24 | Katty | how's things out your way? |
14:08.32 | btracht | or when the source is root? |
14:08.32 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
14:08.41 | jmetro | dead hot. But air conditioned. |
14:08.48 | Katty | yay air conditioning! |
14:09.21 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
14:09.21 | *** mode/#asterisk [+o Qwell] by ChanServ |
14:09.31 | igcewieling | btracht: Could be a number of reasons. Unfortunately, not many people will want to deal with Trixbox |
14:09.35 | *** join/#asterisk jsjc (~Adium@178.Red-79-150-247.dynamicIP.rima-tde.net) |
14:09.57 | btracht | number of reasons such as? |
14:10.25 | btracht | why not deal with trixbox? |
14:10.31 | Katty | there is a channel for trixbox |
14:10.39 | Katty | it's different. |
14:10.44 | igcewieling | btracht: someone hacked into your server, you don't have allowguest=no or you have a config issue on your peers or the call had some kind of off forwarding or routing |
14:10.57 | Katty | i blame squirrels. |
14:10.58 | igcewieling | ~trixbox |
14:10.59 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
14:12.28 | Greenlight | My favourite infobot description ^^ |
14:12.56 | btracht | we are in the process of converting from trixbox but in the mean time i am trying to figure out whats going on from the current server |
14:14.23 | [TK]D-Fender | trixbox died about 3 years ago |
14:14.32 | [TK]D-Fender | And they ran forked versions of Asterisk and FreePBX. |
14:14.35 | [TK]D-Fender | None of which we support |
14:14.42 | slav3_kitten | may it rest in peace |
14:18.35 | jmetro | btracht: look at the UI and make a flowchart. |
14:18.40 | jmetro | btracht: then hand-code it. |
14:18.51 | jmetro | btracht: or just call yourself and see what happens, flowchart, code. |
14:19.25 | [TK]D-Fender | [10:10]igcewielingbtracht: someone hacked into your server, you don't have allowguest=no or you have a config issue on your peers or the call had some kind of off forwarding or routing |
14:19.46 | [TK]D-Fender | This was prety much the answer though as it was a CDR question. That is simply what the channel looked like when it came in |
14:20.08 | [TK]D-Fender | Quite possibly a bad trunk setup failing over to an un-authed call |
14:25.45 | *** join/#asterisk serafie (~erin@206.255.84.113) |
14:25.59 | igcewieling | btracht: you'll need to comb through the Asterisk logs to find the problem call and see what actually happened to cause the issue. |
14:27.39 | Katty | good morning fender bender. did you have a good weekend? |
14:28.15 | bulkorok | jmetro: another idea for no RTP after 183 ?! |
14:28.19 | *** join/#asterisk vlad_starkov (~vlad_star@nat.canmos.ru) |
14:28.48 | jmetro | bulkorok: probably google, i dont know anything about fax at all. |
14:29.18 | bulkorok | jmetro: k... google is not that helpfull in that case |
14:29.38 | igcewieling | fax does not use RTP, so your question doesn't make a lot of sense. |
14:30.10 | bulkorok | well how are the "fax tones" aka CNG transmitted!? |
14:31.26 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
14:31.26 | igcewieling | bulkorok: using RTP, but that is not actually fax. Have you confirmed other applications do not exibit the same issue? Have you tried the z option to SendFax? |
14:31.54 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
14:35.11 | igcewieling | no RTP after a 183 sounds like a more general sip.conf issue rather than something specific to sendfax. |
14:36.41 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
14:36.49 | bulkorok | I didn't try z opetion... |
14:37.15 | bulkorok | Dial has not the r option (not any option...) |
14:37.46 | Cuzner | That trixbox bot description is great |
14:38.28 | Cuzner | I used to set-up trixbox's for small/medium businesses, it did what it was supposed to do... |
14:38.46 | Katty | pokes [TK]D-Fender |
14:38.46 | igcewieling | The "r" option to dial is not useful in any case where you are not getting ringing when you should. |
14:38.57 | bulkorok | agree |
14:39.11 | igcewieling | Cuzner: We've all done things in our lives which we are not proud of. |
14:39.24 | Cuzner | igcewieling: I never said i was ashamed |
14:39.26 | bulkorok | but which option changes the 183 behaviour?! |
14:39.39 | Cuzner | it was good for what it was, quickly set-up PBX in a box |
14:39.40 | igcewieling | Cuzner: You didn't have to. We are ashamed enough for you. |
14:39.46 | [TK]D-Fender | prods Katty |
14:39.58 | Cuzner | asterisk 1.4 + FreePBX + whatever the hud was called |
14:40.14 | Katty | [TK]D-Fender: how was your weekend, dear? |
14:40.15 | Cuzner | and it did what it was supposed to do, i never had a problem with it. |
14:40.26 | Cuzner | it was better than everything else at the time |
14:40.33 | Cuzner | Elastix? wtf was that... |
14:40.49 | igcewieling | bulkorok: nothing obvious in sip.conf.sample included in your Asterisk source code? like something to do with progress and early media? |
14:40.50 | leifmadsen | Cuzner: it's what I wrapped your balls in |
14:40.58 | leifmadsen | oh shit, that wasn't a privmsg?! |
14:40.58 | Cuzner | oh my |
14:41.07 | leifmadsen | carry on! nothing to see here! |
14:41.26 | Cuzner | leifmadsen: can i get your signature? :) |
14:41.31 | leifmadsen | sure! :) |
14:41.31 | igcewieling | adds another checkbox next to "leifmadsen" |
14:41.33 | leifmadsen | $10 |
14:41.42 | leifmadsen | igcewieling: gold star? |
14:41.52 | Katty | LEIF |
14:41.56 | leifmadsen | KATTY |
14:42.00 | *** join/#asterisk zrzerenato (bd2962c6@gateway/web/freenode/ip.189.41.98.198) |
14:42.01 | Katty | hello, dear |
14:42.01 | leifmadsen | Katty: fudge. make it. |
14:42.07 | zrzerenato | im looking forward to change a priority on certain port that i use, example: set priority highest port (8765) TCP. ???? |
14:42.11 | Katty | leifmadsen: why? |
14:42.12 | leifmadsen | Katty: deliciousness ensues. |
14:42.16 | Katty | leifmadsen: it's easy you know. you could make it |
14:42.21 | leifmadsen | Katty: indeed. |
14:42.25 | leifmadsen | I'm just saying, you won't regret it |
14:42.26 | Katty | leifmadsen: oh, so you've had it? excellent. |
14:42.29 | Cuzner | Leaf pronounces his name Life, btw. You all have been saying it wrong this whole time. |
14:42.33 | igcewieling | zrzerenato: Try /join #TheDistroYouUse |
14:42.35 | Katty | leifmadsen: oh, no. i wouldn't. well. kind of. lots of calories :< |
14:42.42 | Katty | leifmadsen: but i have made it before! thank you for thinking of me, regardless |
14:42.45 | leifmadsen | Katty: ikr? :) |
14:42.54 | Katty | leifmadsen: if you run across anything else tasty, let me know. |
14:43.04 | leifmadsen | Katty: ok! |
14:43.11 | leifmadsen | Katty: rum cookies |
14:43.14 | Katty | leifmadsen: that miniature hooman of yours is growing up so fast! |
14:43.20 | Katty | leifmadsen: link? |
14:43.20 | jmetro | blueberry baked goods. i dont care what. |
14:43.28 | leifmadsen | vanilla rum balls :D |
14:43.30 | jmetro | blueberry pound cake, muffins, scones. |
14:43.31 | Katty | jmetro: i've got a killer blueberry muffin recipe, if you're interested. |
14:43.39 | leifmadsen | Katty: I just used google |
14:43.42 | leifmadsen | Katty: he is huge now.... |
14:43.44 | Katty | leifmadsen: ah, right. ok |
14:43.45 | btracht | igcewieling: this is one of the problem calls in our asterisk log - http://pastebin.com/UshLqmMQ |
14:43.58 | Katty | leifmadsen: ohoh |
14:44.05 | Katty | leifmadsen: i found HobNob cookies! |
14:44.13 | Katty | leifmadsen: do they have those up there in Canada? |
14:44.18 | zrzerenato | igcewieling: srry , but im trying there already but no answers. And it is asswell about connections between cti, and starphone,. im having lots of connection_lost, with my asterisk. |
14:44.19 | leifmadsen | I DON'T KNOW BUT MAYBE |
14:44.27 | [TK]D-Fender | btracht: Your r call is not matching a trunk you set up. |
14:44.32 | Cuzner | we have poutine up here! |
14:44.33 | [TK]D-Fender | btracht: So fix your trunk |
14:44.35 | Cuzner | that's all you need |
14:44.38 | Katty | locates a photo |
14:44.39 | igcewieling | btracht: looks pretty obvious to me. Received incoming SIP connection from unknown peer to +972598461260") in new stac |
14:44.46 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
14:45.12 | Katty | leifmadsen: http://ukonlinegoodies.com/images/mcvitties-dark-choc-hobnobs-300g.jpg <- addictive. |
14:45.25 | igcewieling | zrzerenato: English is not your native language? |
14:45.26 | bulkorok | igcewieling: I used the sample sip.conf... allowoverlap=no directmedia=yes ignoresdpversion=yes |
14:45.37 | btracht | in our CDR the channel is SIP/servers ip address, source is root and CLID is "root"<root> |
14:45.52 | igcewieling | btracht: correct. Because there is no peer for that call |
14:45.59 | jmetro | btracht: whats the destination? what is it doing? |
14:46.28 | zrzerenato | igcewieling: nope, but it is readable? rsrs |
14:46.39 | igcewieling | btracht: try prematuremedia and progressinband options. |
14:46.42 | [TK]D-Fender | btracht: Fix your trunk |
14:46.56 | igcewieling | zrzerenato: we have no idea what you are talking about. |
14:46.56 | [TK]D-Fender | igcewieling: It isn't matching... |
14:47.16 | btracht | what do you mean by fix your trunk? other calls work. could it be because this is an international call? |
14:47.21 | igcewieling | sorry ths is for bulkorok: try prematuremedia and progressinband options. |
14:47.32 | [TK]D-Fender | zrzerenato: You seem to be asking a core networking question. This is something you should ask in your distro's channel |
14:47.38 | bulkorok | igcewieling: btracht: try prematuremedia and progressinband options. <= is that for me?! |
14:48.24 | igcewieling | btracht: add "allowguest=no" and "alwaysauthreject=yes" to your setup. that should fix the issue. |
14:48.26 | Cuzner | leifmadsen: your comment made me giggle, but i'm in a video conference with Tom right now, so not the best timing :P |
14:48.39 | leifmadsen | Cuzner: I disagree |
14:48.58 | Cuzner | you'd argue it was perfectly timed, i know. |
14:48.59 | zrzerenato | [TK]D-Fender: thats correct. so it is understanding . my distro is centos, and im trying to ask there but seems no one is online, coz there is not chat on the channel |
14:49.02 | btracht | igcewieling: to my trunk details? |
14:49.02 | [TK]D-Fender | igcewieling: Nope, won't make his peer match any better... he needs to fix that direct |
14:49.16 | [TK]D-Fender | btracht: Make your trunk match |
14:49.27 | igcewieling | btracht: no. to sip_general_custom.conf |
14:49.38 | [TK]D-Fender | btracht: verify the auth your provider is sending (or not) with their calls to you ... and set your trunk accordingly |
14:49.45 | igcewieling | [TK]D-Fender: he doesn't understand enough to make is "trunk match", |
14:50.04 | igcewieling | once he adds the options I recommended and some of his calls break, THEN he will be ready to fix his trunk. |
14:50.34 | [TK]D-Fender | s/some/all |
14:51.52 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
14:51.52 | *** mode/#asterisk [+o sruffell] by ChanServ |
14:52.04 | Korrosion | What is he matching the trunk to? |
14:57.04 | [TK]D-Fender | it ISN'T matching. That is the problem |
14:58.18 | Korrosion | matching what though? |
14:59.00 | [TK]D-Fender | Matching the CALL clearly... |
14:59.59 | *** join/#asterisk Rumbles (~Rumbles@77.107.183.253) |
15:00.16 | btracht | again could it be because it's an international call? |
15:00.36 | [TK]D-Fender | Shouldn't be |
15:00.39 | btracht | i don't have international calling enabled |
15:00.44 | [TK]D-Fender | No. |
15:00.51 | [TK]D-Fender | This has nothng to do with ROUTES |
15:01.50 | btracht | i don't understand what in the trunk matches |
15:02.04 | btracht | or doesn't match |
15:02.43 | jmetro | btracht: i think he means to check authentication with the trunk [your provider] |
15:03.22 | [TK]D-Fender | And look at the CALL |
15:03.38 | igcewieling | btracht: did you add the two options I suggested. |
15:03.44 | btracht | i did |
15:04.08 | igcewieling | btracht: then the issue should be resolved. |
15:04.21 | igcewieling | btracht: you are running a GUI, correct? |
15:04.45 | btracht | i will check with my provider but my concern is that the CDR says the call was answered |
15:05.06 | igcewieling | btracht: with the options I gave you, you should not get any more calls like that. |
15:05.40 | btracht | ok, thank you |
15:06.17 | igcewieling | btracht: not having those options HIDES the real issue. |
15:06.45 | btracht | i would like to know what the real issue is |
15:07.19 | [TK]D-Fender | The call is not matching yoru trunk. |
15:07.27 | [TK]D-Fender | This is not a difficult concept |
15:08.02 | btracht | if it's not matching the trunk why does the CDR say it was answered |
15:08.26 | Greenlight | Going around in circles here aint we |
15:09.23 | [TK]D-Fender | [11:08]btrachtif it's not matching the trunk why does the CDR say it was answered <- it was accepted as a UNAUTHED CALL. |
15:09.27 | [TK]D-Fender | brtjsut like it said |
15:09.57 | [TK]D-Fender | And is making a shit job out of viewing your CDRs |
15:19.45 | *** join/#asterisk jetlag (~jetlag@pool-71-168-192-247.cmdnnj.east.verizon.net) |
15:21.11 | *** join/#asterisk Captain_Proton (~quassel@173.162.32.1) |
15:24.26 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
15:24.29 | Captain_Proton | need some help I upgraded to Asterisk 1.8.23.0 from the repo's late night. every with asterisk is fine but AST does not show any of the ext. only thing I see on restart is this ERROR[1266]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL |
15:24.56 | Captain_Proton | could this be causing it |
15:26.19 | *** join/#asterisk cneb3000 (~cneb3000@2.221.241.85) |
15:28.04 | igcewieling | I recommend talking to the person who created the RPM |
15:30.59 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
15:45.06 | Captain_Proton | igcewieling: where do I find that out? it is just asteriskNow |
15:45.46 | *** join/#asterisk War_Bear (~War_Bear@warbear.co.uk) |
15:46.06 | igcewieling | Captain_Proton: no idea. Maybe try a channel relevant the software you are using, like #AsteriskNow |
15:46.34 | igcewieling | ~asterisknow |
15:46.35 | infobot | i guess asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
15:47.36 | *** join/#asterisk Rumbles (~Rumbles@77.107.183.253) |
15:47.59 | Captain_Proton | thanks |
15:48.21 | *** part/#asterisk Captain_Proton (~quassel@173.162.32.1) |
15:50.23 | *** join/#asterisk ybeddyj (~ybeddyj@72.252.160.99) |
15:51.14 | vlad_starkov | Question: Anyone know how to use ParkAndAnnounce() application? |
15:51.53 | *** join/#asterisk War_Bear (~War_Bear@warbear.co.uk) |
15:53.17 | [TK]D-Fender | vlad_starkov: https://wiki.asterisk.org/wiki/display/AST/Application_ParkAndAnnounce |
15:53.33 | [TK]D-Fender | vlad_starkov: Instructions seem pretty clear... what part of that are you having trouble with? |
15:54.14 | vlad_starkov | [TK]D-Fender: "dial" and "return_context" are not clear for me |
15:54.29 | [TK]D-Fender | vlad_starkov: dial - The app_dial style resource to call to make the announcement. Console/dsp calls the console. |
15:54.40 | [TK]D-Fender | vlad_starkov: ei the same thing you'd give DIAL() |
15:55.38 | [TK]D-Fender | vlad_starkov: return_context - The goto-style label to jump the call back into after timeout. Default priority+1. <-- just like it says.. when the app EXITS... here's where it goes.. unless you tell it to go to ANOTHER context |
16:00.09 | vlad_starkov | [TK]D-Fender: how to understand "dial"? Is it the location where the parked call will go to on timeout ? |
16:00.41 | [TK]D-Fender | vlad_starkov: No... it what it will DIAL... to ANNOUNCE to. |
16:01.11 | [TK]D-Fender | vlad_starkov: You are announcing to SOMETHING ELSE... not the transferer |
16:01.20 | vlad_starkov | [TK]D-Fender: It looks different than builtin parking feature |
16:01.20 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
16:01.47 | *** join/#asterisk [404] (~404]@12.179.117.114) |
16:01.53 | [TK]D-Fender | vlad_starkov: Park() tells the lot to the person doing the attended transfer to parking. Parkandannounce .... DIALS what you tell it to and tells THAT the lot instead |
16:01.56 | vlad_starkov | [TK]D-Fender: why I'd need to announce to someone else? |
16:02.10 | [TK]D-Fender | vlad_starkov: Becausew you want to? |
16:02.29 | vlad_starkov | [TK]D-Fender: Oh, probably I need another app – Park() |
16:02.34 | [TK]D-Fender | vlad_starkov: can you not imagine any possible use for this? |
16:03.48 | vlad_starkov | [TK]D-Fender: hmm, Park() – Park yourself. <-- this makes me confused |
16:04.07 | jmetro | make the caller park itself to wait on hold? |
16:04.09 | [TK]D-Fender | vlad_starkov: I already told you how you use it... |
16:04.45 | [TK]D-Fender | vlad_starkov: ATTENDED TRANFER... YOU get the lot # read to you ... then you FINISH the attended transfer |
16:04.59 | vlad_starkov | [TK]D-Fender: what I need to do is to transfer the callee to *100 and to park him and to get parked slot number said back to me |
16:05.19 | jmetro | park by default will tell you the slot |
16:05.33 | [TK]D-Fender | Which is what I said. Twice |
16:06.35 | vlad_starkov | jmetro: I already tried default park functionality, but it turned out that I need to create my own as the default one does not provide the required functionality |
16:06.52 | [TK]D-Fender | vlad_starkov: It does everthing you've described so far... |
16:06.58 | [TK]D-Fender | vlad_starkov: So what are you not telling us? |
16:08.39 | vlad_starkov | [TK]D-Fender: The default park functionality works just great! One extra thing I need is to retrieve callerid information for explicit parked call. |
16:09.02 | [TK]D-Fender | Clarify "explicit" |
16:09.04 | vlad_starkov | *before that call is being picked up |
16:09.24 | [TK]D-Fender | Pick it up where? How? |
16:11.56 | vlad_starkov | Say we have parked call on slot 704. We have special [parkedcalls_pickup] context that makes some logic with DB and then Dial(Local/704@parkedcalls,60,THK). So I need to retrieve the 704's callerid to save it in DB before I actually pick it up. |
16:12.33 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
16:12.51 | [TK]D-Fender | Then go do it |
16:13.04 | jmetro | i told him that phones can do it |
16:13.10 | jmetro | my aastra has an xmlapp on it that does it. |
16:13.12 | [TK]D-Fender | You can do whatever you want to scan your open channels and see the lot it's in and grab the callerID from that channel |
16:13.19 | [TK]D-Fender | This has nothing to do with actual "parking" |
16:13.49 | [TK]D-Fender | And you should know what you're capable of doing via dialplan apps to access vars from other channels, etc. This may require using AMI to get that info |
16:13.51 | vlad_starkov | [TK]D-Fender: sounds interesting, what should be my strategy here? |
16:14.15 | [TK]D-Fender | look at the dialplan apps & functions that can access other channels info. |
16:14.24 | [TK]D-Fender | and then if you don't see it there.. AMI <---------- |
16:14.25 | vlad_starkov | [TK]D-Fender: It is required to not use AMI. |
16:14.45 | [TK]D-Fender | That may be your only choice |
16:15.00 | [TK]D-Fender | You expect to pull some value out from ANOTHER channel |
16:15.05 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
16:15.13 | [TK]D-Fender | one channel does not normall get to spy on others that way |
16:15.17 | [TK]D-Fender | that is what AMI is for |
16:16.27 | vlad_starkov | [TK]D-Fender: I know that AMI could be solution, as it emits Park events. |
16:16.28 | *** join/#asterisk jsjc (~Adium@238.Red-79-146-120.dynamicIP.rima-tde.net) |
16:17.29 | Greenlight | Can you store the CallerID, *before* parking the call ? |
16:17.51 | vlad_starkov | [TK]D-Fender: But it is required for me to not use AMI in this task, so that I decided to build my own call parking logic |
16:18.03 | *** join/#asterisk Rumbles (~Rumbles@77.107.183.253) |
16:18.12 | vlad_starkov | Greenlight: Yes |
16:18.35 | Greenlight | vlad_starkov: So, do that then. |
16:18.50 | [TK]D-Fender | vlad_starkov: Well if you want to handcuff yourself... best of luck with that. You should already see what little is offered. If the callerID of that channel isn't part of it (and I seriously doubt there is another other way) ... best of luck to you. |
16:19.24 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
16:26.19 | jmetro | regular parking definitely allows you to pull caller ID off the channel/lot |
16:26.37 | jmetro | operator panels dont re-write parking |
16:27.04 | [TK]D-Fender | Operator panels.... USE AMI |
16:27.19 | [TK]D-Fender | </big_print> |
16:28.14 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
16:28.42 | jmetro | ^^^ |
16:28.44 | jmetro | =) |
16:28.56 | jmetro | [that was my point] |
16:29.15 | vlad_starkov | Greenlight: [TK]D-Fender: Ok, I tried it with Park() and it works. And now the question is how to get the parking slot that the call was placed on? |
16:30.02 | Greenlight | You've set yourself the challenge of not using AMI... |
16:30.17 | [TK]D-Fender | vlad_starkov: Go look at the channel and any extra AMI events thrown off |
16:30.18 | vlad_starkov | There is the ability to explicitly set parking slot |
16:30.54 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
16:31.24 | vlad_starkov | Greenlight: that's because this server is standalone (no co-wrokers like adhearsion or so are allowed). |
16:32.22 | WIMPy | And how is that related to *YOU* not using AMI? |
16:32.27 | Greenlight | I don't understand the connectionm |
16:32.31 | Greenlight | Indeed |
16:33.08 | vlad_starkov | Probably I misunderstand something, How is it possible to listen to AMI inside dialplan? |
16:33.12 | Greenlight | It's like coming in here and saying you can get SIP working, and the box isn't allowed to be connected to the network, and people must help you find a way around it. |
16:33.44 | WIMPy | Do you know what AMI is? |
16:33.59 | [TK]D-Fender | Clearly not properly |
16:34.01 | vlad_starkov | WIMPy: manager interface, that I can connect remotly |
16:34.13 | WIMPy | "remote" |
16:34.21 | vlad_starkov | WIMPy: sorry |
16:34.53 | vlad_starkov | I'm not native english speaker :-) |
16:35.34 | WIMPy | Well, it's IP so it is remote in a sense. But I've never used it on anything but 127.0.0.1. |
16:35.35 | Greenlight | So, you don't have a need for a remote user to manage a parked call ? |
16:35.58 | [TK]D-Fender | Nobody reads anymore... |
16:36.00 | vlad_starkov | Greenlight: nope |
16:36.03 | [TK]D-Fender | heads off to lunch |
16:36.54 | Greenlight | I thought you were trying to get the details of a parked call? |
16:37.32 | vlad_starkov | I have the strong terms to implement that logic only using asterisk facilities. |
16:37.56 | WIMPy | AMI is aterisks strongest tool. |
16:38.06 | Greenlight | With AMI you can almost do *anything* |
16:38.53 | vlad_starkov | Am I right thinking that to use AMI I need some additional app(daemon) running on the same server or on remote server? |
16:39.12 | WIMPy | The one you write. |
16:39.40 | igcewieling | AMI or dialplan, you're going to have to write code |
16:40.21 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
16:40.50 | vlad_starkov | WIMPy: that is restrictions. I only allowed to use asterisk facilities to implement it. No additional daemons and open ports. That is what I'm trying to say. |
16:41.11 | Greenlight | Your restriction, not asterisk's or ours |
16:41.35 | WIMPy | So you're not allowed to use "asterisk -r", either? |
16:41.43 | vlad_starkov | Greenlight: So I just asked is it possible to implement it without using AMI. |
16:41.49 | Greenlight | I wonder if you're allowed to have 5060 open? |
16:41.52 | vlad_starkov | WIMPy: that's allowed. |
16:42.15 | WIMPy | And why is one "remote" connection allowed and not the other? |
16:42.19 | igcewieling | vlad_starkov: AMI can be limited to 127.0.0.1 so you don't need open ports accessable from the network to run AMI. |
16:42.25 | WIMPy | I hope you get paid by the hour, BTW. |
16:42.25 | Greenlight | AMI *is* an asterisk facility |
16:42.46 | WIMPy | That's what I said. |
16:42.46 | Greenlight | Ok, what *exactly* is it you're trying to do ? |
16:43.40 | vlad_starkov | Greenlight: the goal is to get parked call's callerid just before the user pick that call up. |
16:44.03 | WIMPy | And what are you trying to do with that caller ID? |
16:44.16 | Greenlight | *before* the user picks up? |
16:44.30 | Greenlight | How is the user "picking up" the call ? |
16:45.16 | vlad_starkov | Greenlight: yes, just before Dial(Local/${EXTEN}@parkedcalls,60,THK) will be executed. |
16:45.45 | vlad_starkov | All the parked calls live in [parkedcalls] context. |
16:45.46 | Greenlight | ${EXTEN} being the parking slot ? |
16:45.53 | vlad_starkov | Correct |
16:45.57 | Greenlight | So, use a global variable |
16:46.01 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
16:46.04 | Greenlight | Have one for each parking slot |
16:46.06 | WIMPy | What do you plan to do with that caller ID? |
16:48.03 | Greenlight | Only thing I could think would be Set(CONNECTEDLINE(number)=.... |
16:49.01 | vlad_starkov | WIMPy: before Dial(Local/${EXTEN}@parkedcalls,60,THK) there is a sub that launches call recording, so I need to place the callerid of parked call in the recorded audio file name, like "call_from_102_to_1234567_(picked_up_from_708).mp3" |
16:51.01 | vlad_starkov | Ahead of you question, I need this special sub as I make stereo call recording (caller on left and callee on the right). |
16:51.11 | Greenlight | Way I see it you have 3 options. 1. Use AMI. 2. Use a global variable. 3. Use the AstDB or another DB/AGI |
16:51.48 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
16:51.52 | vlad_starkov | Greenlight: I chosen AstDB as a storage of parked calls information like callerid. |
16:52.05 | Greenlight | Ok... so what's the issue ? |
16:52.39 | Greenlight | Don't you have access to the parking slot number ? |
16:52.41 | danfromuk | Hi, ive set Events: on with the AMI but I dont see when calls are placed on Hold. Is there any way to get that output to the AMI? |
16:53.10 | Greenlight | danfromuk: "hold" is an odd one, sometimes it's just implemented by the endpoint |
16:53.35 | danfromuk | Greenlight: I thought that but I can hear the asterisk onhold music. |
16:53.50 | Greenlight | If that's the case you should see some events for it then |
16:53.57 | jmetro | you should see it call the onhold music |
16:54.03 | jmetro | started onhold music class "company" |
16:54.19 | *** join/#asterisk navaismo (~navaismo@189.191.233.140) |
16:54.23 | Greenlight | I *think* there are also "call" events that can be enabled |
16:54.27 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
16:54.27 | vlad_starkov | Now, the issue is how to get parked slot number that the call has been parked on? I implement my own logic to park the call using Park() application. It parks calls well but if it return PARKEDAT? |
16:54.28 | Greenlight | If memory serves |
16:55.09 | danfromuk | Greenlight: you are correct. Not sure why it wasnt showing on the AMI, but it is now. Weird. |
16:57.28 | *** join/#asterisk peetaur2 (~peter@x2f12083.dyn.telefonica.de) |
16:57.37 | Greenlight | vlad_starkov: You may need to manage the parking slot number yourself if you can't get it explicitly |
16:57.49 | Greenlight | Perhaps with some useage of the GROUP functions |
16:58.55 | vlad_starkov | Greenlight: That makes sense |
16:59.22 | Greenlight | I've always used AMI to provide this sort of functionality, so can't say that I'm too sure what gets set |
16:59.44 | WIMPy | Greenlight: Connectedline should happen automagically, I think. |
17:00.18 | Greenlight | If that's the case, why not wait till *after* the calls connected and run the sub then |
17:00.25 | Greenlight | And grab the number out of CONNECTEDLINE |
17:00.59 | WIMPy | That requires a current Asterisk version. |
17:01.04 | vlad_starkov | Greenlight: Probably in the future versions of asterisk it would make sense to add PARKEDAT variable to the Park() as it is in ParkAndAnnounce(). |
17:01.20 | Greenlight | vlad_starkov: Or, you could juse use ParkAndAnnounce |
17:01.23 | jmetro | ^ |
17:01.27 | jmetro | typed it before i could |
17:01.33 | danfromuk | Greenlight: it appears that the AMI packets for OnHold are the same as UnHold. Any idea how I can tell the difference? |
17:01.53 | Greenlight | Don't they stay "on" and "off" or something ? |
17:01.56 | Greenlight | @ danfromuk |
17:02.02 | WIMPy | danfromuk: IIRC thera was a change to these events recently. |
17:02.25 | danfromuk | Greenlight: There is a Musiconhold = yes and no. But is there always music? |
17:02.27 | Greenlight | vlad_starkov: Just looked at the docs - ParkAndAnnounce() would do exactly what you require |
17:02.27 | vlad_starkov | Greenlight: I started from ParkAndAnnounce() but as I figure out later this app is not exactly what I need, as it plays announcement to someone else. |
17:02.35 | WIMPy | But I can't remember having had difficulties to distinguish them. |
17:03.12 | Greenlight | vlad_starkov: Doesn't matter about the announcement - just just need that "hook" to set the PARKED EXTENSION number to AstDB with the CALLERID |
17:03.31 | danfromuk | WIMPy: when you say recently, was the change applied to version 1.8? |
17:03.37 | Greenlight | As long as you can execute some dialplan and have access to the CALLERID *and* the PARKED EXTENSION, you're sorted |
17:03.59 | WIMPy | danfromuk: No, much later. |
17:04.12 | danfromuk | Ok, I'm stuck on 1.8 at the moment. |
17:04.21 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/AMI+1.1+Changes <-- perhaps those |
17:04.25 | WIMPy | 1.8 is the oldest version I want to think of. |
17:04.34 | danfromuk | Ok, if thats not possible, is there a channel variable that contains the hold status? |
17:04.39 | Greenlight | Specificaly "For hold, there's a "Status: On" header, for unhold, status is off" |
17:04.59 | Greenlight | So, looks like your version lacks those. That's annoying. |
17:05.06 | WIMPy | You should get a channelstate, I think. |
17:05.32 | jmetro | vlad_starkov: parkandannounce plays audio and is suppressable |
17:05.46 | danfromuk | Ok, I'll try to find it. Or apply to have asterisk updated but not sure how much works involved in that yet. |
17:05.54 | jmetro | vlad_starkov: plays audio to the person putting the other person on park, suppress it by making a local channel do the park |
17:05.56 | Greenlight | Don't think there's an "hold" channelstate |
17:07.04 | WIMPy | No, right, I only get a hold event. |
17:07.43 | Greenlight | danfromuk: You *may* be able to patch your version by checking out the differences from current |
17:07.53 | Greenlight | If you've some major issue preventing upgrade |
17:08.12 | vlad_starkov | jmetro: ParkAndAnnounce(PARKED,300,dial,return_context) <-- 1. what should be "dial"? 2. what extension will be called in "return_context"? |
17:08.16 | Greenlight | I wouldn't imagine there's anything crazy going on, other than an additional line for the event |
17:08.30 | WIMPy | But that changes file says it changed from two different events to one with a status cariable. Both should do what you need. |
17:08.32 | vlad_starkov | Oh, "return_context - The goto-style label to jump the call back into after timeout." |
17:08.48 | jmetro | vlad_starkov: yep, google it |
17:09.44 | WIMPy | Errm |
17:10.25 | jmetro | also you can actually edit out the announce in the source code. Lol |
17:10.33 | WIMPy | Do you have to supply a supported version number to get the new format? |
17:11.18 | WIMPy | Or change the channels language to one that only contains empty files. |
17:11.42 | jmetro | that might be messy. |
17:12.30 | vlad_starkov | jmetro: So far the best solution I found is just to put Console/dsp as "dial" argument. |
17:13.06 | [TK]D-Fender | vlad_starkov: And what is that doing for you? |
17:13.29 | paulc | Anyone used custom device states and Cisco SPA508G phones with BLF+SD to give indication of active/inactive features? I've got it working but have a question about the subscribe target versus speed dial destination (it doesn't seem to work if they're not the same?) |
17:14.36 | vlad_starkov | [TK]D-Fender: ParkAndAnnounce() requires to set "dial" attribute, I don't need it, but as in mandatory it should be stubbed with something. |
17:14.52 | Greenlight | DO make a dummy extension that does nothing but set your data that you need. |
17:15.16 | Greenlight | Presumable ${PARKEDAT} is set. |
17:15.26 | vlad_starkov | Greenlight: Hmm, makes sense |
17:16.08 | [TK]D-Fender | vlad_starkov: If you aren't using the Dial .... THEN YOU ARE USING THE WRONG APP |
17:16.20 | [TK]D-Fender | vlad_starkov: The entire point of that app is TO ANNOUNCE IT. |
17:16.21 | Greenlight | [TK |
17:16.30 | jmetro | ParkAndAnnounce is the right app because it sets variables |
17:16.33 | Greenlight | [TK]D-Fender: He needs to set some stuff in the AstDB |
17:16.43 | Greenlight | Park doesn't seem to allow that to be done. |
17:16.45 | [TK]D-Fender | vlad_starkov: The only other thing to pass is the failover exten which Park() has options for anyway |
17:16.51 | Greenlight | Unless I'm mistaken. |
17:17.16 | Greenlight | Eg, he needs store the association between the parking slot number AND the CallerID. |
17:17.17 | [TK]D-Fender | Greenlight: And where is that happening? |
17:17.37 | Greenlight | Well it's a bit of a hack, but the plan was to set it inside Dial |
17:17.43 | Greenlight | Of ParkAndAnnounce |
17:17.55 | Greenlight | And make use of ${PARKEDAT} which gets set |
17:18.12 | Greenlight | So, he'll need Local/test@setmyparkedstuff |
17:18.30 | Greenlight | And inside that context he'll (in theory) have BOTH CALLERID and PARKEDAT |
17:18.44 | Greenlight | So he can then set those in the AstDB |
17:18.52 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
17:19.07 | vlad_starkov | Greenlight: that's correct |
17:19.07 | Ice_Strike | GSM Gateway - Is it possible to change IMEI? |
17:19.15 | Greenlight | Ice_Strike; No. |
17:19.23 | Greenlight | That's illgal prison time stuff iirc |
17:19.38 | Greenlight | They seriously frown on it |
17:19.41 | WIMPy | Ice_Strike: Check teh gateways software. |
17:19.43 | vlad_starkov | Ice_Strike: If I'm not mistaken, GoIP GSM gateway allows do some stuff. |
17:20.13 | Greenlight | I was told it was uber illegal... |
17:20.18 | Ice_Strike | Is it? |
17:20.26 | Greenlight | Unless I'm thinking of something else |
17:20.37 | WIMPy | AFAIk noone cares about the IMEI. |
17:20.49 | Greenlight | Isn't that the unique identified for the SIM card ? |
17:20.50 | Ice_Strike | If gateway allows IMEI change - how can it be done via asterisk? |
17:21.01 | WIMPy | No, that's IMSI. |
17:21.05 | Greenlight | Ahh... |
17:21.13 | Greenlight | Bloody acronyms |
17:21.23 | WIMPy | Quite definitely not. How could Asterisk do it? |
17:21.39 | Ice_Strike | We can change CallerID |
17:21.44 | WIMPy | [S]ubscriber / [E]quipment |
17:22.17 | Greenlight | ^^ |
17:22.27 | WIMPy | Use something else that can do it as gateway. |
17:22.42 | Ice_Strike | What do you mean, such as? |
17:22.56 | WIMPy | osmocom |
17:23.15 | WIMPy | How do you change caller ID? |
17:23.22 | Ice_Strike | Ah didnt know that |
17:23.30 | Ice_Strike | WIMPy I don't know if it can? |
17:23.30 | WIMPy | AFAIK that's not even possible to send. |
17:23.43 | Ice_Strike | I know we can via VOIP SIP. |
17:24.00 | Greenlight | You trying to send differnt callerid's using a GSM gateway ? |
17:24.11 | WIMPy | Yes, ITSPs and caller ID is an interesting topic. |
17:24.17 | Ice_Strike | Eg: exten => _X.,1,Set(CALLERID(num)=xxx) |
17:24.30 | Ice_Strike | Greenlight Is it possible? |
17:24.32 | Greenlight | We could never get that to work with GSM gateways |
17:24.37 | WIMPy | Greenlight: Unless I'm missing something, definitely no. |
17:24.40 | Greenlight | Was one of the reasons we ditched them |
17:24.44 | WIMPy | AFAIK that's not even possible to send. |
17:24.58 | Greenlight | That's the conclusion I came to at the time. |
17:25.11 | Greenlight | Was glad to see the back of them tbh |
17:25.41 | Greenlight | Scripts to keep track of inclusive minutes usage per SIM and whatnot. |
17:25.47 | Ice_Strike | Somone manage to do but he wont tell me how.. he want £££££ lol |
17:25.50 | Greenlight | Horrible tech |
17:26.14 | Greenlight | Ice_Strike: Take that with a grain of salt... |
17:26.41 | Ice_Strike | He said change the IMEI every call in order to change the caller ID. |
17:26.43 | WIMPy | Well, it is easy. Just hack someone elses SIM and use his account. But otherwise??? |
17:26.44 | Ice_Strike | That it. |
17:26.55 | WIMPy | That makes no sense. |
17:27.09 | Greenlight | But you'd be using SOMEONE ELSES'S SIM |
17:27.14 | Greenlight | As far as the network is concerned |
17:27.20 | Greenlight | Thta's um.... |
17:27.22 | WIMPy | Caller IDs are not set depending on the phone used. |
17:27.27 | vlad_starkov | Greenlight: One more thing that is not clear for me is how to set parking_lot_name for ParkAndAnnounce() ? |
17:27.29 | Ice_Strike | Actually. |
17:27.45 | Greenlight | He does mean IMSI... |
17:27.51 | WIMPy | You have to change IMSI. |
17:27.54 | WIMPy | Yes |
17:28.06 | Greenlight | Ice_Strike: Seriously... that's proper dodgy |
17:28.15 | Ice_Strike | Is it? |
17:28.18 | Greenlight | vlad_starkov: Use the default, no ? |
17:28.32 | Greenlight | Ice_Strike: It scares me that you need to ask.. |
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17:28.43 | WIMPy | Well, change IMSI usually means change to one of the many SIMs you have inserted. |
17:28.58 | vlad_starkov | Greenlight: there is multiple parking lots in the system, it needs to set parking lot explicitly. |
17:29.06 | Greenlight | vlad_starkov: So, choose one. |
17:29.10 | vlad_starkov | Greenlight: the documentation is not clear on it. |
17:29.39 | Greenlight | If the SIM's inserted in the system, then you can choose to just use that SIM... |
17:29.47 | Ice_Strike | Ohh right. |
17:29.49 | Greenlight | Why bother with IMSI's ... |
17:29.53 | Ice_Strike | True. |
17:29.55 | WIMPy | Yes, or in other terms change IMSI. |
17:29.55 | vlad_starkov | Greenlight: You mean I should set CHANNEL(parkinglot)? |
17:30.14 | Greenlight | I *suppose* if you have 1000 SIM's and only 10 slots in the GSM gateway, you could do it. |
17:30.30 | WIMPy | You can also change IMSI by physically removing the SIM and insert another. |
17:30.30 | Greenlight | BUt I was always told that it was a very dodgy thing to play around with |
17:30.47 | Ice_Strike | Why GSM gateway exist? |
17:31.03 | Greenlight | Used to be a cost thing for a lot of companies |
17:31.05 | WIMPy | To make cheaper calls? |
17:31.15 | Greenlight | Nowadays SIP is a lot cheaper and more accessible |
17:31.32 | vlad_starkov | Greenlight: CHANNEL(parkinglot) made the trick |
17:31.37 | Greenlight | Back 10 years ago, mobiles calls were a lot more expensive, and GSM gateways gave a nice alternative |
17:31.45 | WIMPy | So you are one of the guys who say SIP is cheaper? |
17:31.45 | Greenlight | vlad_starkov: Good :) |
17:31.50 | Ice_Strike | WIMPy no, I meant Greenlight is implying GSM gateway is illegal |
17:31.58 | Ice_Strike | or maybe missunderstood him |
17:32.03 | WIMPy | what? |
17:32.12 | Greenlight | Ice_Strike: No, GSM gateway isn't. Altering your IMSI is. |
17:32.16 | Ice_Strike | Ohhh |
17:32.34 | WIMPy | He, everyone does so from time to time. |
17:32.38 | Greenlight | WIMPy: Your one of the haters when we sell SIP on price ? :P |
17:32.51 | vlad_starkov | Greenlight: [TK]D-Fender: Thank you guys and everyone for your help! |
17:32.53 | WIMPy | But it's propper illecal to use someone elses IMSI, off course. |
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17:33.09 | Greenlight | That's the point I was making. |
17:33.12 | Ice_Strike | I am no plan to change IMSI |
17:33.18 | Ice_Strike | That is not my intention |
17:33.20 | WIMPy | I haven't seen cheap SIP, yet. |
17:33.51 | Greenlight | WIMPy: Cheap is relative. I'm comparing to similar volume over traditional PSTN networks. |
17:33.53 | vlad_starkov | Goodbye! |
17:33.55 | vlad_starkov | ) |
17:34.05 | Ice_Strike | For personal voip - i use voip buster |
17:34.06 | WIMPy | Greenlight: So do I. |
17:34.09 | Ice_Strike | much cheaper than skype! |
17:34.27 | Greenlight | WIMPy: And you find, say ISDN30, cheaper than SIP ? |
17:34.41 | WIMPy | Depends on the number of channels you need. |
17:34.49 | Ice_Strike | Greenlight By the way Lease Line installed other day :P |
17:34.59 | WIMPy | But a PRI is much less than the same amount of IP bandwidth. |
17:35.04 | Greenlight | Ice_Strike: Yay :) |
17:35.05 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
17:35.17 | Ice_Strike | Only 10mbit upload and 10mbit download though |
17:35.21 | Ice_Strike | enough for asterisk use only. |
17:35.26 | Greenlight | That's plenty for your needs... |
17:35.29 | Ice_Strike | Yep |
17:35.37 | Ice_Strike | Shame Fibre not around this area :/ |
17:35.43 | Ice_Strike | Would be much cheaper if I go with that |
17:35.45 | Greenlight | WIMPy: Maybe different pricing over here in UK then... |
17:36.08 | Greenlight | Ice_Strike: That's only pretend "fibre", and not suitable for your needs :) |
17:36.38 | Greenlight | Aint you like 50+ seat call centre ? |
17:37.48 | Greenlight | Right am off - laters |
17:44.51 | danfromuk | Just looking at the UPGRADE documentation. Am I correct in saying that an upgrade from 1.8 to 11 shouldn't be that difficult? I can't see anything regarding the dialplans that would need changing. |
17:45.26 | WIMPy | Probably. |
17:45.43 | danfromuk | Ok. I'll try it on a test server and see what the results are like. |
17:45.44 | danfromuk | Thanks |
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19:19.49 | danfromuk | Hi, I'm trying to upgrade from 1.8 to 11 and seeing the following when trying to Dial. "No RTP engine was found. Do you have one loaded?" |
19:20.04 | danfromuk | I've checked the build and the correct options seem to be selected. Any ideas what could be causing it? |
19:20.27 | WIMPy | Is it really built? It needs uuid now. |
19:20.34 | WIMPy | (since 11.5) |
19:20.46 | WIMPy | res_asterisk_rtp |
19:21.12 | danfromuk | I've installed uuid and its marked as [*] in menuselect |
19:21.27 | WIMPy | No autoload? |
19:22.58 | danfromuk | strange. just checked the modules directory and that module is missing. |
19:23.27 | WIMPy | Did you install to the same locations? |
19:23.46 | danfromuk | just used the default locations |
19:25.36 | danfromuk | ah. the /usr/lib folder is now /usr/lib64 |
19:26.04 | WIMPy | o.O |
19:27.18 | WIMPy | still have them in /usr/lib/, even on the 64bit OS. |
19:28.11 | danfromuk | except i was struggling to build and changed the path to lib64 as an attempt to build |
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19:48.04 | igcewieling | Wow, porting my cell number to Vitelity was about as painless as it can get |
19:48.57 | igcewieling | danfromuk: if you are just missing libasteriskssl or whatever when you try loading res_rtp_asterisk then all you need to is run ldconfig so asterisk will pick up the newly installed librart |
19:49.01 | igcewieling | library |
19:49.40 | jmetro | igcewieling: did you just say "here's my number, port it for me" and boom it was there and configured? |
19:49.53 | jmetro | plus a plate of cookies? |
19:50.16 | igcewieling | jmetro: I had to fill out a form and upload a signature graphc. |
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19:53.02 | igcewieling | *sigh* Upgraded to Asterisk 11 and FreePBX 2.11 and now our FreePBX backups are running once per min. |
19:53.29 | jmetro | realtime backups! |
19:57.37 | Vann | :D |
19:57.47 | Katty | hello my asterisk does not work at all how to fix plz? |
19:57.52 | Katty | is urgent, srs answers only thx |
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19:59.19 | jmetro | katty: your R&D division in Black Mesa should be able to provide resolution, |
19:59.47 | Katty | books trip to Arizona |
19:59.55 | Katty | grabs sunblock and jets! |
20:00.12 | jmetro | Dont forget to take a swing by Anomal..i mean Asterisk Materials. |
20:00.22 | igcewieling | jmetro: realtime running out of diskspace and breaking Asterisk |
20:00.47 | jmetro | igcewieling: that needs to be a quote. |
20:02.15 | igcewieling | "When Good Backups Go Bad: Next on Court TV!" |
20:02.17 | jmetro | On a side note "Asterisk Realtime" is actually pretty good. |
20:03.25 | igcewieling | Realtime is like that girl/guy you see from a distance which looks really hot, but the closer you get the less hot they look. Realtime is like that. |
20:04.36 | jmetro | but then you get to know them and theyre into all the same things as you and they become way hotter than original. |
20:05.14 | igcewieling | jmetro: That is where the analogy breaks down. |
20:06.02 | jmetro | no :< |
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20:14.42 | danfromuk | Issuing CORE SET VERBOSE 0 from the AMI doesn't seem to have an effect on the CLI in asterisk 11. Does anyone know whats changed? |
20:15.07 | WIMPy | It is local to the shell now. |
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20:15.40 | danfromuk | How can I affect the CLI's verbosity level from the AMI so that when the AMI issues a DIALPLAN reload, it doesnt end up filling the CLI output? |
20:16.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.17) |
20:17.11 | WIMPy | Interesting idea. Probably doesn't work any more. |
20:17.46 | antonioX | Hi all |
20:17.53 | antonioX | I've a 2 fxs ports linksys ATA connected to an *. If someone calls from the PSTN and press 1, the call should be bridged to port 1 on the ATA, but I got a "302 Moved Temporarily" from the ATA: http://pastebin.com/raw.php?i=L06CK6jL |
20:17.59 | danfromuk | Hmmm. thats a problem. I record the CLI verbose output for later debugging if required. But I dont need to record all the output from dialplan reloads. |
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20:18.14 | antonioX | Calls from PSTN -> * -> port 2 works ok. And calls from port 2 phone to port 1 works ok. The only difference on the ATA config I saw between port 1 and 2 is the username/password and the port (5060 on port 1, 5061 on 2). |
20:18.19 | antonioX | extensions.conf: http://pastebin.com/raw.php?i=R0cyBLJn |
20:18.20 | antonioX | macros.conf: http://pastebin.com/raw.php?i=qkVHY8Z3 |
20:18.29 | danfromuk | I used to set the verbose to 0 before doing a reload which worked fine. |
20:18.33 | danfromuk | in 1.8 |
20:18.55 | antonioX | What should I do to debug this? |
20:19.07 | WIMPy | Now you can have multiple shells open simultaneousely wioth different verbose levels. |
20:19.25 | danfromuk | verbose |
20:19.27 | danfromuk | oops |
20:19.28 | danfromuk | sorry |
20:19.39 | WIMPy | Or debug |
20:19.42 | Katty | i'll sorry your verbose in a minute. |
20:19.50 | Katty | (what?) |
20:19.53 | Cuzner | asterisk -vvvvvvvvvvvvvvvvvr or core set verbose 9000 |
20:20.04 | Cuzner | over 9000 even |
20:20.11 | jmetro | 9001 preferably |
20:20.23 | Katty | make in the olden days, we didn't have verbose 9001 |
20:20.25 | jmetro | [its impossible] |
20:20.33 | Katty | we walked up hill, both ways. |
20:20.37 | Katty | IN THE SNOW. |
20:22.06 | danfromuk | WIMPy: ok, looks like theres no way to resolve that. |
20:24.42 | danfromuk | WIMPy: any idea if its possible to change the verbose level of dialplan reloads so they are higher than originally designed? |
20:25.37 | Kobaz | sooooo umm |
20:25.41 | Kobaz | i have a fun question for you guys |
20:25.50 | jmetro | Ball-pit. |
20:26.08 | Kobaz | on many calls but not all of them... on this one system they get three beeps in a row, it sounds like a truck backing up |
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20:26.57 | danfromuk | When I had an issue like that, it turned out that the calls were being diverted through another PBX at the client's office. They were diverting calls but didn't tell me. |
20:27.25 | Kobaz | could that be like some sort of external call recording beeps or something |
20:27.55 | 44UAAB9Q3 | I am revising an issue from years ago. Doing an AllPage, in the past, when there were -many- sip endpoints, the all page would fail |
20:28.03 | 44UAAB9Q3 | due to the command being "too long". |
20:28.05 | danfromuk | Could be anything. I dont think asterisk plays beeps without showing it in the CLI |
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20:28.09 | Kobaz | yeah |
20:28.11 | 44UAAB9Q3 | is this still an issue? |
20:28.13 | Kobaz | danfromuk: that's what i was thinking |
20:28.24 | Kobaz | danfromuk: and i don't have *any* polycom programming to do anything like that |
20:28.47 | danfromuk | Is it only happening with incoming calls? |
20:28.52 | Kobaz | yeah |
20:28.54 | Kobaz | only inbound |
20:29.20 | danfromuk | Then I would say that the calls aren't coming in directly. They must be going through another system which is adding the beeps. |
20:30.29 | WIMPy | danfromuk: They already have multiple levels. You can always change the source... |
20:31.05 | danfromuk | WIMPy: never changed the source before. and if I upgrade again, it will need redoing. |
20:31.24 | danfromuk | plus, wouldnt know where to start looking. |
20:31.39 | WIMPy | Or you try to filter what you save. |
20:31.58 | danfromuk | i just use logger.conf to save everything thats output to the cli |
20:33.34 | WIMPy | Not an wasy one. |
20:34.02 | WIMPy | You could change the logger config instead of the verbosity. |
20:35.57 | danfromuk | Yeh, but it still needs to record other cli output for the call flows etc. Unless I can manipulate the logger from the AMI |
20:36.26 | danfromuk | Still a bit annoying trying to use the CLI and having the dialplan automatically reloading every so often. |
20:36.35 | WIMPy | I guess you would have to change the config file and "logger reload". |
20:36.53 | danfromuk | i think changing the source code is the simplest option. |
20:37.04 | WIMPy | Probably the safest. |
20:37.25 | danfromuk | Any idea which file contains the code for dialplan loading? |
20:38.20 | WIMPy | pbx_config would be my first guess. |
20:39.45 | file | pbx_config loads and parses extensions.conf, with it using defined APIs in pbx.c to manipulate the internal PBX core |
20:40.12 | Qwell | file: go get on a plane |
20:40.21 | file | I can not. |
20:40.31 | file | there is a maintenance delay of unknown length |
20:40.33 | Qwell | You should try |
20:40.35 | Qwell | eep |
20:40.51 | danfromuk | @file, thanks. It is easy to spot where the verbose level is specified for outputs that are generated when reloading the dialplan? |
20:41.08 | file | should be |
20:41.23 | 44UAAB9Q3 | is there a better way of doing all page than we were doing back in the 1.4 and 1.6 days with an allpage.agi ? |
20:41.31 | 44UAAB9Q3 | using 1.8 or 10/11 now? |
20:41.41 | danfromuk | @file. Ok, thanks |
20:42.11 | 44UAAB9Q3 | dunno how the hell my nick is "44UAAB9Q3" Geesh. should be "pigpen" |
20:42.28 | igcewieling | 44UAAB9Q3: Try a channel for your GUI? |
20:43.03 | 44UAAB9Q3 | lets see if I can clear this up. |
20:43.05 | *** part/#asterisk 44UAAB9Q3 (~mark@fw.seamans.cc) |
20:43.52 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
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20:44.08 | pigpen | better? |
20:44.15 | pigpen | Yes. back to my pigpen self. |
20:44.48 | pigpen | anyway, I see multicast rtp in 1.8, dunno how that works with the poly's. I know cyberdata has options for it. |
20:45.30 | igcewieling | pigpen: from my understanding the polycoms do the multicast all on their own, without involving Asterisk |
20:46.17 | pigpen | well, years ago, I ran across an issue, when all-paging more than 30 phones (ie: I was tring from 30 - 300 sip devices), that the command length was too long |
20:46.19 | pigpen | and it failed. |
20:46.35 | pigpen | so I had to chop them up into smaller batches and call the batches in a meetme request. |
20:46.38 | pigpen | odd, pita. |
20:46.58 | pigpen | I'll look into the multicast way, could be nice. |
20:50.39 | igcewieling | pigpen: read the polycom documentation before anything else. |
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20:56.37 | pigpen | years ago. |
20:56.46 | pigpen | surly it hasn't changed. ;-) |
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21:00.39 | jmetro | If i allow ulaw,g722 and the connection has ulaw,g722 which one does it choose |
21:02.31 | [TK]D-Fender | ulaw |
21:03.05 | jmetro | and if i reverse that? |
21:03.12 | jmetro | g722,ulaw on both |
21:04.18 | [TK]D-Fender | g722 |
21:04.39 | leifmadsen | jmetro: order matters |
21:04.44 | leifmadsen | first ordered is highest priority |
21:04.51 | leifmadsen | if supported, will use that |
21:05.00 | leifmadsen | e.g. first codec listed is preferred |
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21:05.24 | jmetro | right, now that i think about it |
21:05.39 | jmetro | makes more sense as a list rather than a pool |
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21:33.42 | guldgossen | hello :) |
21:34.06 | WIMPy | A golden what? |
21:34.40 | pigpen | egg |
21:34.44 | guldgossen | hmm..boy..or something :p |
21:34.54 | psanders | Any agi/ami dev in this channel? I need some help to validate the RC1 of my framework (http://astivetoolkit.org). |
21:35.01 | WIMPy | oh |
21:35.04 | WIMPy | ok |
21:35.42 | WIMPy | psanders: And what exactely is your question? |
21:35.52 | pigpen | hmm..I am kinda disappointed now. |
21:38.17 | psanders | WIMpy: I'm asking for people with experience with agi/ami willing to help testing this software. |
21:38.51 | WIMPy | psanders: Aren't that those people who don't need it? |
21:40.32 | psanders | WIMPy: Not sure I understand your question. |
21:40.36 | guldgossen | I have a question about asterisk |
21:41.01 | guldgossen | what does failed mean under disposition in the cdr log? |
21:41.09 | pigpen | guldgossen, just one? shit, I have been using/developing on it for years and I probably have 10 a day. |
21:41.22 | guldgossen | hehe |
21:41.37 | WIMPy | psanders: The stuff you did there seems only of interest to those who don;t have the knowledge to check for its sanity. |
21:41.46 | guldgossen | oh..i should say freepbx btw |
21:41.59 | guldgossen | freepbx/asterisk :) |
21:45.33 | psanders | WIMPy: You may be right... |
21:46.29 | guldgossen | anyone knows? |
21:46.58 | guldgossen | seems like I only get it when making calls using my vpn server |
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21:47.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
21:48.37 | WIMPy | Intereesting. I just got two calls at about the same time, but they didn't get to any phone before one of them gave up. |
21:51.58 | navaismo | guldgossen, failed means failed --captain obvious attack again- so cant contact the peer the trunk or so... |
21:57.02 | guldgossen | navaismo: ok..thanks |
21:57.23 | WIMPy | I have an issue with Asterisk answering a MESSAGE with "415 Unsupported Media Type" which then unfortunatly kills the call. Is there anything I can do to change that behaviour? |
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21:57.45 | WIMPy | I see there is soem MESSAGE related configuration, but I think that only for SIMPLE? |
22:00.49 | WIMPy | Or rather only for out-of-call messages, which are not my concern. |
22:02.45 | psanders | WIMPy: Googleit -> https://issues.asterisk.org/jira/browse/ASTERISK-15802 |
22:03.09 | WIMPy | I see that I'm not alone, But I haven;t found the answer, yet. |
22:03.29 | psanders | WIMPy: One of the 2 SIP endpoints does not have a G.729(or any other codec) license |
22:03.57 | WIMPy | Did you read my question? It's about MESSAGE messages. |
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22:05.44 | psanders | WIMPy: Yes. |
22:06.11 | WIMPy | So no CODECs involved here. |
22:06.40 | psanders | WIMPy: Ok... |
22:08.16 | WIMPy | Google gives me several identical questions, but the only extremely dirty suggestion so far was to tchange the error message in the source to "202 Accepted". |
22:10.11 | WIMPy | is constantly amazed at how unusable all this stuff is in practice :-( |
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22:21.28 | WIMPy | wonders if chan_gulp could help. |
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22:56.29 | WIMPy | Ok, patch is available. |
22:57.14 | WIMPy | wonders how much brainfuck it would be to use libpri from chan_sip. |
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23:03.48 | igcewieling | edges away from WIMPy and makes sure there is garlic on hand |
23:04.32 | WIMPy | I want a whole acre of garlic between me and SIP. |
23:04.42 | WIMPy | (at least) |
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