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| 09:00.18 | benngard | may i ask some questions regarding CONNECTEDLINE function? (i have done some digging in the code, some "almost working" example but need some help to change "almost working" to just "working") |
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| 14:20.52 | jmls | seanbright: latest changes for MuteMixMonitor applied, thanks for the comments |
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| 17:01.44 | niekvlessert | hello ppl |
| 17:01.51 | niekvlessert | i'm having trouble with sip in asterisk |
| 17:01.52 | Kobaz | we don't want any |
| 17:02.15 | Kobaz | we already have girlscout cookies |
| 17:02.17 | jsmith | niekvlessert: If it's a problem with the source code, discuss it here |
| 17:02.28 | jsmith | niekvlessert: If it's a configuration problem, ask in #asterisk |
| 17:02.29 | niekvlessert | yes it is. :) |
| 17:02.35 | niekvlessert | We use Aastra phones. Look at this ticket: |
| 17:02.35 | niekvlessert | http://www.freepbx.org/v2/ticket/2563 |
| 17:02.57 | niekvlessert | anyone who wants to earn some money? |
| 17:03.02 | niekvlessert | fix this for us :) |
| 17:03.25 | jsmith | isn't interested in the money, but will look at the ticket anyway |
| 17:03.43 | jsmith | niekvlessert: That's not "trouble", that's a feature |
| 17:03.49 | niekvlessert | true |
| 17:03.54 | niekvlessert | but it's trouble for me :) |
| 17:03.57 | jsmith | niekvlessert: Lucky for you, that feature has already been implemented in trunk, and will be available in 1.8 |
| 17:04.06 | niekvlessert | yeah, i heard |
| 17:04.19 | Kobaz | interesting |
| 17:04.23 | niekvlessert | but I need it very quickly |
| 17:04.30 | jsmith | niekvlessert: Then use trunk until 1.8 comes out |
| 17:04.46 | Kobaz | niekvlessert: #asterisk-biz |
| 17:04.58 | Kobaz | how much are you paying? |
| 17:05.15 | jsmith | niekvlessert: A need on your part doesn't constitute an emergency on our part. You can ask on the asterisk-biz mailing list to get someone to backport it, but it would be very difficult and probably quite expensive |
| 17:05.26 | jsmith | niekvlessert: Or, you can use trunk until 1.8 is released. |
| 17:05.30 | Kobaz | jsmith: haha, my business partner uses that phrase... it's great |
| 17:05.44 | niekvlessert | i agree it's not your problem :) |
| 17:05.50 | niekvlessert | i already joined asterisk-biz |
| 17:05.58 | niekvlessert | kobaz, u wanna talk overthere? |
| 17:06.10 | Kobaz | not in particular... just curious |
| 17:06.22 | niekvlessert | well ok. |
| 17:06.25 | Kobaz | i have 3904023948324 other things to work on |
| 17:06.38 | niekvlessert | what simpiflies things is that we only need it for sip |
| 17:06.39 | Kobaz | but if the money makes sense, i might be interested |
| 17:06.53 | Kobaz | if it's for like 100 bucks then it's not worth it |
| 17:07.34 | niekvlessert | well, i'd need an estimate on how much time it would take to fix |
| 17:07.51 | niekvlessert | i've came up with 3 scenario's where this is a problem |
| 17:07.53 | niekvlessert | if one is fixed |
| 17:08.11 | niekvlessert | we can probably fix the rest ourselves |
| 17:08.38 | Kobaz | what version are you running on |
| 17:08.47 | niekvlessert | the most important is the attended transfer thing |
| 17:08.58 | niekvlessert | 1.4.26.2 |
| 17:09.01 | Kobaz | ouch |
| 17:09.06 | Kobaz | that's a lot of back porting |
| 17:09.28 | niekvlessert | well that brings me to something else |
| 17:09.39 | jsmith | niekvlessert: You're probably talking at least forty hours or work to backport it to 1.4, at a very minimum |
| 17:09.40 | niekvlessert | how stable do you ppl think 1.6 is? |
| 17:09.52 | jsmith | niekvlessert: It's going to be *very* *very* difficult |
| 17:10.05 | niekvlessert | jsmith: u mean to fix it in sip only? |
| 17:10.05 | jsmith | niekvlessert: 1.6.0, 1.6.1, and 1.6.2 have all been very stable on my systems |
| 17:10.06 | Kobaz | 1.6 is good for most things... if you're doing lots of local channels it can be crashy |
| 17:10.16 | jsmith | niekvlessert: Yes, even if you (hackishly) only do it for SIP |
| 17:11.12 | Kobaz | i managed to crash 1.6.2 with odbc on several occasions |
| 17:11.13 | niekvlessert | what do u mean, local channels, when not using re-invite? |
| 17:11.23 | Kobaz | local channels as in the Local channel |
| 17:11.28 | Kobaz | Dial(Local/....) |
| 17:11.50 | niekvlessert | hmm ok |
| 17:11.53 | Kobaz | now i try not to do any database queries from inside asterisk, things seem much more stable |
| 17:12.07 | niekvlessert | well we use the database a LOT |
| 17:12.17 | niekvlessert | well not ture |
| 17:12.19 | niekvlessert | true |
| 17:12.25 | niekvlessert | we do agi, and then access db |
| 17:12.44 | Kobaz | odbc in 1.6.0 is pretty good, although it leaves dangling connections pretty often... but i haven't managed to track down the reason |
| 17:12.55 | Kobaz | whether it's asterisk or unixodbc i don't know |
| 17:12.58 | niekvlessert | jsmith: too bad!! :( how much work would you think it is in 1.6? |
| 17:13.18 | jsmith | niekvlessert: Less, but still significant |
| 17:13.40 | niekvlessert | well, why not use agi, and then db |
| 17:13.51 | Kobaz | yeah i do my db from agi now |
| 17:14.14 | niekvlessert | jsmith: that's too bad!! i rly need this :( |
| 17:14.28 | niekvlessert | can i do it with sipsak? |
| 17:14.34 | niekvlessert | do agi then sipsak |
| 17:14.56 | jsmith | niekvlessert: Not sure... Anything is possible... |
| 17:15.29 | jsmith | If it were me, I'd try using trunk on a development (non-production) machine, and see how things go |
| 17:15.44 | niekvlessert | yeah... maybe so |
| 17:15.44 | jsmith | It'll either work for you, or you'll find bugs that need to be addressed before 1.8 can be released |
| 17:15.47 | Kobaz | just make sure you do lots and lots of testing |
| 17:15.58 | jsmith | The earlier we get things working well, the earlier 1.8 gets released, and you get your needed feature |
| 17:16.03 | niekvlessert | but a LOT of ppl will be running on those systems... |
| 17:16.31 | niekvlessert | jsmith: agree |
| 17:17.02 | jsmith | niekvlessert: Hence the reason I said to test it will before putting on a production system |
| 17:17.10 | niekvlessert | agree :) |
| 17:17.15 | jsmith | niekvlessert: But you're probably further ahead in going that route, rather than wasting money on a backport |
| 17:17.37 | niekvlessert | i will tell the 3 scenario's, maybe you can answer if these are solved |
| 17:17.49 | Kobaz | scenarios |
| 17:18.03 | niekvlessert | remember, those are things users need, not me :) |
| 17:18.26 | Kobaz | puts on his grammar-nazi hat |
| 17:18.38 | niekvlessert | a dials b, b picks up, phone b will display username from a and number |
| 17:18.44 | niekvlessert | hey i'm dutch, bare with me :) |
| 17:18.59 | niekvlessert | problem: phone will only display number dialed |
| 17:19.07 | niekvlessert | phone A i mean |
| 17:19.12 | Kobaz | correct |
| 17:19.12 | jsmith | niekvlessert: That's supported in 1.4... it's called "Caller ID" |
| 17:19.37 | niekvlessert | not on phone A |
| 17:19.46 | Kobaz | there is also a new feature, i think it's in trunk... to update the presentation on phone A |
| 17:19.53 | Kobaz | i forgot what it's called |
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| 17:19.59 | jsmith | Oh, on phone A |
| 17:20.02 | kpfleming | connected line presentation |
| 17:20.03 | jsmith | Yes, that's supported |
| 17:20.08 | Kobaz | yeah that's the name |
| 17:20.20 | jsmith | waves to kpfleming and jeffg |
| 17:20.32 | kpfleming | are you enjoying the cold white fluffy stuff? |
| 17:20.37 | Kobaz | oh yes |
| 17:20.39 | niekvlessert | ok thanks |
| 17:20.40 | niekvlessert | nex |
| 17:20.41 | Kobaz | foot and a half so far |
| 17:20.42 | niekvlessert | next |
| 17:20.56 | niekvlessert | someone dials B |
| 17:20.57 | jsmith | kpfleming: 22" and counting! |
| 17:21.01 | kpfleming | wow |
| 17:21.04 | niekvlessert | A does a call pickup |
| 17:21.07 | Kobaz | the local ski resort is opening at 2 |
| 17:21.08 | kpfleming | is it wet or dry snow? |
| 17:21.12 | Kobaz | dry here |
| 17:21.13 | niekvlessert | with *8 (that's our way) |
| 17:21.18 | Kobaz | POWDER! |
| 17:21.22 | jeffg | jsmith! |
| 17:21.27 | jsmith | kpfleming: The bottom 19 inches are wet (heavy as concrete!), and the top three inches are fluffy stuff |
| 17:21.37 | niekvlessert | phone A will display *8 instead of the number |
| 17:21.42 | niekvlessert | same fix i guess? |
| 17:21.43 | kpfleming | already supported in 1.8 |
| 17:21.44 | jsmith | jeffg: How you been? Long time, no chat... |
| 17:21.50 | Kobaz | niekvlessert: same thing |
| 17:21.57 | kpfleming | really, you should just try this stuff rather than spending more time asking whether it is supported/fixed |
| 17:22.09 | jeffg | jsmith, has been a bit. bumped into a guy a while back who knows you from DCLUG or whatever it's called :) |
| 17:22.11 | niekvlessert | kpflemming |
| 17:22.21 | niekvlessert | lemme just make sure the last one :) |
| 17:22.22 | jsmith | jeffg: Either FredLUG or NoVaLUG |
| 17:22.28 | jeffg | fred, yeah :) |
| 17:23.25 | niekvlessert | a dials b, b picks up, b puts a on hold, b dials c, c picks up, b announces that he will transfer a, b transfers a to c |
| 17:23.34 | kpfleming | already supported |
| 17:23.46 | niekvlessert | problem: phone c will still display the connection with b instead of a |
| 17:23.50 | niekvlessert | same thing again i guess... |
| 17:23.51 | Kobaz | it's CONNECTEDLINE, or something |
| 17:23.52 | Kobaz | the feature |
| 17:24.13 | kpfleming | and also redirected (forwarded) calls will show the redirecting party's name/number, if the phone can display it |
| 17:24.25 | kpfleming | including redirections that happen on the PSTN and arrive via ISDN signaling |
| 17:24.51 | niekvlessert | forwarded by the phone u mean? when i forward in asterisk it's blind transfer |
| 17:24.55 | niekvlessert | and that'll work |
| 17:25.04 | kpfleming | no, forward != transfer |
| 17:25.25 | niekvlessert | you said redirected (forwarded) |
| 17:25.34 | kpfleming | right... neither of those are transfers |
| 17:25.55 | niekvlessert | so they work now |
| 17:25.56 | niekvlessert | ok |
| 17:26.02 | niekvlessert | so my 3 problems are all i have |
| 17:26.12 | niekvlessert | or can you experts think of some other scenario? |
| 17:26.33 | kpfleming | we can come up with scenarios all day long but what would be the point? |
| 17:26.55 | niekvlessert | so i can decide if the missing stuff is acceptable... |
| 17:27.13 | niekvlessert | because using trunk in production environments is not an option... |
| 17:27.22 | niekvlessert | maybe i should opensips to do this stuff... |
| 17:27.40 | kpfleming | how can you know what is missing if you don't know what scenarios you want to support? |
| 17:27.41 | niekvlessert | and use asterisk for the media |
| 17:27.45 | jeffg | jsmith, hate that i missed astricon last year, just had no time to work on a talk. this year i'm resolved to have a nice bench of adaptable talks prepared before end of Q1 :) |
| 17:29.07 | niekvlessert | kpfleming: easy, you know my 3 problems... maybe there are situations where similar things will happened which i didn't think of yet... |
| 17:29.14 | niekvlessert | *happen |
| 17:29.23 | kpfleming | there are probably thousands |
| 17:29.42 | kpfleming | but all the ones that have been thought of have been covered in the trunk code |
| 17:29.50 | niekvlessert | maybe in your definition of similar |
| 17:29.51 | kpfleming | and none of them are supported in previous branches |
| 17:30.26 | niekvlessert | would it be hard to backport connectedline? |
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| 17:30.33 | kpfleming | extremely |
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| 17:30.46 | niekvlessert | ok, too bad |
| 17:32.52 | niekvlessert | i will look into the trunk code tonight |
| 17:33.00 | kpfleming | why? |
| 17:33.43 | niekvlessert | to see if i can locate the lines doing the stuff.. |
| 17:33.52 | kpfleming | there are thousands of them, all over the tree |
| 17:34.02 | kpfleming | it affected chan_sip, chan_dahdi, the core, lots of stuff |
| 17:34.34 | niekvlessert | as stated before, i only need the sip support |
| 17:34.40 | kpfleming | that's not possible |
| 17:34.40 | niekvlessert | and the only thing it is, is this: |
| 17:34.41 | niekvlessert | by modifying the SIP Contact header in the 200 OK message and/or in a re-INVITE message |
| 17:34.50 | kpfleming | no, that is wrong |
| 17:34.59 | kpfleming | the Contact header is never modified |
| 17:35.09 | niekvlessert | ah ok |
| 17:35.37 | kpfleming | and since asterisk is not sip-based, the sip support requires all the core functionality |
| 17:36.20 | niekvlessert | well, can you please tell me what needs to be changed then? |
| 17:36.25 | niekvlessert | so i know where to look |
| 17:36.30 | kpfleming | what you are talking about is the very end of the process...how to supply the proper information to the phone. the hard part is knowing where that information is supposed to come from and when to send it. |
| 17:36.32 | Kobaz | niekvlessert: these guys take cash and checks |
| 17:36.40 | seanbright | 12:33 <@kpfleming> there are thousands of them, all over the tree |
| 17:36.41 | seanbright | 12:34 <@kpfleming> it affected chan_sip, chan_dahdi, the core, lots of stuff |
| 17:36.49 | kpfleming | i have already told you that backporting the code changes would be extremely difficult |
| 17:36.56 | kpfleming | for anyone, including the core developers |
| 17:37.01 | niekvlessert | as stated before, maybe sipsak can do it |
| 17:37.10 | kpfleming | sipsak cannot modify asterisk's behavior |
| 17:37.13 | niekvlessert | no |
| 17:37.20 | niekvlessert | but it can put information on the display of a phone |
| 17:37.33 | kpfleming | sure... so can netcat, sipp, lots of other ways to send SIP packets |
| 17:37.33 | niekvlessert | so i use agi, then sipsak |
| 17:37.49 | niekvlessert | you guys think i can get what i want then? |
| 17:37.49 | kpfleming | that will not work, because you won't be able to trigger the AGI when the display needs to be updated |
| 17:37.58 | Kobaz | well yeah |
| 17:37.59 | Kobaz | er |
| 17:38.01 | niekvlessert | sure i can with ami |
| 17:38.11 | Kobaz | wrong chan |
| 17:38.12 | kpfleming | good luck then |
| 17:38.27 | niekvlessert | well what I would like to know is this |
| 17:38.38 | niekvlessert | my knowledge of sip is not huge |
| 17:38.42 | kpfleming | the likelihood of that being worth the time to implement, and actually working, is very low, and it will *not* be more stable or production ready than using what is already in asterisk |
| 17:38.45 | niekvlessert | so what sip message are we talking about here |
| 17:38.54 | niekvlessert | true |
| 17:38.55 | kpfleming | it is not a message |
| 17:39.07 | niekvlessert | but the rest of asterisk 1.4.26.2 is more stable then trunk |
| 17:39.12 | kpfleming | it is a modification to existing INVITE messages, or possibly UPDATE messages, depending on what the phone supports |
| 17:39.33 | niekvlessert | we only use one type of phones |
| 17:39.52 | kpfleming | and you can't modify asterisk's existing INVITE messages with sipsak, and you can't send INVITE messages to a phone to have them treated as part of an existing dialog without breaking chan_sip |
| 17:40.44 | niekvlessert | ok, i'm reading this last thing 5 times |
| 17:41.32 | niekvlessert | and so it's also impossible to hack these sip invites for this 1 type of phone |
| 17:41.50 | niekvlessert | sorry for being persistent :) |
| 17:42.03 | kpfleming | it's not impossible to insert the additional headers required, but as i said, that is not the hard part. the hard part is knowing when to do it, and collecting the information to be sent. |
| 17:42.33 | kpfleming | it took one of our core developers 3-4 months to build this functionality, test it, and ensure it worked as expected |
| 17:42.44 | kpfleming | and that did not include the the chan_dahdi work, only chan_sip and the core |
| 17:44.16 | niekvlessert | maybe i can hack it in for my 3 scenario's... :) |
| 17:44.26 | seanbright | kpfleming: can you tell me which lines were added to which files? |
| 17:44.40 | niekvlessert | problem with the transfer thing: the phone does the transfer, not asterisk |
| 17:44.40 | kpfleming | sure, i can get you the first commit revision, and then you can find the subsequent ones |
| 17:44.42 | seanbright | kpfleming: i have to run - but you can e-mail that info to me |
| 17:44.49 | seanbright | kthx |
| 17:44.51 | seanbright | heh |
| 17:45.10 | kpfleming | that's right... transfers are hard to handle |
| 17:47.06 | niekvlessert | hmmmmmmm |
| 17:47.11 | niekvlessert | but what sipsak |
| 17:47.27 | niekvlessert | what if I push it to the phone AFTER the conversation has been esthablished |
| 17:47.27 | seanbright | 's head explodes |
| 17:47.49 | niekvlessert | it's dirty, i know |
| 17:48.13 | niekvlessert | but would you ppl think i can push stuff to phones with sipsak and rewrite lines on the display? |
| 17:48.21 | kpfleming | you cannot do that |
| 17:48.25 | niekvlessert | ok too bad |
| 17:48.34 | kpfleming | SIP dialogs have dialog identifiers and sequence numbers, you can't break the sequence |
| 17:48.50 | niekvlessert | ok, thank 4 the info |
| 17:49.01 | niekvlessert | then xml push is the only option i have left |
| 17:49.15 | niekvlessert | aastra phones support that |
| 17:49.29 | niekvlessert | i can just push whatever i want to the phone whenever i want... |
| 17:49.45 | kpfleming | and in the meantime, as jsmith said, you can start testing trunk and help to ensure that when we are ready to release 1.8, that it will work for you |
| 17:50.01 | niekvlessert | true |
| 17:50.07 | niekvlessert | installing a debian vm already... so.. |
| 17:50.10 | kpfleming | sure, you can push whatever you want, but it will difficult to find the proper mechanisms to trigger the data being sent, and collect the data to be sent |
| 17:50.15 | niekvlessert | well, i'm going to cook |
| 17:50.20 | niekvlessert | thanks a lot for the info |
| 17:50.25 | kpfleming | you are welcome |
| 17:50.56 | niekvlessert | cya! |
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| 17:51.08 | seanbright | yeesh |
| 17:51.48 | kpfleming | if he's dutch, he should head to brussels today or tomorrow and talk to putnopvut about how hard that work was |
| 17:52.08 | seanbright | he gets all the tough assignments |
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| 19:52.21 | RichardLynch | I'm a PHP guy with very rusty C skills, messing with asterisk source, and want to get the data behind "sip show peers". I'm a DRY kind of guy, and looking at _sip_show_peers, it seems like it would almost be best to pass in something for fd first arg that would be a like a string buffer, and just parse the output, rather than copy/paste that ASTOBJ_CONTAINER_TRAVERSE macro code/logic... So, can anyone refresh my memory about how one would pass |
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| 20:10.12 | niekvlessert | hello! when will csta be supported? |
| 20:10.17 | niekvlessert | asterisk 2.0? :) |
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| 20:19.59 | jsmith | niekvlessert: When someone writes the code, obviously. |
| 20:20.23 | jsmith | niekvlessert: You know the mantra -- "We accept patches" |
| 20:20.47 | niekvlessert | agree, but are there others who care? |
| 20:21.08 | niekvlessert | is installing trunk now... |
| 20:21.30 | jsmith | has no idea if people do or don't care about CSTA... jsmith can certainly live without it |
| 20:32.12 | niekvlessert | well, if we can just connect to jabber and have ocs features it will be fine... :) |
| 20:32.42 | niekvlessert | CONNECTEDLINE: Gets or sets Connected Line data on the channel. |
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| 20:37.40 | niekvlessert | well that is what i need... |
| 20:43.48 | niekvlessert | https://reviewboard.asterisk.org/r/256/ |
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