| 00:01.33 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 00:13.53 | igustin | question about filename format for recording... |
| 00:14.10 | igustin | I'm using touchtone combination *3 for start recording |
| 00:14.35 | igustin | which variable contains this filename? |
| 00:19.34 | *** join/#asterisk CeBe (~CeBe@89.204.135.141) |
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| 00:42.25 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 01:03.05 | dar123 | is there a way to check timing configuration of the service provider, my fxo inbound calls don't work. Was asked by digium to check with telco but they have no clue |
| 01:03.21 | dar123 | :( |
| 01:04.32 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
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| 02:03.45 | Micc | Can asterisk 12 support more channels than the older versions? I remember a limit of like 100-120 channels before, is that changed with PJSIP? |
| 02:04.17 | daemon | is there anyway I can tel asterisk to do a custom SQL query to find extension details |
| 02:10.02 | [TK]D-Fender | Micc: even a decade ago you could commonly get more than that |
| 02:10.36 | [TK]D-Fender | daemon: In the dialplan, yes |
| 02:11.15 | daemon | [TK]D-Fender, cool I want to tie my 'authed users' to my redmine server |
| 02:11.31 | daemon | I could just not figure howto pass it custom SQL to execute |
| 02:11.38 | daemon | to verify against the redmine db instead |
| 02:11.40 | [TK]D-Fender | daemon: You don't |
| 02:11.49 | [TK]D-Fender | daemon: * is it's own world. |
| 02:11.49 | Penguin | [TK]D-Fender: its |
| 02:11.54 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 02:12.23 | daemon | [TK]D-Fender, so I cannot override where its looking for information for specific modules? |
| 02:12.49 | [TK]D-Fender | daemon: by normal means, no.... |
| 02:14.08 | daemon | oh yay source hacking |
| 02:14.56 | daemon | mind you I could probably code triggers into psotgresql to keep a constant copy of redmines auth db in the asterisk db name in the format asterisk expects |
| 02:14.59 | daemon | might be easier |
| 02:15.16 | [TK]D-Fender | That would do it. |
| 02:15.25 | [TK]D-Fender | And I concur |
| 02:16.09 | [TK]D-Fender | A minor trigger to sync those is FAR easier than pretty much anything else. |
| 02:35.51 | *** join/#asterisk apwelsh (~apwelsh@71-95-60-31.dhcp.rvsd.ca.charter.com) |
| 02:36.00 | apwelsh | Anyone get Asterisk to install on OSX? |
| 02:40.33 | apwelsh | hello? |
| 02:41.44 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 02:55.30 | *** join/#asterisk apwelsh (~apwelsh@71-95-60-31.dhcp.rvsd.ca.charter.com) |
| 02:56.21 | apwelsh | Been trying to get help, not sure if IRC is working out. Anyone on use OSX for Asterisk? |
| 02:58.45 | [TK]D-Fender | On here now? Maybe |
| 03:01.22 | dar123 | any suggestions what should i do? |
| 03:01.49 | dar123 | i am stuck now |
| 03:02.10 | apwelsh | I have been trying to build Asterisk on OSX 10.9 (Mavericks) for days.. I'm a programmer, but very rusty with C. I was able to fix the Makefiles so that the linker links bundle1.o correctly, and I turned off optimizations so that all the code compiles, but now the linkers are failing.. they all report that the duplicate symbol on (and list around 313 functions) for chan_sip.o and sip/dialplan_functions.o |
| 03:03.07 | apwelsh | I have asked for help in asterisk-dev but no responses there... |
| 03:04.23 | [TK]D-Fender | Yes... over the past half hour... on a Friday night. You're lucky ANYONE is awake and here at that hour :) |
| 03:11.07 | apwelsh | where are the servers hosted? for me it's only 7pm, and I was also trying thursday since about 6pm (PST) is it maybe that I need to try in the morning/noon hours here? |
| 03:11.26 | [TK]D-Fender | What servers? |
| 03:12.43 | apwelsh | this IRC server(s)... (nevermind.. stupid question -- just realized what I was asking.. It's the users, not the servers that is the issue. haha |
| 03:13.41 | [TK]D-Fender | Best odds is always business hours, EST. Weekend lower than weeknight |
| 03:13.57 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 03:14.00 | [TK]D-Fender | Sit around be patient... maybe someone will walk in late and check in and answer |
| 03:16.32 | apwelsh | thanks. I typically use the IRC for support with jboss products, and since they are essentially funded by RedHat, they are only all the time so it just didn't cross my mind. |
| 04:04.34 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 04:17.53 | *** join/#asterisk Oreoboros_ (6c501543@gateway/web/freenode/ip.108.80.21.67) |
| 04:18.16 | Oreoboros_ | Hey, where can I get some help on this network that isn't PBX related? |
| 04:21.23 | Penguin | Do you need help milking your cat? |
| 04:53.00 | Oreoboros_ | Penguin: if only it were that easy. |
| 04:53.14 | Oreoboros_ | I'm dealing with a four line system and one of the lines is acting all sorts of strange. |
| 04:53.41 | Penguin | Sounds PBX related to me. |
| 05:04.07 | Oreoboros_ | Well, there's no PBX involved. |
| 05:04.19 | Oreoboros_ | And I can't find any other telephony related channels on #freenode. |
| 05:05.11 | Penguin | "four line system" is certainly related. You said your problem wasn't related. |
| 05:07.28 | Oreoboros_ | As far as I know, there is no PBX involved, at least not on my side of the demarc, unless someone hid it inside of a wall. |
| 05:07.41 | Oreoboros_ | The four lines are all static and run through a few modular jacks. |
| 05:16.34 | [TK]D-Fender | So what is "all sorts of strange"? |
| 05:17.00 | [TK]D-Fender | I've have a 50-pair bundle corrode over years where we'd lose pair after pair every year |
| 05:17.16 | [TK]D-Fender | So it'd be staticy, drop calls, etc |
| 05:17.33 | [TK]D-Fender | (POTS analog) |
| 05:32.42 | Oreoboros_ | [TK]D-Fender: the line reads 0 but it isn't the modular jack at all. I can rewire the pairs and get something through the same jack but the wire pair for that line never reads anything. |
| 05:32.59 | Oreoboros_ | The system is supposed to ring once then go to machine. When I call the respective line it just rings endlessly. |
| 05:33.12 | [TK]D-Fender | sounds like a cut... |
| 05:33.42 | Oreoboros_ | That's what I imagine but it just dropped out two weeks ago. All the other lines that run through the same bundle are just fine. |
| 05:33.56 | [TK]D-Fender | my situation was similar |
| 05:34.00 | Oreoboros_ | But the phone company does report the line is open after doing their multi-loop-whatever diagnostics. |
| 05:34.03 | [TK]D-Fender | a dozen good, one bad. |
| 05:34.13 | [TK]D-Fender | then I swap it.. then months later another pair would go. |
| 05:34.27 | [TK]D-Fender | Sometimes it's just a matter of time |
| 05:34.29 | Oreoboros_ | I don't think I can swap it. I would have to pull the whole deal out of the wall for that. |
| 05:34.52 | Oreoboros_ | I did notice however that the company's boxen outside are all open for whatever weird thing. |
| 05:34.59 | Oreoboros_ | Maybe someone fucked with them? |
| 05:35.01 | [TK]D-Fender | if you've got a big bundle with several unused they should be able to swap pairs easily |
| 05:35.09 | Oreoboros_ | Nothing unused. |
| 05:35.14 | [TK]D-Fender | THAT sucks |
| 05:35.16 | Oreoboros_ | Four pairs coming through, one to each RJ-11. |
| 05:35.36 | [TK]D-Fender | well dead wire is dead wire |
| 05:35.46 | Oreoboros_ | Would it keep ringing even if it was dead? |
| 05:35.55 | Oreoboros_ | I thought I would get the disconnected message. |
| 05:35.56 | [TK]D-Fender | to the caller? that's the telco for you... |
| 05:36.11 | Oreoboros_ | Could I try calling out from it? |
| 05:37.30 | [TK]D-Fender | Feel free to try. Do you get dialtone on pick-up? |
| 05:37.35 | [TK]D-Fender | I would doubt it.... |
| 05:37.57 | Oreoboros_ | Huh, let me see if I can dig out a handset to try with. |
| 05:38.02 | Penguin | We're running low on dial tone. |
| 05:44.57 | Oreoboros_ | Nope, no handset, and I really don't want to unplug one of the business lines to do it with. |
| 05:45.16 | Oreoboros_ | [TK]D-Fender: what's the best way to handle this if it is corrosion? Have the phone company replace it? |
| 05:47.46 | [TK]D-Fender | If it's from the demarc back, yes |
| 05:48.15 | Oreoboros_ | And if it is from my side? |
| 05:48.20 | Oreoboros_ | Don't tell me I'll have to rip and replace. |
| 05:51.14 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 05:51.31 | [TK]D-Fender | bad wire is bad wire |
| 05:51.36 | [TK]D-Fender | You already know the answer |
| 05:51.43 | Oreoboros_ | Alright, thank you for your help, [TK]D-Fender. |
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| 10:07.54 | JamKo | mjordan: You here? |
| 10:26.08 | LiuYan | hi, i registered to iptel.org using Asterisk 11.5.1 (Fedora 20), and added 'language=zh_CN' to [general] section sip.conf, but when a call come in, Playback application show the language is still 'en', is there some configs i'm missing? |
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| 12:37.46 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 12:38.47 | BKhan | Hi every body. A quick question |
| 12:40.00 | BKhan | I am taking input afetr palying BackGround function but sending dtmf and accepting on asterisk there is 8 seconds delay |
| 12:40.47 | BKhan | Is any option as we put dtmf immediately go to that option |
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| 15:23.42 | ThoMe | hello! |
| 15:24.51 | ThoMe | i use snom phones and i would like pickup a all when ringing. i use PickupChan. i can pick the call successfully. but i can't see the callerid when i pickup. |
| 15:24.55 | ThoMe | is it posible? |
| 15:27.24 | Gugge | ThoMe: i dont think so, but if it is, i would like to know how :) |
| 15:28.15 | ThoMe | Gugge: ok, dank ;) |
| 15:28.45 | *** join/#asterisk jsjc (~Adium@192.Red-79-151-33.dynamicIP.rima-tde.net) |
| 15:30.47 | ThoMe | Gugge: deutsch? alemen? german? |
| 15:30.59 | ThoMe | Gugge: my solution was sip.conf sendrpid=pai |
| 15:31.01 | ThoMe | done :P |
| 15:31.05 | k3asd` | hi |
| 15:31.06 | k3asd` | why I'm not receiving the ML by asterisk-users aprox 18 November? |
| 15:31.13 | k3asd` | there are problems? |
| 15:32.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 15:34.31 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 15:35.08 | ThoMe | emm. why works PickupChan(SIP/10); but not Pickup(10@intern); |
| 15:35.09 | ThoMe | ? |
| 15:35.41 | [TK]D-Fender | because the first IS a channel and the second ISN'T |
| 15:36.23 | ThoMe | [TK]D-Fender: :-) |
| 15:36.30 | ThoMe | [TK]D-Fender: good evening |
| 15:36.33 | [TK]D-Fender | actually... I may have missed something... |
| 15:36.39 | [TK]D-Fender | show your active channels and the attempt |
| 15:37.11 | ThoMe | [TK]D-Fender: i need Set(CHANNEL(callgroup)=1); and Set(CHANNEL(pickupgroup)=1); or ? |
| 15:37.38 | [TK]D-Fender | no |
| 15:37.49 | ThoMe | no? hm. |
| 15:37.59 | [TK]D-Fender | those are for featues.conf based pickup requests |
| 15:38.07 | ThoMe | [TK]D-Fender: hm. ok. |
| 15:38.19 | ThoMe | [TK]D-Fender: and how i can get the call from sip/10 with pickup() ? |
| 15:38.20 | [TK]D-Fender | Show what you're actually doing so we can see what is needed |
| 15:39.03 | [TK]D-Fender | what do you mean "FROM" sip/10? |
| 15:39.04 | ThoMe | [TK]D-Fender: [Dec 7 16:38:55] NOTICE[4554][C-0000004b]: app_directed_pickup.c:302 pickup_exec: No target channel found for $10@intern. |
| 15:39.27 | [TK]D-Fender | <PROTECTED> |
| 15:39.47 | [TK]D-Fender | ThoMe: No target channel found for $10@intern. <---- I see something bad right there |
| 15:39.56 | *** join/#asterisk Synx|hm_ (~Synx@unaffiliated/synx-hm/x-1623004) |
| 15:39.57 | ThoMe | [TK]D-Fender: hm. |
| 15:40.17 | [TK]D-Fender | ThoMe: there shouldn't be a $ in that error message |
| 15:40.24 | [TK]D-Fender | ThoMe: You put extra characters in |
| 15:40.33 | ThoMe | jojo. |
| 15:40.37 | ThoMe | <PROTECTED> |
| 15:40.41 | ThoMe | [Dec 7 16:40:20] NOTICE[4563][C-0000004d]: app_directed_pickup.c:302 pickup_exec: No target channel found for 10@intern. |
| 15:42.33 | [TK]D-Fender | ThoMe: Show your call attempt and your dialplan and active channels. |
| 15:42.35 | [TK]D-Fender | ~pb |
| 15:42.35 | 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. |
| 15:42.37 | [TK]D-Fender | ^^^^^ |
| 15:44.02 | ThoMe | [TK]D-Fender: but i try also this: [Dec 7 16:43:38] NOTICE[4588][C-00000057]: app_directed_pickup.c:302 pickup_exec: No target channel found for 10@intern i dont know what asterisk meaning. |
| 15:44.20 | ThoMe | [TK]D-Fender: http://paste.keks.be/452 |
| 15:47.34 | [TK]D-Fender | show the call |
| 15:47.44 | [TK]D-Fender | and the list of active channels before you do it |
| 15:48.23 | ThoMe | [TK]D-Fender: http://paste.keks.be/453 |
| 15:48.35 | ThoMe | core show channels is empty. |
| 15:49.27 | *** join/#asterisk micols (~t@81.161.188.225) |
| 15:49.40 | [TK]D-Fender | <PROTECTED> |
| 15:49.56 | [TK]D-Fender | +498920005256@eingehend <--- THIS is the EXTEN that is being "rung" |
| 15:50.11 | ThoMe | jep. the number is dialing. |
| 15:50.18 | ThoMe | [TK]D-Fender: have a solution |
| 15:50.34 | ThoMe | Set(nst=${EXTEN:2}); Set(GLOBAL(PICKUPMARK)=${nst}); Pickup(${nst}@PICKUPMARK); |
| 15:52.35 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:52.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:52.58 | [TK]D-Fender | You're using the wrong app. |
| 15:53.02 | BKhan | <PROTECTED> |
| 15:53.05 | [TK]D-Fender | just use PickupChan( |
| 15:53.19 | BKhan | Is any option as we put dtmf immediately go to that option |
| 15:53.40 | [TK]D-Fender | BKhan: "core show function TIMEOUT" <- |
| 15:53.50 | ThoMe | [TK]D-Fender: hm. |
| 15:54.04 | ThoMe | [TK]D-Fender: but Pickup and PickupChan bring a same results, or? |
| 15:54.23 | [TK]D-Fender | RESULT? Yes... both pickup something that is ringing... |
| 15:54.26 | [TK]D-Fender | BUT |
| 15:54.34 | [TK]D-Fender | they do not TARTET the same thing |
| 15:54.38 | [TK]D-Fender | TARGET* |
| 15:54.59 | [TK]D-Fender | PickUp() targets a DIALPLAN EXTENSION that is casing something to ring |
| 15:55.10 | [TK]D-Fender | causing* |
| 15:55.13 | ThoMe | [TK]D-Fender: hm. ok. thank you very much. |
| 15:55.24 | [TK]D-Fender | pickupchan() targets a DEVICE that is ringing |
| 15:55.30 | [TK]D-Fender | just like the instructions show |
| 15:56.01 | [TK]D-Fender | Pickup([extension[@context][&extension2[@context2][&...]]]) <-- dialplan very clearly |
| 15:56.20 | [TK]D-Fender | PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) |
| 15:56.34 | BKhan | D-Fender: Let me explain your scenario call land on IVR when user press (e.g) 1 or 2 it waits 7,8 second then take diesion against 1 or 2. Is it any scenario as we press 1 asterisk imediately take desion against DTMF |
| 15:56.35 | [TK]D-Fender | ^ tech/resouce = device |
| 15:56.59 | [TK]D-Fender | BKhan: Look at what is in your IVR |
| 15:58.05 | BKhan | [TK]D-Fender: Let me paste code on pastebin |
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| 16:01.15 | BKhan | [TK]D-Fender: http://pastebin.com/KfR9nwH4 |
| 16:01.22 | BKhan | PLease check its simple |
| 16:01.54 | [TK]D-Fender | BKhan: exten => _X.,1,Answer <- THIS is why your input is slow. this is a terrible way to make an IVR |
| 16:02.14 | [TK]D-Fender | BKhan: they can type 11 and wait and just keep getting the IVR in circles |
| 16:02.24 | [TK]D-Fender | BKhan: Never run an IVR on a pattern like that |
| 16:02.25 | ThoMe | [TK]D-Fender: can i ask the last question? :-) |
| 16:02.36 | [TK]D-Fender | ThoMe: Can you? |
| 16:02.39 | ThoMe | [TK]D-Fender: i run a script agi script (php). |
| 16:02.47 | ThoMe | [TK]D-Fender: works fine but asterisk said: [Dec 7 17:01:49] ERROR[4911][C-00000063]: utils.c:1321 ast_carefulwrite: write() returned error: Broken pipe |
| 16:03.30 | ThoMe | [TK]D-Fender: is my script shit? :-) |
| 16:03.33 | [TK]D-Fender | ThoMe: Your AGI is writing to an output where it should not |
| 16:03.43 | ThoMe | hm. |
| 16:03.48 | ThoMe | [TK]D-Fender: can i debug this? |
| 16:04.10 | BKhan | [TK]D-Fender: no when we press 1 call will go to 1 also we can see from CLI but issue is after pressing the DTMF and going to exten => 1,n,Dial(SIP/${EXTEN1}@provider) there is 7 seconds delay |
| 16:05.11 | BKhan | [TK]D-Fender: is any option as we press 1 and call will immeditly go to what we put against it |
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| 16:05.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:07.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:07.11 | [TK]D-Fender | Sorry crash... |
| 16:07.25 | [TK]D-Fender | missed everything after ThoMe asking if he can debug his AGI |
| 16:08.04 | ThoMe | [TK]D-Fender: yes. can i debugung this agi script |
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| 16:08.08 | ThoMe | ? |
| 16:08.40 | [TK]D-Fender | ThoMe: can you? Why are you asking me that? |
| 16:08.53 | ThoMe | because [Dec 7 17:01:49] ERROR[4911][C-00000063]: utils.c:1321 ast_carefulwrite: write() returned error: Broken pipe |
| 16:08.56 | ThoMe | :-) |
| 16:09.12 | [TK]D-Fender | ThoMe: Why COULDN'T you go debug your script? Are you asking me if you have the knowledge to debug it? Are you asking me for permission? I don't see why you should be asking that as a question... |
| 16:09.19 | [TK]D-Fender | FIX YOUR AGI |
| 16:09.39 | ThoMe | [TK]D-Fender:ok |
| 16:29.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:29.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 16:49.18 | ThoMe | emm |
| 16:49.27 | ThoMe | which variable i must override for the "to" header? |
| 16:50.04 | ThoMe | i would like change "to: +498920005256 " > "to 08920005256 " |
| 16:50.11 | ThoMe | CALLERID(dnid) ? |
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| 17:02.36 | [TK]D-Fender | to = what you dial |
| 17:02.54 | ThoMe | [TK]D-Fender: yes. can i change it? |
| 17:02.58 | ThoMe | only with ${SIP_HEADER(TO) ? |
| 17:03.03 | [TK]D-Fender | ... |
| 17:03.20 | [TK]D-Fender | no |
| 17:03.29 | [TK]D-Fender | look at what you are dialing |
| 17:04.21 | ThoMe | [TK]D-Fender: i would like change the dnid id. |
| 17:17.08 | ThoMe | hey. HOW i can change the dialing number? |
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| 17:19.29 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:19.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:20.24 | ThoMe | pabelanger: hello? |
| 17:20.34 | pabelanger | ThoMe, yes |
| 17:20.56 | ThoMe | pabelanger: hello. one question, i would like change the dnid/exten. rewrite the dialing number (incomming). |
| 17:21.22 | pabelanger | *CLI> core show function CALLERID |
| 17:21.27 | ThoMe | pabelanger: my sip provider sending "incomming call from +4917812345 to +4989123455). |
| 17:21.34 | ThoMe | pabelanger: but i would like 089123455 |
| 17:21.42 | ThoMe | pabelanger: i have Set(CALLERID(dnid)=123456789); |
| 17:21.45 | ThoMe | but doesnt works. |
| 17:22.06 | ThoMe | my snom phone show "to +4989123455" indeat "123456789". |
| 17:22.12 | ThoMe | pabelanger: can you help me please? |
| 17:23.22 | areay | hi all. i'm getting 'Could not set SRTP policies' when making a call over websockets on 11.6 using SIPml5. i've tried applying various patches to various versions of asterisk as per several blogs and posts, and attempted setting up following both SIPml5's and asterisk's own docs separately but to no avail... has anyone managed to get it working without using webrtc2sip? |
| 17:23.54 | ThoMe | pabelanger: :-( |
| 17:24.07 | pabelanger | ThoMe, <pabelanger> *CLI> core show function CALLERID |
| 17:24.21 | pabelanger | dnid is for outbound IIRC |
| 17:24.22 | ThoMe | pabelanger: i have. but i have Set(CALLERID(dnid)=123456789); |
| 17:24.23 | pabelanger | try num |
| 17:24.32 | ThoMe | pabelanger: num is the "FROM" or? |
| 17:24.37 | pabelanger | try |
| 17:24.49 | pabelanger | There is like 50 different settings you can use |
| 17:25.04 | pabelanger | Set(CALLERID(all)="Foo" <1234>) |
| 17:26.18 | ThoMe | pabelanger: yes. i have also to try Set(CALLERID(all)="Foo" <1234>). then said snom "you have a call from foo > + +49891234567 |
| 17:26.28 | ThoMe | but i would like change the value 49891234567 |
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| 18:06.42 | JamKo | mjordan: you around? |
| 18:08.39 | JamKo | Why is it that asterisk has to renegotiate to a different RTP source port, from the original rtp source port, when setting up T.38 calls? This behavior is specific to asterisk, and I believe |
| 18:09.22 | JamKo | the reason for why the setup doesn't work with commercial equipment. Sonus, Cisco etc, they all use the same rtp port when the call is reinvited to t.38 |
| 18:10.16 | JamKo | Why not asterisk as well? Is there something inherent to asterisk which requires this change? It's makes for a rather nasty environment. |
| 18:12.52 | file | T.38 isn't conveyed over RTP, it uses UDPTL |
| 18:13.17 | file | to have two protocols share the same port requires a network architecture that easily allows that, which doesn't exist in Asterisk right now |
| 18:16.14 | JamKo | file: Well then why the setting in sip.conf for t38pt_rtp? |
| 18:16.43 | file | there is a standard for transporting T.38 over RTP, but there exists only one or two actual implementations in the wild so it's pretty much not used |
| 18:17.16 | file | and the source of the T.38 support probably envisioned implementing it one day, but never happened |
| 18:18.06 | JamKo | I look at this as a bug, but it's probably viewed as an "improvement." Should I try it under bug? It's definitely buggy. |
| 18:18.44 | file | the issue mjordan linked to is the same thing |
| 18:21.08 | JamKo | It's unfortunate to the point where T.38 pass-through support is misleading. It doesn't work if you have t.38 traffic to and from commercial equipment. |
| 18:21.26 | JamKo | Does anyone know of a carrier that does not use sonus or dialogic? |
| 18:26.47 | JamKo | file: The part of your explanation for why it has to change ports gets fuzzy when you change the t38pt_usertpsource to "yes." With it set to yes, |
| 18:27.16 | file | that doesn't change the port that it is locally being sent from |
| 18:27.19 | JamKo | Asterisk does in fact send t.38 packets from the original rtp source port, but until it gets the 200ok from the far end on the t.38 reinvite. |
| 18:27.35 | JamKo | then it switches to the new port. |
| 18:27.44 | JamKo | yes it does. |
| 18:28.02 | JamKo | It sends 4 packets from whatever the original rtp source was in the 10000-20000 range. |
| 18:28.29 | JamKo | But it then switches after the 200, which kills the setup. The far end doesn't like it sending from the original port when it has a new port in the reinvite. |
| 18:29.33 | JamKo | So i believe the ability is there to keep the original rtp port, based on the evidence of changing that setting. |
| 18:30.51 | JamKo | and to clarify, it sends four t.38 packets from the original source port. |
| 18:32.50 | file | no... it doesn't |
| 18:33.14 | JamKo | Actually it does. I have a pile of captures here showing it. |
| 18:33.19 | file | that option changes the address that it is sent TO |
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| 18:35.28 | JamKo | Changes the address it's sent to for what reason, and what would it possibly change it to? That makes no sense. |
| 18:35.45 | JamKo | And regardless, I have the captures showing this behavior. |
| 18:35.47 | file | it changes it to the IP address that RTP was received from |
| 18:36.13 | JamKo | Well that might be what someone defined it as, but it's not what it does in real life. |
| 18:36.14 | file | it does NOT change the *source* of the UDPTL traffic |
| 18:36.18 | JamKo | yes it does. |
| 18:36.21 | JamKo | who wants to see a capture? |
| 18:36.23 | file | show me a capture |
| 18:37.01 | JamKo | Ok, message me with a place to send it. I'd rather not post a public capture until absolutely necessary. |
| 18:37.08 | file | jcolp@digium.com |
| 18:37.34 | JamKo | Ok give me a few to find the right ones. I have about 75 here. |
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| 18:52.15 | JamKo | file: Sent. ty for taking a look. |
| 18:53.47 | file | packet 618 is not T.38 |
| 18:54.27 | file | it's normal RTP |
| 18:55.19 | JamKo | Then why is wireshark showing it has protocol t.38? |
| 18:55.27 | JamKo | **as |
| 18:55.29 | file | doesn't for me... |
| 18:55.36 | file | shows RTP with G.729 |
| 18:56.23 | file | then at 636 UDPTL actually starts passing |
| 19:01.14 | JamKo | file: what version of wireshark are you using? Im sending you a screenshot showing this as t.38. |
| 19:01.25 | file | 1.10.2 |
| 19:07.48 | file | yeah, that's not UDPTL |
| 19:07.59 | file | it's malformed plus the sequence number is at the ceiling |
| 19:08.36 | file | Right Click, hit Decode As, and select RTP |
| 19:09.54 | JamKo | Did that, no change. |
| 19:10.08 | file | your wireshark is confused because of the protocol change, as it's the same dst port |
| 19:10.21 | JamKo | I see what you're saying though about the sequence number. |
| 19:11.13 | file | 636 is a perfectly fine UDPTL packet, and the first - you can tell because it's sequence number is 0 |
| 19:13.46 | JamKo | So would changing the t38pt_usertpsource to "yes" cause this? Because this is what I changed and it only does it when set to yes. |
| 19:14.06 | JamKo | Or I should say I only see it in wireshark as t.38 when this setting is yes. |
| 19:14.08 | file | cause what? |
| 19:14.25 | JamKo | Cause wireshark to think it's udptl t.38. |
| 19:15.00 | JamKo | If I set it to yes, I don't see these initial packets. And this was level 3s big complaint about asterisk. |
| 19:15.19 | file | I don't know what wireshark does for that, and the behavior has clearly changed since it's fine in mine |
| 19:15.27 | JamKo | They kept barking that it's sending t.38 from the original rtp port, and then switching. |
| 19:15.39 | JamKo | Let me upgrade my wireshark and see what happens. |
| 19:15.42 | file | yeah, no, it wasn't |
| 19:15.52 | file | or isn't, rather |
| 19:16.28 | file | it will only send RTP from the original RTP port, but if that's still in-flight it's possible for it to happen very closely to the UDPTL packets being sent |
| 19:16.35 | JamKo | They were also saying, we are not getting these t.38 packets. Well this would explain why they are not getting them, because it's not t.38. |
| 19:16.56 | file | the equipment is probably switching from UDPTL back to RTP on receipt of them |
| 19:17.17 | file | and then discarding the UDPTL packets |
| 19:17.29 | JamKo | yup that makes sense. |
| 19:17.56 | JamKo | Thanks, so I'll leave this setting off regardless. |
| 19:18.37 | apb1963 | Is there anyone here in the USA that wouldn't mind calling me for a minute? I'm having a delay issue and I'm not sure if it's just outgoing, or if it's incoming as well. |
| 19:18.49 | apb1963 | So I'd like to test a couple of numbers. |
| 19:19.50 | JamKo | file: I'm still fuzzy on why this is confusing wireshark. Its coming from the original rtp port, and the reinvite is for a different port, so why would wireshark think it's t.38? |
| 19:20.02 | file | the destination port is the same. |
| 19:20.48 | JamKo | The same as later in the capture when we get the 200 on the new port? |
| 19:20.58 | JamKo | or same port i should say |
| 19:20.59 | apb1963 | Testing from a standard PSTN landline preferred if possible. |
| 19:21.51 | JamKo | Like you said it's a completely different sequence. |
| 19:24.44 | file | has to run |
| 19:25.10 | apb1963 | I suspect my delay issue is because of Google Voice |
| 19:25.19 | apb1963 | Anyone else using GV? |
| 19:25.47 | lvlinux | i am |
| 19:26.02 | apb1963 | Any problems with delays when you make calls? In the audio? |
| 19:26.24 | lvlinux | a bit, usually not bad though |
| 19:26.32 | lvlinux | maybe 200ms or so |
| 19:26.57 | lvlinux | sometimes it seems a little worse, but not too bad |
| 19:27.12 | apb1963 | Mine is pretty bad.... several seconds.. the conversation constantly goes "Hello? Hello??? Is anyone there? Oh. Ok, I asked you.... blah blah blah. Hello??" |
| 19:27.30 | lvlinux | hmmm yeah that's not right |
| 19:27.54 | apb1963 | Yeah... the question is... is it google... or is it something else? |
| 19:27.58 | apb1963 | And if it's something else... then what? |
| 19:28.17 | apb1963 | I'm not sure if it's on incoming calls too, I don't get enough to be sure. |
| 19:28.25 | lvlinux | well, how is it setup? Motif? |
| 19:28.30 | apb1963 | yes |
| 19:29.41 | lvlinux | do you have a SIP address too---does it work fine? |
| 19:30.11 | apb1963 | I dialed an echo test sip address from sip2sip.info .. and yes it worked fine |
| 19:32.21 | lvlinux | does GV work fine in a browser? |
| 19:34.03 | apb1963 | I've been meaning to test that. Hmmmm.... need someone to call... |
| 19:34.46 | lvlinux | go to my website www.ruelphoto.com and call the number there |
| 19:36.53 | apb1963 | I can ping 3000 miles away on the other coast, 100ms, locally 30ms... but phone calls 20 minutes up the road have 3 -5 second delays. |
| 19:37.00 | apb1963 | OK, going there now... thank you |
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| 20:33.56 | JamKo | file: ok so good news here. I am having success with sonus through another peering network. Question, what exactly is the t38pt_rtp=yes telling asterisk to do with the udptl? It's working with it set to yes. |
| 20:40.51 | daemon | has anyone here ever used asterisk with skype connect? |
| 20:41.46 | daemon | no technical questions but I was wondering about cost |
| 20:41.53 | daemon | I am not interested in making outbound calls per se |
| 20:42.04 | daemon | I just want it so people on skype can call people on my voip network and vice versa |
| 20:42.07 | daemon | softphones |
| 20:42.13 | daemon | mainly for conference calls etc |
| 20:42.20 | daemon | and chat groups on online games |
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| 20:52.45 | ChannelZ | I used to use the Digium channel driver one (Skype For Asterisk) but that's dead, and I've not used SIP For Skype or whatever it's called |
| 20:52.46 | bsdice | good evening |
| 20:53.48 | ChannelZ | I don't know if if SIP For Skype allows for Skype users to call you (IE Skype-to-Skype) or not. I'd think so, but this is Microsoft we're talking about now. |
| 20:54.00 | bsdice | How can I report a regression for this commit: https://reviewboard.asterisk.org/r/2988/ |
| 20:54.15 | bsdice | Jira issue https://issues.asterisk.org/jira/browse/ASTERISK-12117 |
| 20:54.45 | ChannelZ | Add a new one and reference them |
| 20:54.52 | bsdice | Create login on jira and submit a ticket? |
| 20:55.55 | [TK]D-Fender | [15:41]daemonno technical questions but I was wondering about cost <- it's right on their site |
| 20:56.21 | [TK]D-Fender | [15:53]ChannelZI don't know if if SIP For Skype allows for Skype users to call you (IE Skype-to-Skype) or not. I'd think so, but this is Microsoft we're talking about now. <- it does |
| 20:56.25 | ChannelZ | yeah.. you can add a comment to the existing bug but I don't know that anyone will notice, it doesn't re-open the issue AFAIK |
| 20:56.48 | daemon | [TK]D-Fender, I been looking and all it keeps doing is asking me to signyupo |
| 20:56.48 | ChannelZ | is amazed |
| 20:57.15 | bsdice | ChannelZ usually when bug is closed in jira it will not send any note out |
| 20:57.27 | ChannelZ | so there you go |
| 20:58.18 | [TK]D-Fender | daemon: http://www.skype.com/en/features/skype-connect/ |
| 20:58.43 | [TK]D-Fender | daemon: Buy channels and allocate credit to your SIP Profile. <--- when you actually sign in you'll SEE the costs. |
| 20:58.45 | bsdice | Account creation worked |
| 21:00.47 | daemon | ah |
| 21:15.05 | bsdice | got it |
| 21:24.43 | bsdice | Jira is best bugtracker by a long shot. I put all info into ASTERISK-22946 |
| 21:39.21 | lvlinux | apb1963: did you get the email? |
| 21:47.24 | dar123 | i do not have chan_dahdi.conf under /etc/asterisk/ |
| 21:47.50 | Penguin | Are you going to be using hardware for PSTN connectivity? |
| 21:48.22 | dar123 | my bad, i renamed it . sorry |
| 21:54.03 | apb1963 | lvlinux: Yes... just responded to it. Thank you! |
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| 22:04.00 | dar123 | good night all |
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| 22:56.13 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.6.0 (2013/10/21), 10.12.3 (2013/08/27), 1.8.24.0 (2013/10/21), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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