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| 00:18.36 | CCFL_Man2 | tzafrir: you there? |
| 00:21.31 | flenders | hey, how do I use the #include "filename.conf" thing on conf files? it says on extensions.conf that 'this is different from the "include" command that includes contexts within other contexts.' |
| 00:21.53 | Qwell | flenders: You just put it where you want the stuff to be added |
| 00:22.05 | Qwell | mid-context even... it's like a #include in C |
| 00:23.08 | flenders | so within a context, I can just add a bunch of 'exten' on a different file and just #include thatfile.conf after the [context_name]? |
| 00:23.14 | Qwell | sure |
| 00:23.19 | flenders | beauty |
| 00:23.21 | flenders | thanks mate |
| 00:23.51 | flenders | hey, how do you guys manage DNDs and call forwardings? |
| 00:24.38 | flenders | cause here, if someone dials 0 on the IVR, it dials all phones, and if any is set to forward calls, it will ring this external number, as well |
| 00:24.57 | flenders | so I just disabled CFWD on all handsets |
| 00:25.40 | flenders | but I was planning on putting together a UI, that people would just login using their browsers and set call forwarding there |
| 00:26.41 | c0dz3r0 | I setup asterisk as a voicemail system integrating it with a definity g1 -- pri tie trunk |
| 00:26.43 | c0dz3r0 | I setup asterisk as a voicemail system integrating it with a definity g1 -- pri tie trunk |
| 00:26.44 | flenders | so with this #include thing, I thought I could manage each person's config file, and change the Dial command to dial an external number, for example |
| 00:26.48 | *** kick/#asterisk [c0dz3r0!n=north@pdpc/sponsor/digium/Qwell] by Qwell (Spam much?) |
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| 00:28.09 | flenders | then, if someone dials 0 on the IVR, the hardphone will still ring, but if you dial one's extension, it would forward to an external number |
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| 00:37.30 | Strom_M | flenders: there are WAY more elegant ways to do call forwarding |
| 00:39.14 | flenders | Strom_M: can you give me an example? |
| 00:39.30 | Strom_M | yeah |
| 00:39.39 | Strom_M | you could set call forward status in the asterisk DB |
| 00:40.00 | Strom_M | then, before you ring a telephone set, you check to see whether the phone is forwarded elsewhere |
| 00:40.05 | Strom_M | if it is, then you dial that number instead |
| 00:40.43 | flenders | and do you set the number to be dialed on the db as well? |
| 00:41.13 | Strom_M | yes; call forward status is determined by the presence or absence of a value; if there's a number there, you assume the phone is forwarded to that number |
| 00:41.37 | flenders | ok, I'll read up on that |
| 00:41.43 | flenders | thanks |
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| 00:55.16 | flenders | Strom_M: any ideas how users would turn on call forwarding remotely? |
| 00:55.51 | CCFL_Man2 | when configuring a linux kernel for asterisk, what specific options need to be set other than timebase? |
| 00:57.31 | Strom_M | flenders: from outside the PBX? |
| 00:57.58 | *** part/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
| 00:58.32 | flenders | yeah |
| 00:58.47 | Strom_M | flenders: have a DID that answers with an IVR menu |
| 00:58.54 | Strom_M | "which extension would you like to forward?" |
| 00:58.58 | Strom_M | "enter your passcode" |
| 00:59.05 | Strom_M | "enter the number to forward to" |
| 00:59.23 | Strom_M | "extension 400 will be forwarded to 1-311-555-2368. To confirm, press 1." |
| 00:59.34 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:59.35 | Strom_M | something like that |
| 00:59.39 | flenders | gotcha |
| 00:59.57 | Strom_M | you can use vmauthenticate() to authenticate with their voicemail passcode |
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| 01:50.34 | AirCoder | most guys use debian distro for asterisk? |
| 01:51.53 | Strom_M | use whatever distro you like best |
| 01:52.31 | AirCoder | lol thats a tough call. |
| 01:52.32 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-175-234-137.socal.res.rr.com) |
| 01:52.55 | CoolGuy21 | hi, for some reason asterisk is seeing FROM_DID=s") in new stack "s" as the DID |
| 01:54.11 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 01:54.15 | Lucky7 | yawn |
| 01:54.28 | Lucky7 | just to get it out there |
| 01:54.30 | Maliuta | AirCoder: why should distro make a differnce? |
| 01:54.36 | Lucky7 | XO Communcations is retarded. |
| 01:54.56 | AirCoder | learnd that yeas ago lucky. |
| 01:54.59 | Maliuta | AirCoder: and unless you are running unstable or rolling your own packages the version of asterisk in debian is ancient |
| 01:54.59 | weasel00 | Lucky7, HAHAHHAHA... |
| 01:55.14 | Lucky7 | They forgot to put any kind of Call confirmation on my T1 |
| 01:55.27 | Lucky7 | so for the last 3 days, I've been poking around every possible hole in asterisk |
| 01:55.32 | Lucky7 | trying to figure out WTF was wrong |
| 01:55.33 | AirCoder | i dont like using packages |
| 01:55.43 | AirCoder | would rather compile myself. |
| 01:55.48 | Maliuta | AirCoder: FWIW I run debian/unstable on my asterisk box here at home ... I am a little anal about controling configs though |
| 01:55.50 | Lucky7 | and I've been on and off with XO Commun, and them always saying thier crap is right |
| 01:55.57 | Lucky7 | damn idiots x.x |
| 01:55.58 | Maliuta | AirCoder: so roll you're own packaged |
| 01:56.04 | Maliuta | packages even |
| 01:56.35 | AirCoder | ive been building my systems on umbutu compling myself. |
| 01:56.38 | Maliuta | otherwise why bother with a distro that is package based |
| 01:56.42 | AirCoder | was just courious what the rest of ya where doing. |
| 01:56.44 | weasel00 | Lucky7, send them a bill for your wasted time.... it tends to get their attention and they will run straight for a few months before return back to retards |
| 01:58.23 | Lucky7 | yea. |
| 01:58.31 | Maliuta | AirCoder: building debs isn't that much over just straight compilation from source, and you get all the added benefits of a package management system when it comes to upgrading and/or running the same binaries on multiple systems |
| 01:58.32 | Lucky7 | i need to |
| 01:58.41 | Lucky7 | On a side note |
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| 01:58.51 | AirCoder | took me 3 months to get a ptsn line installed at a local with xo lucky..... finaly made them break the contract and i moved on. |
| 01:58.59 | Lucky7 | wow |
| 01:59.06 | Lucky7 | 3 months, I would have died |
| 01:59.15 | Lucky7 | I wil lsay, our Field tech in the Austin area is great |
| 01:59.26 | Lucky7 | its just EVERYONE on the switch side / tech support side who is a moron |
| 01:59.34 | Lucky7 | I talked to a girl last night |
| 01:59.39 | Lucky7 | told her i was setting up a * box |
| 01:59.45 | AirCoder | maliuta, your sayen just build my own debian package? |
| 01:59.46 | Lucky7 | first, she didn't know what asterisk was. |
| 01:59.55 | Lucky7 | which, ya know, whatever, its not the biggest PBX in the world |
| 02:00.02 | Lucky7 | but then she didn't know what open-source ment. |
| 02:00.05 | weasel00 | i finally got a point to point installed at out office in HongKong... took 10 months... 2 months for the local teclo in hk to redo the circuit cause MCI ordered the wrong one =) |
| 02:00.17 | Lucky7 | lol |
| 02:00.20 | Lucky7 | that blows |
| 02:00.53 | Lucky7 | Hm. Wierd, for some reason, I have 5-6 softphones currently setup |
| 02:00.57 | weasel00 | Lucky7, it was supposed to be 6 months from the contract sign date they promised.. i get free data for 8 months because of them ;() |
| 02:01.05 | Lucky7 | but any time i call them via the outside, it goes to voicemail o,0 |
| 02:01.29 | Lucky7 | http://rafb.net/p/9yocBR91.html //// sip show peers |
| 02:02.20 | Lucky7 | http://rafb.net/p/X0BRO837.html ///// calllog for incoming call to 140. (my phone) |
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| 02:02.46 | Maliuta | AirCoder: yes |
| 02:02.50 | Lucky7 | yea, its freePBX, and currently #freePBX is being eaten by someone who has DID issues, figured i might give it a shot here, and see if someone sees something i dont. |
| 02:03.53 | Maliuta | AirCoder: if you are going to run a distribution that is package managed your software should be part of that system |
| 02:04.32 | Lucky7 | ah |
| 02:07.44 | AirCoder | havent built packages yet but i'll look into it cant be to difficult. |
| 02:10.38 | Maliuta | AirCoder: it's not, you just need to wrap your head around a few things, 'specially doing multibinariy and shlib packages |
| 02:11.32 | Maliuta | AirCoder: it is worth basing your stuff heavily on the packaging work done by others, the guys that roll the asterisk packages for debian do a good job. look at their stuff |
| 02:11.59 | AirCoder | will do |
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| 02:16.09 | luke-jr | Anyone know how to unlock a PAP2-NA? |
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| 02:34.17 | flenders | Strom_M: you around? |
| 02:34.39 | flenders | or even anyone else, is there anything wrong with this? |
| 02:34.40 | flenders | exten => s,n(dial-cfim),Dial(Local/${DB(CFIM/${ARG3})@intern,15) |
| 02:35.28 | flenders | No such extension/context XXXXXXXXX@default creating local channel |
| 02:36.07 | flenders | it should be dialing XXXXXXXXX@intern, shouldn't it? |
| 02:36.41 | flenders | and intern includes my outgoing context, which then can dial out |
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| 02:38.17 | rnelson | I have a question about strings in expressions in the dialplan. |
| 02:38.17 | Penggu | hi all. how would you check if an (sip) extension has voicemail configured for it or not? |
| 02:40.30 | flenders | Penggu: voicemail.conf? |
| 02:41.09 | flenders | the extension number and mailbox number on voicemail.conf are not tied. so you can: |
| 02:41.29 | flenders | exten => 1000,n,Voicemail(2000) |
| 02:42.02 | flenders | and on sip.conf, you can add mailbox=2000 for sip user [blah] |
| 02:42.16 | rnelson | I'm trying to do something which, after variable expansion, looks like Set(var=$["\"xxxx\" yyyy" | ""]) |
| 02:43.20 | rnelson | But I get a parse error after the second quote. It looks like it is ignoring the fact that the quote is escaped. |
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| 02:48.59 | Penggu | flenders: is there any way to read the sip.conf lines in the dial plan? |
| 02:49.16 | Penggu | eg SipFeat("223","mailbox") |
| 02:49.44 | Penggu | basically i want to check if a sip peer has a mailbox configured and then decide whether or not to send them to voicemail |
| 02:49.53 | Penggu | (them = callers to the number in question) |
| 02:50.38 | Penggu | or does the mailbox= option go in to any ${VARIABLE} ? |
| 02:51.03 | flenders | Penggu: not sure you can do that just using the dialplan |
| 02:51.28 | Penggu | hmm, also the $variables would normally be related to the current channel, not necessarily who we're tyring to call.. |
| 02:51.43 | flenders | the mailbox setting on sip.conf is used to notify the peer when there's voicemail |
| 02:52.39 | flenders | Penggu: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MailboxExists |
| 02:54.18 | Penggu | ah great |
| 02:54.30 | Penggu | our mailbox numbers are == sip extension numbers |
| 02:54.36 | Penggu | so it's no feat to work out |
| 02:56.06 | *** join/#asterisk Aeudian (n=Aeudian@c-69-250-24-154.hsd1.md.comcast.net) |
| 02:57.16 | weasel00 | i keep getting this one extsion.. what does it mean? Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/2003-081fe7d0' in macro 'stdexten' |
| 02:57.44 | Aeudian | I am getting the following error when i reload asterisk "Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical'" now i have gone through the extensions.conf file and dundi-e164 and all references are commented out. Why am I still recieving this error? |
| 02:58.21 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:58.53 | ai-a | you missed one. |
| 02:59.21 | Strom_M | Aeudian: look in extensions.ael :) |
| 02:59.49 | rnelson | Hello? Anyone have any ideas about my question about strings with spaces AND double quotes? |
| 02:59.51 | Aeudian | correct me if i am wrong, but isnt the ael generated by my .conf? |
| 03:00.02 | dug | I am new to asterisk and I have setup ubuntu 7.04 with a zaptel card and build the modules which appear in the dmesg but when I run ztcfg -vvv I get channel one no such device |
| 03:00.43 | Strom_M | Aeudian: you're wrong |
| 03:01.11 | Strom_M | dug: does the card show up when you run lspci -bv? |
| 03:01.21 | dug | I am sure the devices are correct (fx0 in module one) and fxs in module 3 |
| 03:01.49 | Strom_M | FXO, not FX0 |
| 03:02.01 | Strom_M | Foreign eXchange Office |
| 03:02.06 | dug | Strom_M: no it doesnt |
| 03:02.17 | dug | Strom_M: appear in lspci |
| 03:02.19 | Strom_M | dug: pastebin the output of lspci -bv |
| 03:04.04 | dug | Strom_M: http://pastebin.ca/674294 |
| 03:04.34 | sevard | Foriegn eXchange 0ffice |
| 03:05.25 | Strom_M | dug: it's a TDM400? |
| 03:06.15 | dug | Strom_M a ZAPMICRO ZMA800P11 (TDM811B) |
| 03:06.26 | sevard | it's a ZOMG400 |
| 03:06.37 | Strom_M | dug: what the hell is "ZAPMICRO"? |
| 03:07.23 | dug | Strom_M: I thought it was a zaptel card .... not sure now |
| 03:07.43 | dug | Strom_M: Zaptel compat |
| 03:07.45 | Strom_M | link? |
| 03:08.30 | rnelson | Strom_M: I think it is a clone of the Digium cards. http://www.voip-info.org/wiki/view/ZapMicro |
| 03:08.50 | rnelson | http://www.zapmicro.com/ |
| 03:08.53 | dug | http://stores.ebay.com/TSpire-Solutions |
| 03:09.02 | Strom_M | looks like junk |
| 03:09.35 | Strom_M | ZMA800P looks like they took the TDM400P and did terrible terrible things to it :) |
| 03:09.56 | Penggu | i rmember reading this somewhere, but i can't find it |
| 03:10.11 | Penggu | if for eg we use execif(blah|AGI|arguments) |
| 03:10.18 | Penggu | where we want argumnets to be more than 1 thing |
| 03:10.30 | Penggu | would they be separated by commas? |
| 03:11.57 | Aeudian | Storm_M: what is the purpose of the extensions.ael file? its almost like my changes in extensions.conf are not making a difference unless i modify the .ael |
| 03:12.15 | Strom_M | who's storm? |
| 03:12.53 | Aeudian | Strom_M: sorry you, mistyped |
| 03:12.59 | Penggu | to answer myself, ExecIf($[${VMBOXEXISTSSTATUS} = FAILED]|AGI|festival-script.pl,'No!') works nicely |
| 03:12.59 | heelios | Aeudian: http://www.voip-info.org/wiki/view/Asterisk+AEL |
| 03:14.38 | Aeudian | heelios: thanks |
| 03:20.16 | Aeudian | What is "WARNING[4567]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener." and how do I stop this warning from appearing on a reload? |
| 03:22.38 | flenders | Aeudian: you can delete .ael files if you don't want them to be read |
| 03:22.45 | flenders | then reload |
| 03:23.45 | Aeudian | flenders: from what i am reading, new to the .ael side, these are experamental? What is the benefit if any over .conf |
| 03:24.05 | codefreeze | heelios: yes, the example AEL files are just translations of the extensions.conf files, and are not necessary. Also, see the http://voip-info.org/wiki/view/Asterisk+AEL2 page. The current impl in 1.4 and trunk is AEL2. |
| 03:24.07 | flenders | Aeudian: apparently they're more flexible |
| 03:25.12 | weasel00 | hmm..too bad i cant figure a way to get asterisk to use a standard modem to forward voip calls to my cell when im out of the office |
| 03:25.18 | flenders | Aeudian: I don't use it, and am not missing it |
| 03:25.35 | Aeudian | flenders: if the .ael file exists which does it reference this file before the .conf or only the .aef |
| 03:25.40 | Aeudian | pardon me .ael |
| 03:26.06 | flenders | it parses both .conf and .ael |
| 03:26.17 | flenders | you should stick to only one of them |
| 03:26.31 | codefreeze | Aeudian: the .ael can ref things in the extensions.conf stuff. |
| 03:27.09 | Aeudian | flenders: ya i just noticed the .ael after doing the newest asterisk compile and it was giving a lot of errors, ill probally stick to .conf |
| 03:28.04 | codefreeze | Aeudian: I need to clean it up, then.... |
| 03:29.02 | Aeudian | codefreeze: its not a ton of errors just a lot of dundi errors on fresh install |
| 03:29.36 | flenders | ael sounds very interesting, but I haven't had time to look into it yet, so I just delete .ael files and stick to the .conf ones. |
| 03:30.28 | codefreeze | Sigh. |
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| 03:46.28 | b11d | sigh indeed |
| 03:51.02 | Krurst | I'm having trouble getting on demand recording (*1) working after picking up a parked call. Can someone let me know it this is a bug, or if I have something screwed up in my config. |
| 03:52.53 | Krurst | I'm using the svn 1.4 branch |
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| 03:57.20 | *** join/#asterisk DustinO (n=aeolianm@c-24-11-125-186.hsd1.mi.comcast.net) |
| 03:57.58 | DustinO | Why would my 's' extension not be triggering? |
| 03:58.15 | DustinO | I always get a busy signal... Shouldn't the s extension always execute? |
| 03:58.26 | DustinO | ..regardless of extension |
| 03:59.49 | DustinO | Can anyone give me a minute of their time? |
| 04:00.10 | tzanger | DustinO: how are you trying ot get there? |
| 04:00.24 | tzanger | from TDM, you'll only hit 's' if the driver can't provide an extension (i.e. analog) |
| 04:00.32 | DustinO | My IP phone and softphone are directed to that context |
| 04:00.36 | tzanger | if from SIP/IAX, they provide an extension, but default to 's' I believe |
| 04:01.10 | DustinO | for a sip phone, what extension would i use to immediately drop the user to an ivr prompt? |
| 04:01.36 | DustinO | I have a context with just an s, and it's just not doing anything |
| 04:01.43 | DustinO | i'm not routing to it |
| 04:01.46 | [TK]D-Fender | DustinO, depends if the phone is CAPABLE of jsut throwing a call to a given exten upon going off-hook |
| 04:01.55 | [TK]D-Fender | DustinO, typically you DIAL a number from a phone. |
| 04:02.10 | DustinO | well, yeah |
| 04:02.11 | [TK]D-Fender | DustinO, Sever phones do support the "bat-phone" function |
| 04:02.15 | [TK]D-Fender | Several* |
| 04:02.24 | DustinO | but if i dialed a number, i would not be using an s, right, then? |
| 04:02.30 | DustinO | i would be using that number |
| 04:02.50 | fujin | bat phone? |
| 04:02.53 | DustinO | i'm just clarifying why it is that the s extension appears everywhere, but I can't seem to get it to work |
| 04:03.11 | tzanger | DustinO: when they pick up the set? you need to configure the phone to do that |
| 04:03.25 | [TK]D-Fender | DustinO, Correct |
| 04:03.27 | tzanger | if you dialed a number, it'd look to match THAT number in the dialplan |
| 04:03.33 | DustinO | ok |
| 04:03.41 | [TK]D-Fender | DustinO, "s" tpically is MEANINGLESS toa SIP phone. |
| 04:03.46 | tzanger | DustinO: exten => _X.,Goto(s,1) would probably do what you want :-) |
| 04:03.52 | DustinO | i thought the s extension was used to just ignore everything that had been entered to that point, and start fresh in that context |
| 04:03.58 | [TK]D-Fender | tzanger, CLOSE ;) |
| 04:03.58 | DustinO | ok |
| 04:04.12 | DustinO | so, 's' is used when, then? |
| 04:04.18 | tzanger | [TK]D-Fender: haha exten => _X.,1,Goto(s,1) |
| 04:04.29 | tzanger | DustinO: when the channel is incapable of giving you an extension |
| 04:04.31 | [TK]D-Fender | DustinO, but even using that exten you'd have to dial a number to start thre process, it wouldn't be on off-hook |
| 04:04.36 | tzanger | it means 'start' I believe |
| 04:04.44 | DustinO | yeah, but when does it trigger? |
| 04:04.47 | [TK]D-Fender | tzanger, getting WARMER, but still not quite ;) |
| 04:04.56 | DustinO | unless in a bat phone situation, aren't you always going to have to match against a number? |
| 04:04.57 | tzanger | [TK]D-Fender: what's wrong with that? |
| 04:05.18 | tzanger | 1 or more digits, priority 1, goto s exten, 1st priority in that same context |
| 04:05.18 | [TK]D-Fender | tzanger, Doesn't account for "any" number, let alone other chars ;) |
| 04:05.20 | fujin | read the docs :\ |
| 04:05.26 | tzanger | [TK]D-Fender: you picky bitch |
| 04:05.30 | [TK]D-Fender | tzanger, no, thats *2* or more ;) |
| 04:05.40 | tzanger | I thought '.' matched 0 or more digits |
| 04:05.42 | [TK]D-Fender | tzanger, not me, the DIALPLAN, I'm just the MESSENGER :p |
| 04:05.46 | fujin | yeah, but you've got _X. |
| 04:05.49 | fujin | waht you want is _. => |
| 04:05.51 | [TK]D-Fender | tzanger, Strike one! |
| 04:05.52 | DustinO | I've been reading the documentation, but I must be missing where it says that |
| 04:05.53 | Strom_M | no no no |
| 04:05.53 | fujin | -_- |
| 04:05.59 | Strom_M | ! is 0 or more |
| 04:05.59 | DustinO | they all just say that you just start there |
| 04:06.01 | DustinO | i think |
| 04:06.02 | Strom_M | you want _X! |
| 04:06.04 | Strom_M | never use _. |
| 04:06.09 | fujin | lol |
| 04:06.10 | tzanger | _. matches oshiat, which is probably not what you want |
| 04:06.15 | [TK]D-Fender | fujin, And words can barely describe how DUMB "_." is :D |
| 04:06.16 | tzanger | ahh |
| 04:06.19 | fujin | ;D |
| 04:06.33 | tzanger | ! is 0 or more, I have never had to use that |
| 04:06.37 | fujin | someone needs to put proper regex into the dialplan, like quantifiers |
| 04:06.40 | [TK]D-Fender | tzanger, So show me the RIGHT apttern now :) |
| 04:06.41 | fujin | game over |
| 04:07.02 | tzanger | haha |
| 04:07.06 | [TK]D-Fender | Strom_M, You're MUCH closer, but still a shade off :) |
| 04:07.11 | tzanger | _X! would be it then |
| 04:07.29 | Qwell | heh, I've been seeing _X! more and more lately |
| 04:07.35 | tzanger | fujin: no way, the dialplan needs to be written in an existing language instead of what's there now or AEL |
| 04:07.49 | [TK]D-Fender | Qwell : And STILL, none of them have gotten it quite right! |
| 04:07.58 | Qwell | _X! is exactly right |
| 04:08.00 | tzanger | [TK]D-Fender: enlighten me |
| 04:08.15 | Qwell | _X! > _X. :D |
| 04:08.17 | *** join/#asterisk ManxPower (n=manxpowe@015-843-184.area5.spcsdns.net) |
| 04:08.18 | [TK]D-Fender | qwell : Nope! |
| 04:08.23 | Qwell | _X. won't match 1 digit |
| 04:08.43 | Qwell | or, hmm, wait |
| 04:08.58 | DustinO | tzanger, I'm trying to understand your previous statement (sorry :) ): how do you land on an 's' ext when you're not already in that context? |
| 04:09.02 | [TK]D-Fender | the answer for his need : _[0-9*#]!,1, <------------ Account for Pound & Asterisk! |
| 04:09.10 | [TK]D-Fender | :p |
| 04:09.13 | Qwell | I didn't see his question : |
| 04:09.14 | Qwell | :p |
| 04:09.19 | DustinO | tzanger: that's just not possible? |
| 04:09.20 | [TK]D-Fender | tzanger, Silly you! |
| 04:09.21 | ManxPower | . means "1 or more" X means "one" |
| 04:09.24 | tzanger | DustinO: you never hit 's' from a device capable of giving you an extension... SIP phones can do that |
| 04:09.30 | [TK]D-Fender | tzanger, Accoutn for all touch-tone at least! |
| 04:09.31 | ManxPower | one + one or more = not match one. |
| 04:09.43 | tzanger | DustinO: so to hit it from SIP/IAX/PRI, you need to pass 's' as the extension, literally |
| 04:09.45 | Qwell | ManxPower: We're discussing _X. vs _X! |
| 04:09.51 | fujin | should really be .* |
| 04:09.58 | tzanger | DustinO: and since that's just silly, configure your phone to automatically dial something when it's picked up |
| 04:09.59 | fujin | .? |
| 04:10.04 | Qwell | huh... is . buggy? |
| 04:10.07 | [TK]D-Fender | tzanger, Well technically I *CAN* hit "s" from mine :p |
| 04:10.10 | ManxPower | Remember kids, don't irc when drunk. |
| 04:10.14 | tzanger | [TK]D-Fender: pssh, now you're just being pedantic |
| 04:10.29 | [TK]D-Fender | tzanger, And thats why my SIP invites WORK :p |
| 04:10.33 | tzanger | [TK]D-Fender: technically you forgot ABCD as well then |
| 04:10.38 | DustinO | txanger, all this time, i've been assuming someone was dialing in an extension, not transferring SIP-wise to an extension |
| 04:10.46 | DustinO | txanger, that explains our disconnect |
| 04:10.54 | tzanger | DustinO: ? |
| 04:11.00 | [TK]D-Fender | tzanger, * doesn't SUPPORT ABCD, so your argument is somewhat moot ;) |
| 04:11.08 | Qwell | [TK]D-Fender: sure it does |
| 04:11.16 | tzanger | I could have sworn I dialed abcd before with * |
| 04:11.20 | [TK]D-Fender | Qwell : Orly? |
| 04:11.22 | DustinO | tzanger, if i dial a DID into my SIP phone, asterisk will interrupt that as what extension |
| 04:11.26 | Qwell | You've just gotta have a phone with those keys |
| 04:11.28 | DustinO | tzanger, the did itself? |
| 04:11.32 | tzanger | DustinO: correct |
| 04:11.35 | DustinO | ah! |
| 04:11.36 | DustinO | ok |
| 04:11.38 | tzanger | you need to have a basic understanding of what's going on |
| 04:11.48 | [TK]D-Fender | Qwell : And what in the dialplan accounts for it? |
| 04:11.56 | Qwell | [TK]D-Fender: I've seen people use ABCD in their VM password. :) |
| 04:12.02 | tzanger | Qwell: hahaha |
| 04:12.08 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 04:12.14 | [TK]D-Fender | Qwell : Doesn't mean it WORKS :p |
| 04:12.16 | Qwell | [TK]D-Fender: the chars ABCD - no pattern match required |
| 04:12.18 | tzanger | limits the phones you cna send it from unless you've got a little recorder to play them back |
| 04:12.20 | DustinO | tzanger, well, if you always have a static did or are transferring in via SIP with an extension, how do you end up in 's' |
| 04:12.21 | Qwell | but it does work |
| 04:12.25 | DustinO | tzanger, bear with me here :) |
| 04:12.31 | tzanger | DustinO: by exten => 1234,1,Goto(s,1) |
| 04:12.44 | tzanger | when yo udial 1234, it matches that extension, and jumps to s, priority 1 |
| 04:12.59 | DustinO | tzanger, and that just allows a funnel of a bunch of extensions to end up in a simple 's'? |
| 04:13.13 | tzanger | DustinO: no, that allows the extenison '1234' to end up at 's' |
| 04:13.29 | [TK]D-Fender | DustinO, the whole point is you DON'T! You don't NEED "s"! If you want an "all-roads-lead-to-this-function", then you need a pattern-match (or series of them) that all GOTOT this other singular place. |
| 04:13.35 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 04:13.47 | [TK]D-Fender | DustinO, Whats that a metric bunch, or imperial? |
| 04:13.49 | DustinO | fender, ok |
| 04:14.28 | DustinO | fender, that's what I'm trying to find. it seems more significant than it is because so much documentation is laden with it |
| 04:14.40 | DustinO | fender, what do you mean? |
| 04:15.13 | Yourname` | Hi, is there a way I can set queue_priority BEFORE the call goes into the queue exten? For example, on an inbound call, like exten => 4192220000,1,Set(QUEUE_PRIORITY=10); exten => 419222000,n,Goto(testqueu,100, 1) |
| 04:15.21 | [TK]D-Fender | DustinO, You basically want a phone that does NOTHING except dump into an IVR pretty much immediately? |
| 04:15.28 | Yourname` | flenders: It worked! Thanks a lot for that Set callerID (name) thing. :) |
| 04:15.30 | DustinO | fender, yeah |
| 04:15.33 | DustinO | fender, exactly |
| 04:15.48 | [TK]D-Fender | DustinO, Do you already have the phone you want to do this to? |
| 04:16.01 | DustinO | yeah |
| 04:16.07 | [TK]D-Fender | DustinO, What is it exactly? |
| 04:16.08 | DustinO | already an extension and everything |
| 04:16.15 | DustinO | clearly, i'll just hardwire the extension in |
| 04:16.24 | DustinO | or pattern, etc.. |
| 04:16.43 | DustinO | by hardwire, i mean write the static did or pattern into the dialplan |
| 04:16.45 | flenders | Yourname`: no worries |
| 04:17.13 | DustinO | fender, i have both x-lite and a grandstream connecting over sip |
| 04:17.15 | DustinO | and iax |
| 04:17.32 | flenders | wow, I never really had a look on the asterisk sounds directory, and it's got EVERYTHING you need, hehe |
| 04:17.51 | [TK]D-Fender | DustinO, Ok, you danced around the question. Exactly WHAT HARDWARE to you want to enable this method of operation for? |
| 04:18.13 | DustinO | model-wise, gxp-2000 |
| 04:18.20 | DustinO | which seems to be crap, by the way |
| 04:18.34 | DustinO | i got it connecting via a static did |
| 04:18.40 | [TK]D-Fender | DustinO, Sad.... taking a multi-line office phone with screen, etc, and turning it into a 10 + ATA Bat-phone |
| 04:18.45 | DustinO | i just wanted to reconcile the 's' complication |
| 04:18.55 | [TK]D-Fender | DustinO, I believe you can' actually have it so you don't even have to dial anything |
| 04:18.58 | DustinO | no need for a bat-phone, sorry |
| 04:19.10 | DustinO | no... i'm just testing something else that's on a far bigger scale |
| 04:19.13 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:19.18 | [TK]D-Fender | DustinO, then you missed the point. "s" is not NEEDED for anything by you then |
| 04:19.52 | [TK]D-Fender | DustinO, Stop thinking about "s" when you want the phone to dial multiple extens to do DIFFERENT things. |
| 04:20.00 | DustinO | it was more of figuring out why it was made out to be so simple in all of the asterisk documentation/handbook, etc, and i couldn't get it to work the way they implied it should |
| 04:20.13 | DustinO | i'm not using s at all, by the way |
| 04:20.24 | DustinO | i just wanted to make sure i understood its significance, which is nil |
| 04:20.40 | DustinO | thanks for your help |
| 04:20.45 | [TK]D-Fender | DustinO, It IS simple. Its where calls coming from ANALOG LINES, and ANALOG PHONES in "bat-phone" mode go! |
| 04:20.45 | DustinO | <PROTECTED> |
| 04:20.50 | [TK]D-Fender | DustinO, Period. |
| 04:21.00 | DustinO | ok, gotcha |
| 04:21.04 | DustinO | thanks, fender |
| 04:21.06 | [TK]D-Fender | DustinO, Anything else comes IN with a target # |
| 04:21.37 | DustinO | tzanger: thanks, also |
| 04:21.39 | [TK]D-Fender | DustinO, You pick up your sip phone you dial a number. That gets passed to * from your phone. Bingo! * KNOWS the number you dialed. No need to cafall to a stupisd "s" |
| 04:22.26 | tzanger | DustinO: think of 's' as 'shitty phone tech that can't pass an extension' |
| 04:22.31 | [TK]D-Fender | DustinO, If you have an ISDN PRI (digital voice link), callers land on your B-channels because they dialed a DID that TARGETS your PRI. That TARGET is a KNOWN NUMBER. |
| 04:22.50 | [TK]D-Fender | DustinO, Ever had a line with 'distinctive ring' attached to it? |
| 04:22.56 | DustinO | i gok. currently, i configured the GS phone to connect to an Asterisk VM with SIP, from there to as asterisk server in coloc via IAX (to get through nat), and then from there to our main switches via SIP, and then out to our providers |
| 04:22.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:23.16 | DustinO | it's all working great, but I'm just trying to use good practice |
| 04:23.21 | DustinO | so, i'm good |
| 04:23.28 | DustinO | never needed one |
| 04:23.36 | [TK]D-Fender | DustinO, Ok, well as long as we've cleared som stuff up I guess. |
| 04:23.37 | DustinO | (with a distinct. ring) |
| 04:24.38 | [TK]D-Fender | DustinO, Well an analog line just rings when you call it. Your telco could forward a HUNDRED different numbers to it and your phone would have no idea WHAT number was dialed to make it ring. All it would know is that it is RINGING. Thats's why it dumps into "s". Because it doesn't know WHY. |
| 04:24.59 | [TK]D-Fender | Darn, a great answer WASTED :p |
| 04:25.10 | tzanger | [TK]D-Fender: actually that is a really great answer |
| 04:25.16 | tzanger | jbot should have that added |
| 04:25.23 | [TK]D-Fender | Here : |
| 04:25.26 | [TK]D-Fender | ~stdextens |
| 04:25.27 | jbot | [stdextens] "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. The "s" ... |
| 04:25.28 | [TK]D-Fender | ^^^^^^^^^^ |
| 04:25.34 | [TK]D-Fender | this was already pretty decent |
| 04:26.00 | tzanger | hmm |
| 04:26.03 | [TK]D-Fender | Rather explicit and stating many examples. |
| 04:26.11 | ManxPower | "s" does not stand for "start", it stands for "stupid" as in the tech used is too stupid to deliver the dialed number. |
| 04:26.21 | [TK]D-Fender | Someone should donate 100$ to the guy who wrote that ;) |
| 04:26.48 | [TK]D-Fender | ManxPower, "Stupid" always has a "Start", but no "End".... YOU should know that :p |
| 04:28.27 | b11d | haha |
| 04:28.29 | luke-jr | Anyone here done Sipura provisioning? :) |
| 04:29.27 | b11d | ... |
| 04:29.39 | [TK]D-Fender | weasel00, and the lesson is "Yes, [TK]D-Fender really CAN say the same thing 100 times nearly tirelessly!" |
| 04:29.54 | weasel00 | squestion... do i need to reload asterisk inorder for it release i have been dumping more moh music into it? |
| 04:30.36 | weasel00 | release/realize |
| 04:31.46 | [TK]D-Fender | weasel00, worst case : module reload res_musiconhold.so |
| 04:32.27 | weasel00 | i just reloaded the whole thing.. couldnt take it anymore =) |
| 04:36.31 | weasel00 | [TK]D-Fender, where do i start in debugging this error that came up when i did the reload 'Remote host can't match request NOTIFY to call ' |
| 04:37.15 | [TK]D-Fender | weasel00, Not a clue personally... Google is youre freebie start... |
| 04:37.23 | weasel00 | [TK]D-Fender, no one is in the system except me.. |
| 04:37.33 | weasel00 | [TK]D-Fender, tried.. no luck there =) |
| 04:38.03 | [TK]D-Fender | weasel00, wondering if thats a VM notification issue |
| 04:38.27 | [TK]D-Fender | weasel00, I think I heard of something recently concerning some sort of change from a "poll" to a "push" method. |
| 04:38.47 | [TK]D-Fender | weasel00, Just extrapolating, but I could be entirely off-track... |
| 04:39.51 | weasel00 | [TK]D-Fender, is vmail handled by chan_sip.c ? |
| 04:40.36 | [TK]D-Fender | weasel00, For the notification part, sure |
| 04:51.51 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 04:54.14 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:55.07 | weasel00 | [TK]D-Fender, client coudlnt re-register after the reload to get off of hold and back to the call. weird. |
| 04:57.19 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 05:02.13 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 05:07.45 | *** join/#asterisk sacitec (n=tobi@189.149.88.118) |
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| 05:12.18 | *** join/#asterisk Kenton (n=chatzill@h-64-105-100-225.mclnva23.dynamic.covad.net) |
| 05:13.25 | *** join/#asterisk ManxPower (n=manxpowe@015-843-184.area5.spcsdns.net) |
| 05:14.10 | CCFL_Man2 | which 2.6 kernel had 1000Hz timebase by default? |
| 05:15.09 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:25.27 | CCFL_Man2 | anyone here? |
| 05:25.28 | weasel00 | wierd.. the the conference room voice system is really choppy... |
| 05:29.20 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:34.00 | denon | weasel00: you have a zap timing device? |
| 05:34.33 | weasel00 | denon, ya... i dont think the box can handle it or the timing is off.. |
| 05:34.47 | denon | make sure you dont have any irq issues |
| 05:34.52 | denon | check cpu |
| 05:34.53 | denon | etc |
| 05:36.02 | weasel00 | almost 2 second delay... |
| 05:41.03 | grimsy | luke-jr: you were asking about sipura provisioning? |
| 05:41.44 | grimsy | I've done some with the SPA942 (which is linksys now) |
| 05:45.12 | flenders | grimsy: can you give details on that? |
| 05:45.57 | grimsy | basically i have php scripts that run each night via cron jobs |
| 05:46.12 | grimsy | pulling info out of LDAP and creating the config files for each phone |
| 05:46.27 | grimsy | the phones then pick up any changes automatically via tftp |
| 05:47.36 | grimsy | i've made some other scripts that populate the personal directory too |
| 05:47.42 | grimsy | those are on my blog if you're interested |
| 05:47.48 | grimsy | http://grimsy.blogspot.com/search/label/SPA942 |
| 05:48.34 | b11d | another double.. |
| 05:49.09 | flenders | grimsy: interesting |
| 05:49.41 | grimsy | if people are interested, i can tidy up the provisioning scripts and post them |
| 05:49.48 | flenders | I read your previous post about the personal directory a while ago |
| 05:49.53 | grimsy | might take a little while though |
| 05:49.57 | grimsy | ah |
| 05:49.58 | grimsy | :D |
| 05:50.03 | flenders | tried to find your contact details |
| 05:50.04 | flenders | :D |
| 05:50.23 | flenders | but your profile has only your name on it |
| 05:51.03 | grimsy | just sent you my email |
| 05:51.22 | flenders | nice |
| 05:51.23 | flenders | thanks mate |
| 05:51.26 | grimsy | np |
| 05:55.53 | *** join/#asterisk mjmarrio (n=mike@219-90-205-152.static.adam.com.au) |
| 05:56.23 | mjmarrio | can anyone tell me if I can put macro extensions into realtime extensions table? |
| 06:00.53 | *** join/#asterisk Inez (n=faceoff@devel4.net) |
| 06:00.55 | Inez | hi ho |
| 06:01.14 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 06:03.24 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 06:03.51 | tengulre | Hi,all |
| 06:04.37 | luke-jr | grimsy: yeah |
| 06:04.45 | luke-jr | grimsy: any idea on decoding the binary format? |
| 06:04.55 | grimsy | ah, no i'm sorry |
| 06:05.07 | grimsy | you want to make your own firmware? |
| 06:05.12 | luke-jr | nah |
| 06:05.17 | luke-jr | I want to unlock my PAP2-NA |
| 06:05.19 | luke-jr | :p |
| 06:05.21 | grimsy | ah |
| 06:05.23 | grimsy | :D |
| 06:05.29 | grimsy | can't help i'm sorry |
| 06:07.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
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| 06:17.45 | Krurst | does parking break on demand record (feature *1) or is it just my setup? |
| 06:21.25 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 06:21.53 | Snake-eyes | is the following error message only appears when asterisk makes a new connection to mysql cause the old one was stale ? |
| 06:21.55 | Snake-eyes | mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away |
| 06:23.48 | famicon | eh, quick question |
| 06:23.52 | famicon | what IS asterisk |
| 06:24.16 | famicon | so far my view of it is "apache for SIP" |
| 06:25.14 | Snake-eyes | famicon, no, its software pbx system supporting multiply features and protocols |
| 06:25.45 | Snake-eyes | famicon, http://www.voip-info.org/wiki/index.php?page=Asterisk&utm_source=voip-info&utm_medium=navbox&utm_content=Asterisk |
| 06:27.32 | famicon | ok it get it |
| 06:29.05 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:35.08 | coil | i went on voip-info.org and was trying to look for a good standard sip setup guide, but the only one i found, the link is a 404... |
| 06:35.13 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:43.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:47.49 | Snake-eyes | coil, sip guide for what ? setting up sip in asterisk ? |
| 06:47.50 | *** join/#asterisk sacitec (n=tobi@189.149.88.118) |
| 06:48.00 | coil | yes |
| 06:48.00 | sacitec | hi |
| 06:48.30 | sacitec | does asterisk support T.38 passtrought to a zap channel ? |
| 06:49.10 | Snake-eyes | coil, I would look at this page http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 06:49.30 | Snake-eyes | coil, also look at the "See also" section |
| 06:49.42 | flenders | sacitec: I'm doing fax using a PRI and a TDM400 with a single FXS channel |
| 06:49.46 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 06:49.58 | flenders | though, people say it doesn't work, it does for me |
| 06:50.10 | sacitec | that's exactly what i want to do |
| 06:50.20 | sacitec | but with zap channel |
| 06:50.27 | flenders | I had no dramas to set it up. |
| 06:50.30 | sacitec | and fxs port |
| 06:50.35 | sacitec | asterisk 1.4 ? |
| 06:50.37 | flenders | yeah |
| 06:50.45 | sacitec | cool |
| 06:50.46 | flenders | sangoma A101 card for the PRI |
| 06:50.54 | flenders | and TDM400 with a single FXS |
| 06:51.07 | sacitec | i'll use sangoma A200 serie |
| 06:51.21 | sacitec | 1 fxs port for fax machine |
| 06:52.05 | flenders | what I heard was that sangoma a200 works better for fax than tdm400 |
| 06:52.14 | sacitec | hope so |
| 06:52.16 | sacitec | jijiji |
| 06:52.25 | flenders | :P |
| 06:54.07 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 06:54.28 | sacitec | thanks =) |
| 06:55.26 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-87-111.lns10.syd6.internode.on.net) |
| 06:55.51 | *** join/#asterisk nacer (n=nacer@l.alcolo.a.mpl.pastIX.net) |
| 06:56.59 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 06:58.26 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:59.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:01.53 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 07:02.01 | flenders | hey, can I match a caller id to 2 digits? I mean, a GotoIf that checks for the lenght of the callerid(num)? |
| 07:02.11 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 07:04.36 | kaldemar | flenders: use function LEN. |
| 07:10.32 | *** join/#asterisk dug (n=chatzill@adsl-71-131-39-119.dsl.sntc01.pacbell.net) |
| 07:13.07 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 07:13.22 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 07:13.33 | FlatFoot | morning all |
| 07:14.41 | FlatFoot | just looking at agents/queues , does anyone know is it possible to send via header a different ringtone per queue ? using the setvar |
| 07:16.06 | *** part/#asterisk Krurst (n=me@eth244.wa.adsl.internode.on.net) |
| 07:17.14 | sparq | Does anyone know of a WiFi SIP that isn |
| 07:17.21 | sparq | whoops |
| 07:17.28 | sparq | That isn't terrible, I was going to say |
| 07:21.20 | FlatFoot | sparq: I used the UTStar F5000 ( I think ) that wasn't to bad so long as you have good WiFi coverage |
| 07:26.25 | *** join/#asterisk obnauticus (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net) |
| 07:26.47 | obnauticus | Hey, im calling into my PBX from my cell phone and I don't think it can hear me or my DTMF tones. |
| 07:26.59 | obnauticus | I get this error: http://pastebin.ca/674429 |
| 07:28.41 | obnauticus | That's when my pbx is doing a WaitExten |
| 07:29.50 | kaldemar | are you sure it is doing a WaitExten? that looks like you're trying to reach some sip device that is unreachable. |
| 07:30.16 | obnauticus | huh? |
| 07:30.36 | obnauticus | It's an IVR menu. |
| 07:30.48 | obnauticus | I mean.... |
| 07:30.50 | obnauticus | a regular menu ... |
| 07:33.37 | kaldemar | obnauticus: you're very unlikely to get any help with that little output. |
| 07:33.51 | obnauticus | k hold on. |
| 07:33.52 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 07:34.05 | obnauticus | http://pastebin.ca/674435 |
| 07:35.11 | obnauticus | http://pastebin.ca/674436 |
| 07:35.13 | obnauticus | there's my config. |
| 07:37.50 | Strom_M | obnauticus: have you tried just having that SIP connection bridge directly to a phone and seeing if you can talk both ways? |
| 07:38.09 | obnauticus | Now, but it has worked with this config and setup before. |
| 07:38.27 | Strom_M | with the same SIP connection to the ITSP? |
| 07:38.33 | obnauticus | Yes. |
| 07:38.33 | kaldemar | there seems to be something wrong with the SIP connection. maybe your provider has changed something? |
| 07:38.44 | obnauticus | IPKall>? |
| 07:39.03 | Strom_M | isn't that the free and really sucky one? |
| 07:39.14 | obnauticus | Better than nothing :/ |
| 07:39.29 | Strom_M | step 1: try paying money for something of actual quality |
| 07:39.42 | b11d | . |
| 07:39.43 | obnauticus | They are in my LOCAL area.. |
| 07:39.50 | obnauticus | the local NPA is 360 here. |
| 07:39.52 | obnauticus | That is why I use it. |
| 07:40.24 | Strom_M | 360 is a fairly large area code |
| 07:40.32 | Strom_M | are you sure their rate center is local to your rate center? |
| 07:40.39 | obnauticus | Yes. |
| 07:40.53 | Strom_M | less than twelve miles away? |
| 07:40.58 | obnauticus | No idea :/ |
| 07:41.04 | obnauticus | I'm sure i pay for local rates to them though |
| 07:41.04 | Strom_M | merely being in the same area code is not the definition of "local" |
| 07:41.20 | obnauticus | IS there anything wrong with my menu config though:? |
| 07:41.48 | Strom_M | apart from some deprecated syntax and complete misuse of the # key? no |
| 07:42.07 | obnauticus | Okay. |
| 07:42.09 | Strom_M | what prefix is your DID, and what prefix are you calling from? |
| 07:42.26 | obnauticus | calling from 771 -> 968 |
| 07:43.21 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 07:43.50 | Strom_M | 771 is Vancouver, WA |
| 07:43.59 | Strom_M | 968 is Shelton, WA |
| 07:44.05 | obnauticus | damn. |
| 07:44.26 | Strom_M | welcome to telephony; enjoy your stay |
| 07:46.01 | *** join/#asterisk The_LightSide (n=JBouncer@dsl-241-102-56.telkomadsl.co.za) |
| 07:49.28 | *** join/#asterisk tld (n=terje@elde.net) |
| 07:50.21 | tld | Is there any way to make a free call from or to a cellphone? A cellphone user still has to pay when calling an 800-number? |
| 07:51.02 | Strom_M | tld: if you're in north america, then yes, the mobile subscriber pays for airtime regardless |
| 07:51.08 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 07:52.42 | tld | Wondering about USA yes. So no way for someone calling a cellphone to accept paying the airtime charges, and make the call free for the cellphone user? |
| 07:52.48 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:53.01 | Strom_M | tld: no, sorry, it doesn't work like that |
| 07:53.19 | tld | too bad. :( |
| 07:53.22 | tld | thanks for the info. |
| 07:53.25 | Strom_M | what are you trying to do, exactly? |
| 07:54.03 | tld | I have a friend in the US, and I'd like to be able to call her and chat without it costing her money. |
| 07:54.17 | tld | I'd snailmail her an ATA, but she doesn't have internet at the moment. |
| 07:54.40 | Strom_M | well, usually mobile phone plans in the US include ridiculous amounts of airtime anyway |
| 07:55.10 | Strom_M | for example, the voice portion of my plan is US$40 for 1,000 minutes each month |
| 07:55.13 | tld | hmmm. I guess I should ask her to look into it. Or see about getting a landline or internet connection. |
| 07:55.23 | tld | sounds good. :) |
| 07:57.09 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.8.115.revip2.asianet.co.th) |
| 07:58.05 | coil | Strom_M: tmobile? |
| 07:58.23 | *** part/#asterisk BugKhaM (n=LAMER@ppp-58.8.8.115.revip2.asianet.co.th) |
| 07:58.46 | Strom_M | coil: yep |
| 08:06.23 | denon | tld: she doesn't have a landline? |
| 08:06.34 | denon | lots of cell companies are free unlimited inbound |
| 08:06.38 | denon | but all landlines are |
| 08:06.50 | denon | (in the US) |
| 08:07.45 | denon | if she doesn't have a landline at all, it's probably because her cell plan has 50 billion minutes a month like strom said |
| 08:07.59 | Strom_M | or because she's stupid |
| 08:08.03 | Strom_M | i've seen both |
| 08:08.10 | denon | I wasn't going to say that :) |
| 08:08.25 | denon | odds are this is his shiney new internet girlfriend, and he's trying to make a good impression :) |
| 08:10.11 | *** join/#asterisk shtoom (n=shtoom@123.252.144.92) |
| 08:14.34 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 08:15.10 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:15.17 | sparq | tld: I get free nights and weekends on my cell phone |
| 08:15.46 | tld | Sorry, was afk for a sec. |
| 08:16.01 | tld | denon, No landline. She moves around some, so hasn't made sense to get one. |
| 08:16.20 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 08:16.27 | McDouglas | hi |
| 08:16.38 | tld | denon, Real life female friend (not girlfriend though). She's Norwegian, moved over there to work for a while. |
| 08:16.55 | McDouglas | i got a strange error with asterisk |
| 08:17.00 | tld | denon, She's on a fairly tight budget, and from what I understood, probably on a prepaid. |
| 08:17.17 | McDouglas | i have it running, an analog phone is plugged in and an analog pots lie is also plugged in, al ok i can dial out and in |
| 08:17.35 | McDouglas | now, if i replace the analog line with a line coming frm a GSM adapter |
| 08:17.51 | McDouglas | it does not dial (giving a busy signal after a while) |
| 08:18.21 | sparq | tld: ask her to look into plans with free nights and weekends. Also, if she is ever around a WiFi connection, T-Mobile is selling phones that will hop on any available WiFi connection and give you unlimited calling. |
| 08:18.23 | McDouglas | but if i connect my analog phone directly to this GSM adapter i can dial without a problem (although it takes a bit longer to set up the call) |
| 08:18.54 | tld | sparq, thanks |
| 08:19.55 | Strom_M | McDouglas: well, first off, why the hell are you going GSM -> analog -> TDM -> IP? |
| 08:21.24 | McDouglas | what do you mean? |
| 08:21.48 | McDouglas | we have an analog line and a gsm line (via the gsm adapter) going into the pbx |
| 08:21.59 | McDouglas | and depending on the telephone number, the cheaper will be used |
| 08:22.06 | Strom_M | the GSM adapter is a horrid kludge |
| 08:22.16 | Strom_M | I wouldn't use it in production if I were you |
| 08:22.23 | McDouglas | if it cuts the price into half you have to deal with it |
| 08:22.35 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 08:22.36 | McDouglas | we have been using it since years (with an old bosch pbx) |
| 08:22.38 | Strom_M | yeah, and if it cuts your call quality into 1/5? |
| 08:23.02 | McDouglas | there is nothing worng with the quality |
| 08:23.05 | McDouglas | *wrong |
| 08:23.25 | Strom_M | you seriously can't hear the difference between g.711 and gsm> |
| 08:23.26 | Strom_M | ? |
| 08:23.40 | McDouglas | g.711 ? |
| 08:23.50 | Strom_M | sigh |
| 08:23.53 | Strom_M | ~thebook |
| 08:23.54 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:24.14 | McDouglas | oh you mean ulaw |
| 08:24.33 | McDouglas | well |
| 08:24.47 | McDouglas | there is a slight difference, but thats the price you pay for using GSM service |
| 08:24.55 | Strom_M | you call that "slight"? |
| 08:25.01 | Strom_M | yow. |
| 08:25.08 | McDouglas | but as i said we have been using this setup for years and it was sufficient |
| 08:25.28 | McDouglas | well |
| 08:25.37 | McDouglas | its the same quality as a cell phone cal |
| 08:25.39 | McDouglas | *call |
| 08:25.44 | McDouglas | whats wrong with that? |
| 08:26.54 | Strom_M | it's worse; you introduce plenty of extra problems by putting unnecessary A/D conversions in the link |
| 08:27.27 | The_LightSide | Strom_M, by nature the audio coming out of the GSM engine is analouge |
| 08:27.54 | McDouglas | Strom_M: BUT... we use the GSM adapters to call GSM numbers |
| 08:27.55 | McDouglas | nothing else |
| 08:27.59 | The_LightSide | altho GSM is basically ISDN |
| 08:28.03 | McDouglas | so even if we used land line to call a gsm phone |
| 08:28.13 | McDouglas | it wouldn't be that good anyway |
| 08:28.42 | The_LightSide | Strom_M, altho i do agree you get some really crappy gsm routers |
| 08:28.57 | Strom_M | McDouglas: if you insist on doing it that way, you can get GSM adapters which avoid the A/D conversion headaches by interfacing a GSM radio directly with the PC |
| 08:28.57 | McDouglas | router? |
| 08:29.35 | The_LightSide | McDouglas, in SA we call them gsm routers, but its really just a gsm interface |
| 08:30.42 | The_LightSide | Strom_M, avoid the A/D? thats A coming out the gsm engine, and then converted to D for the PC! |
| 08:31.28 | McDouglas | we cant buy new hardware |
| 08:31.49 | McDouglas | the decision to change to asterisk was made because its compatible with our current setup |
| 08:31.56 | McDouglas | well, i tought... |
| 08:32.07 | McDouglas | but i dont understand what is the difference... |
| 08:32.17 | McDouglas | if i connect an analog phone to the land line i can dial |
| 08:32.28 | McDouglas | if i connect an analog phone to the line coming from the gsm adapter i can dial |
| 08:32.39 | McDouglas | why cant asterisk? |
| 08:33.02 | Strom_M | The_LightSide: enlighten me, because as far as I'm aware, after it comes out of the vocoder, it's still a digital representation of a waveform which then has to go through a D->A converter |
| 08:33.10 | The_LightSide | McDouglas, if you can, maybe try increase txgan in asterisk, or DTMF sensitivity on the gsm device |
| 08:33.35 | Strom_M | McDouglas: or plug in a buttset and see what's going wrong by monitoring the call setup |
| 08:34.40 | The_LightSide | Strom_M, all the gsm engines used (the pc based one uses the motorola engine) have the analouge readily available on its pins |
| 08:35.14 | Strom_M | is that the sole option? |
| 08:35.23 | McDouglas | The_LightSide: how can i increase the tx gain? (can't access the gsm device) |
| 08:35.36 | The_LightSide | and i know for a fact that you still get awesome quality even converting back to D |
| 08:35.44 | McDouglas | Strom_M: buttwhat? :P sorry, i'm not a telco guy, just a sysadmin |
| 08:36.42 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 08:36.49 | Strom_M | GSM and awesome quality don't belong in the same sentence :) |
| 08:36.57 | The_LightSide | McDouglas, i think its /etc/asterisk/zapata.conf option txgain. Strom_M, can you confirm please? |
| 08:37.15 | The_LightSide | lol Strom_M.. for the larger part i agree... |
| 08:37.22 | McDouglas | dunno about your gsm services but our is comparable to the land line |
| 08:37.40 | Strom_M | McDouglas: then either your landlines are terrible or your ears are terrible :) |
| 08:37.55 | The_LightSide | and the gsm device is also VERY important |
| 08:38.09 | Strom_M | yes, rxgain and txgain are settable in zapata.conf |
| 08:38.24 | McDouglas | we use a Nokie TFE-2 adapter |
| 08:38.28 | McDouglas | *Nokia |
| 08:39.00 | The_LightSide | i used to work for a manufacturer of GSM to E1 converters. which give excellent quality |
| 08:39.07 | The_LightSide | McDouglas, change that ASAP |
| 08:39.21 | McDouglas | lol, why? |
| 08:39.24 | The_LightSide | its not half rate or dual band compliant |
| 08:39.25 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:39.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:39.34 | The_LightSide | altho its call quality is very good |
| 08:39.58 | Strom_M | full-rate is terrible enough, much less half-rate :) |
| 08:40.05 | The_LightSide | and the only router that allows analouge faxing |
| 08:40.11 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-41-46.w81-251.abo.wanadoo.fr) |
| 08:40.15 | McDouglas | what are half rate and dual band? |
| 08:40.15 | The_LightSide | Strom_M, agreed ;) |
| 08:40.35 | The_LightSide | thats a long story McDouglas |
| 08:40.51 | Strom_M | McDouglas: this is a buttset |
| 08:40.51 | Strom_M | http://www.etool.ca/GFX/PRODS/409-948.jpg |
| 08:41.25 | McDouglas | Strom_M: lol, probably couldnt operate it, or make any diagnostics |
| 08:41.33 | Strom_M | uh, it works just like a phone |
| 08:41.38 | Strom_M | because it IS a phone |
| 08:41.48 | McDouglas | The_LightSide: the manual says its GSM 900 and GSM 1800 compatible |
| 08:41.55 | McDouglas | isnt that dual band? |
| 08:42.09 | The_LightSide | the TFE 2?! |
| 08:42.28 | The_LightSide | the nokia 22 is |
| 08:42.55 | The_LightSide | is it black or white? |
| 08:42.55 | McDouglas | http://www.iptech.com.ua/downloads/gsmgate/nokia/nokia-premicell-users-guide-tfk2-eng.pdf |
| 08:43.12 | McDouglas | This guide describes the second generation of |
| 08:43.12 | McDouglas | Nokia PremiCell terminals TFK-2 and TFE-2 for |
| 08:43.12 | McDouglas | Nokia’s Fixed Wireless solution based on GSM |
| 08:43.12 | McDouglas | 900 and GSM 1800 wireless radio technology. |
| 08:43.23 | McDouglas | Its black |
| 08:43.49 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:43.55 | The_LightSide | interesting! |
| 08:44.25 | The_LightSide | my inet is super slow, still trying to load the page |
| 08:44.37 | McDouglas | but i dont think we use gsm 1800 |
| 08:45.30 | The_LightSide | hmmm, i think i was wrong... |
| 08:45.52 | The_LightSide | its just not half rate compliant, and it can lock up if the tower changes to half rate |
| 08:46.55 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.51) |
| 08:47.01 | McDouglas | well, it never locked up so far |
| 08:47.06 | McDouglas | and as i said we have been using it for years |
| 08:47.18 | McDouglas | btw, i just checked again |
| 08:47.30 | McDouglas | if i connect the analog phone to the nokia |
| 08:47.35 | McDouglas | after i dial |
| 08:47.38 | McDouglas | tere is a long pause |
| 08:47.44 | McDouglas | then a strange sound |
| 08:47.46 | McDouglas | then it dials |
| 08:47.55 | McDouglas | its like a click, or something |
| 08:48.04 | McDouglas | could that cause the problem with asterisk? |
| 08:48.26 | The_LightSide | line reversals..... it could indeed |
| 08:49.26 | The_LightSide | you need to check in the different signaling methods for analouge lines in zapata.conf |
| 08:49.36 | The_LightSide | im not too sure off hand |
| 08:49.55 | McDouglas | you mean LS and KS? |
| 08:51.11 | The_LightSide | yeah |
| 08:51.27 | The_LightSide | Strom_M, how do you set the reversal detection? |
| 08:51.46 | Strom_M | why would it be reversing /before/ supervising? |
| 08:51.49 | Strom_M | that's moronic |
| 08:52.07 | The_LightSide | good point... |
| 08:52.15 | Strom_M | are you sure it's a reversal sound? |
| 08:52.25 | McDouglas | i'm not sure what it is, lol |
| 08:52.47 | McDouglas | but its something i normaly dont hear if i dial on the land line |
| 08:52.54 | The_LightSide | it could be reversing if the number was incorrect.... |
| 08:53.01 | Strom_M | The_LightSide: no |
| 08:53.06 | McDouglas | the number is correct, dialed my cel phone |
| 08:53.15 | Strom_M | generally it only reverses if the number supervises |
| 08:53.16 | Strom_M | sigh |
| 08:53.32 | Strom_M | welcome to #asterisk, where the clueless and telephony crash head-on, typically causing spectacular messes |
| 08:53.36 | Strom_M | i'm going to bed |
| 08:53.42 | The_LightSide | lol Strom_M |
| 08:53.49 | The_LightSide | just a quick one tho |
| 08:53.52 | Strom_M | ~101 |
| 08:53.52 | jbot | i guess 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 08:54.08 | The_LightSide | all the devices in sa have a reversal and break on disconnect... |
| 08:54.19 | The_LightSide | which is the same as a wrongly dialed number |
| 08:54.52 | Strom_M | yeah, but it doesn't reverse and break until /after/ it plays you a recording, right? |
| 08:55.18 | The_LightSide | if the number is valid, yes |
| 08:55.43 | Strom_M | "ag man, you domkop, you've dialed the wrong number" |
| 08:56.05 | Strom_M | click buzz electrocution |
| 08:56.15 | The_LightSide | if its an invalid number... by invalid i mean less then the 10 digits (sa dialing) |
| 08:57.04 | The_LightSide | it just reversal, break engaged tone |
| 08:57.35 | The_LightSide | but anyways.... sleep well ;) |
| 08:58.13 | Strom_M | you didn't even laugh at my joke |
| 08:58.14 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:58.31 | The_LightSide | haha, i did actually, just didnt type it |
| 08:58.37 | Strom_M | :) |
| 08:59.00 | Strom_M | damnit, now i actually want boerewors. |
| 08:59.11 | Strom_M | i guess i'll have to go to that shop in beverly hills tomorrow |
| 08:59.16 | The_LightSide | boerewors?! |
| 08:59.28 | Strom_M | boerewors |
| 09:00.05 | Strom_M | is there something wrong with boerewors? |
| 09:00.05 | The_LightSide | wow... never thought id hear that on a chan like this ;) |
| 09:00.47 | Strom_M | heh |
| 09:01.06 | *** join/#asterisk AsteriskProblems (n=pbarnsle@81.171.174.178) |
| 09:01.15 | Strom_M | it's a staple at my family's barbecues here |
| 09:01.36 | AsteriskProblems | can anyone help me with a dialplan? Im using asterisknow but no-one is awake on that channel |
| 09:01.57 | Strom_M | AsteriskProblems: are you writing the dialplan by hand? |
| 09:02.14 | AsteriskProblems | i can do - but i tried using the gui first, but i cant dial out |
| 09:02.36 | AsteriskProblems | i can call between extensions, though i dont think that is set up right either because the consolde gives me "trunk" errors |
| 09:02.53 | Strom_M | ok, i'll ask it a different way then |
| 09:03.16 | Strom_M | are you having problems with a hand-written dialplan, or are you having problems configuring your system via the gui |
| 09:03.23 | AsteriskProblems | via the gui |
| 09:03.37 | Strom_M | well then i don't think you'll find much help here |
| 09:03.44 | AsteriskProblems | doh |
| 09:05.43 | AsteriskProblems | ok i'll try something else then... |
| 09:05.55 | Strom_M | try plain asterisk :) |
| 09:05.57 | AsteriskProblems | has anyone ever had a problem when you can call a phone, but you cant call from that phone? |
| 09:06.08 | AsteriskProblems | im fast beginning to think i should.... |
| 09:08.39 | *** part/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 09:09.22 | AsteriskProblems | do u know if there are any other groups on other irc networks that would help with asterisknow ? |
| 09:10.21 | The_LightSide | *sigh* Strom_M, now youve made me want too :/ |
| 09:12.51 | *** join/#asterisk kv0s (n=kv0s@p4FD27E54.dip.t-dialin.net) |
| 09:12.52 | kv0s | Hi! |
| 09:12.54 | *** join/#asterisk Turt|e (n=a@80.196.52.186) |
| 09:14.07 | kv0s | One zapata hfc-s problem ... :-( I've connected my asterisk through one bri-interface to my providers isdn-network. my incoming calls answered all very well. All works perfectly ... but one ... incoming faxes should be answered from another machine, that nothing to do with my asterisk, but uses the same ntba isdn bri ... |
| 09:14.12 | Turt|e | When a user dials my asterisk server the tone that appears in the callers phone is not the regional tone, im from denmark and would like the danish tone, is this a asterisk feature, or is it in my ip phone ? |
| 09:14.28 | kv0s | ... is there any chance to say asterisk it don't answer calls for a speciall msn? |
| 09:14.46 | *** join/#asterisk dg (i=dgl@otherwize.co.uk) |
| 09:15.41 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:22.08 | tzafrir | kv0s, you want asterisk not to answer all calls? |
| 09:22.24 | tzafrir | simple - confige it in the incoming context |
| 09:24.05 | kv0s | tzafrir: Yes. I've different numbers on the same ntba /isdn-bri/. Asterisk shouldn't answer calls to my number 8365838 - but it do! :-( |
| 09:24.39 | tzafrir | kv0s, does the same context has the extension _X. or s ? |
| 09:26.32 | kv0s | tzafrir: I don't have any context for the number shouldn't answered. |
| 09:26.52 | tzafrir | You have some context for the channel |
| 09:26.58 | kv0s | Yes. |
| 09:26.58 | tzafrir | context= in zapata.conf |
| 09:27.06 | kv0s | Ok. One Moment. |
| 09:27.14 | tzafrir | zap show channels |
| 09:27.24 | tzafrir | will show the context |
| 09:28.12 | kv0s | tzafrir see http://pastebin.com/d2c61c4b9 |
| 09:28.39 | tzafrir | show dialplan from-pstn |
| 09:29.07 | kv0s | tzafrir update -> http://pastebin.com/d47abf419 |
| 09:30.03 | kv0s | tzafrir update -> http://pastebin.com/d20606e54 |
| 09:30.15 | kv0s | can i exclude the number at zapata.conf? |
| 09:30.32 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:31.31 | *** join/#asterisk saftsack (n=saftsack@p57A74761.dip.t-dialin.net) |
| 09:36.47 | kv0s | tzafrir? |
| 09:36.55 | tzafrir | kv0s, hang on: it's from-pstn of from-isdn ? |
| 09:37.23 | tzafrir | from-pstn mostly includes other extensions . Is it freepbx? |
| 09:37.38 | kv0s | from-pstn is the context which used for my external connection ... from-isdn is used for my internal isdn-s0-bus. |
| 09:37.53 | kv0s | i've installed asterisk, and a further days later freepbx ... |
| 09:39.38 | tzafrir | so: show dialplan from-isdn |
| 09:40.26 | kv0s | http://pastebin.com/d1f4f6755 |
| 09:40.44 | kv0s | actually not used! at the moment i'm only using internal sip-softphones |
| 09:53.29 | alin` | ,ping |
| 09:53.51 | *** part/#asterisk alin` (n=user@193.226.173.50) |
| 09:58.58 | *** join/#asterisk Woifi1988 (n=anon@M1524P011.adsl.highway.telekom.at) |
| 09:59.04 | Woifi1988 | morning |
| 10:00.34 | Woifi1988 | is it possible to call a number with a certain extension from an isdn line and then manage that this extension should be connected to sip/2000??? |
| 10:01.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:02.39 | tzafrir | kv0s, generally configure this in freepbx. from-isdn also include frm-pstn |
| 10:03.06 | tzafrir | so just remove the catch-all trunk and add one(s) with an explicit DID |
| 10:08.01 | Woifi1988 | is it possible to call a number with a certain extension from an isdn line and then manage that this extension should be connected to sip/2000??? |
| 10:10.40 | Wonka | *sigh* |
| 10:10.52 | Wonka | i try to Monitor() a call... |
| 10:11.15 | Wonka | Monitor(); Dial(${ext},,twW); |
| 10:11.43 | Wonka | and * tells "Packet2Packet bridging ... and ..." |
| 10:12.01 | Wonka | the wave files are about 2284 bytes big |
| 10:12.14 | Wonka | (read: nothing in there) |
| 10:12.32 | Wonka | both channels are sip channels with canreinvite=no |
| 10:12.41 | Wonka | any ideas? |
| 10:12.42 | *** join/#asterisk appelza (n=d@dsl-240-133-188.telkomadsl.co.za) |
| 10:13.37 | Wonka | it's * 1.4.11, btw |
| 10:14.14 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 10:18.59 | appelza | Hi, I have an ISDN and a digium card installed. Got SIP calls working, but strulling to get calls over the land line working, how can I confirm that both cards are 'detected' and in use? |
| 10:19.15 | appelza | (using latest asterisk/zaptel) |
| 10:22.59 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 10:23.00 | kippi | hey |
| 10:23.18 | kippi | is there a tool to check packet loss on a call though asterisk |
| 10:23.37 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 10:23.42 | appelza | anyone? |
| 10:25.28 | b11d | you know what the real question here is.. can you lock the asterisk console, so you can view the diag messages but not enter commands w/o a password? |
| 10:25.43 | *** join/#asterisk zepmantra (n=dsadsa@125.212.110.114) |
| 10:26.55 | AsteriskProblems | does anyone know if asterisk sends out its calls through an IAX provider on the same IP that it would receive them from the same provider? |
| 10:27.53 | zepmantra | hello there we have an TE205P connected to a Adtran Atlas 550 ... the problem is that it cannot detect a dtmf tone on the ivr prompt, we have tried to use a tdm400p analog digium card and the dtmf detection works nicely , we have tried wiki page on editing dsp.c and relaxdtmf=yes and toneduration=300 still no luck ..... |
| 10:34.55 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:36.28 | heelios | AsteriskProblems: that.. really depends on your setup. |
| 10:37.00 | AsteriskProblems | oh |
| 10:37.05 | kippi | is there a tool to check packet loss on a call though asterisk |
| 10:37.22 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-56d9ded753782614) |
| 10:42.25 | Wonka | soxmix: Unknown output file format for '20070829122718-1188383238.77-04-s.wav': File type '77-04-s.wav' is not known |
| 10:42.29 | Wonka | <- crying |
| 10:43.20 | JunK-Y | Wonka: use -t |
| 10:44.38 | AsteriskProblems | well ive done it - ive got asterisknow working, using a combination of the GUI and standard asterisk file configurations, its a bit of a hodge podge but it works |
| 10:44.59 | AsteriskProblems | asterisknow is definately not anywhere near working right :P |
| 10:47.17 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 10:48.15 | deegan | I have a phone menu with some GotoIfTime statements. Now, i have one choice that's available mon-fri 08:00-16:00 and if i want that same choice to be available at sat x-x and sun x-x how would i go about doing that? can i just add gotoiftime's after another? |
| 10:49.08 | shinao1 | hi i want to configure an asterisk system and im a bit stumped on what im sure is a trivial issue. I need to Integrate * with a legacy Panasonic PBX. i intend to buy an 8 port digium card (4FXO, 4FXS) and plug the Panasonic's CO ports (1-4) to the digium FXS ports, and whatever PSTN ports to the digium FXO ports. I have a seen a post for integration of this sort in an asterisk forum online, but my issue is with the Pana PBX (a 616 with 16 extensions an |
| 10:49.08 | shinao1 | d 6 COs). The Pana has never been connected to the PSTN. ever. I want to ask, do i have to put any phone number(s)/extension number(s) on the Panasonic's CO port(s) to get them to work? i dont know if this makes any sense to any one.. |
| 10:52.23 | FlatFoot | hello all .. just about to build a new * been given a Dual Intel Server board . What would be the best choice of OS to make full use of both procs ??????? |
| 10:52.28 | zepmantra | <- crying |
| 10:52.31 | zepmantra | hello there we have an TE205P connected to a Adtran Atlas 550 ... the problem is that it cannot detect a dtmf tone on the ivr prompt, we have tried to use a tdm400p analog digium card and the dtmf detection works nicely , we have tried wiki page on editing dsp.c and relaxdtmf=yes and toneduration=300 still no luck ..... |
| 10:53.51 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 10:55.38 | McDouglas | is there a way to debug a zap channel? (like adding debug=4 to misdn.conf) |
| 10:56.41 | b11d | i wish |
| 10:56.48 | b11d | i know pri debug, and sip debug.. |
| 10:56.50 | b11d | but no zap debug |
| 10:57.08 | JunK-Y | u cant debug zap, only pri. |
| 10:57.30 | McDouglas | how do i know what happens then? :P |
| 10:57.37 | b11d | zepmantra.. do you get ANY DTMF at all? |
| 10:57.48 | b11d | mcd.. set verbose 100 and set debug 100 i guess.. |
| 10:57.50 | b11d | and look :) |
| 10:58.13 | b11d | whats with that tool "ztmonitor" anyways? is that for zap debugging? |
| 10:58.20 | zepmantra | b11d not a single one, no output movement from CLI |
| 10:58.53 | b11d | your indicatons file is correct right? right country code set for your tones, etc? |
| 11:00.05 | b11d | what ver of asterisk are you running? |
| 11:00.56 | b11d | fuck i've been up for 34 hours fixing the campus email system.. im too tired.. must go home now. |
| 11:01.04 | b11d | good luck with that zapmantra.. |
| 11:01.06 | b11d | ttyl all |
| 11:02.13 | tzafrir | McDouglas, there are a number of methods. What level exactly do you want to debug? |
| 11:02.45 | tzafrir | Simply increasing the debug level of asterisk gives you some excessive information in the debug log |
| 11:02.52 | tzafrir | which can be handy |
| 11:03.22 | zepmantra | ok thanks i forgot the indications thinggy.. |
| 11:03.23 | tzafrir | ztmonitor is also a very handy tool indeed |
| 11:03.31 | McDouglas | well, i got a tricky problem |
| 11:03.37 | tzafrir | and for PRI: pri debug span NNN |
| 11:03.55 | McDouglas | i have a tdm400 with an analog line and an analog phone connected |
| 11:04.07 | McDouglas | works like a charm, can dial out and can dial in |
| 11:04.24 | McDouglas | if i replace the analog land line with one from my GSM fixed wireless temirnal |
| 11:04.32 | McDouglas | i can not dial from the analog phone |
| 11:04.40 | McDouglas | BUT can dial from a soft-sip phone!! |
| 11:04.52 | McDouglas | the analog phone just gets a busy signal |
| 11:05.05 | McDouglas | <PROTECTED> |
| 11:05.06 | McDouglas | <PROTECTED> |
| 11:05.06 | McDouglas | asterisk*CLI> |
| 11:05.11 | McDouglas | and this is all i get |
| 11:05.26 | McDouglas | if i dial from the sip phone: |
| 11:05.27 | McDouglas | <PROTECTED> |
| 11:05.27 | McDouglas | <PROTECTED> |
| 11:05.27 | McDouglas | <PROTECTED> |
| 11:05.27 | McDouglas | <PROTECTED> |
| 11:05.27 | McDouglas | <PROTECTED> |
| 11:05.32 | McDouglas | strange :\ |
| 11:07.00 | tzafrir | dial out from an analog phone (connected to an FXS port of the card) to a device connected to an FXO port of the card? |
| 11:07.19 | tzafrir | McDouglas, please use a pastebin |
| 11:08.21 | tzafrir | Zap/1 is the FXS port? |
| 11:08.59 | tzafrir | McDouglas, smells like a dialplan issue |
| 11:09.59 | tzafrir | please pastebin: zap show channels' and the dialplan context of the FXS port (zap channel 1) |
| 11:16.20 | McDouglas | tzafrir: sry had something to do |
| 11:16.30 | McDouglas | yes zap/1 is an fxs module |
| 11:16.37 | McDouglas | phone connected to that |
| 11:17.20 | McDouglas | http://pastebin.com/d135c1ebd |
| 11:17.48 | McDouglas | my extension.conf http://pastebin.com/d2b09e70 |
| 11:18.15 | McDouglas | tzafrir: why can i dial then if i replace the gsm line with land line? |
| 11:18.35 | McDouglas | (also reduce the Xs in the dialplan) |
| 11:19.26 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.51) |
| 11:24.06 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:25.02 | florz | is it a feature or a bug that the thread debugging code is not thread safe? |
| 11:26.25 | McDouglas | tzafrir: any idea? |
| 11:26.33 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:32.16 | *** join/#asterisk Aeudian (n=Aeudian@c-69-250-24-154.hsd1.md.comcast.net) |
| 11:32.47 | Aeudian | I am recieving a warning and notify upon a reload for "No SMDI interfaces were specified to listen on, not starting SDMI listener." "indications.c:505 ast_unregister_indication_country: Removed default indication country 'us'" What does this mean? Can you stop it or correct it? |
| 11:32.50 | tzafrir | sorry, busy |
| 11:34.56 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:35.34 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.170.212) |
| 11:42.39 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 11:42.39 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 11:45.04 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.54.22) |
| 11:47.21 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:54.16 | *** join/#asterisk coppice (n=chatzill@83.155.17.210.dyn.pacific.net.hk) |
| 11:55.06 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 11:57.50 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 12:04.50 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 12:04.51 | *** join/#asterisk saftsack (n=oliver@p54A7FBEE.dip.t-dialin.net) |
| 12:04.56 | saftsack | hi |
| 12:04.58 | saftsack | <PROTECTED> |
| 12:04.58 | saftsack | Segmentation fault |
| 12:06.57 | saftsack | the best way to solve this would be the creation of a coredump? |
| 12:07.55 | *** part/#asterisk jfitzgibbon (n=jfitzgib@64.72.237.130) |
| 12:09.54 | jsmith | saftsack: Yes. |
| 12:11.30 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:14.14 | Wonka | i want to start Monitor()ing calls to a queue when an Agent takes the call - any ideas? |
| 12:14.19 | saftsack | asterisk -vvvvvvvvgc |
| 12:14.26 | saftsack | then it creates a coredump? |
| 12:14.55 | saftsack | http://www.voip-info.org/wiki-Asterisk+debugging here they are just taking from safe_asterisk |
| 12:14.55 | [TK]D-Fender | Wonka: Yes, read the SAMPLE queues.conf config which shows you the parameters to set to do automatic queue recording. |
| 12:14.56 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 12:15.24 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 12:16.45 | JoelSolanki | Hi anybody recommend me e911 services workable on asterisk. |
| 12:17.15 | jsmith | saftsack: Yes, it should. |
| 12:17.34 | Wonka | [TK]D-Fender: thanks - seems i just skipped over that |
| 12:17.36 | jsmith | saftsack: It would hurt to type "ulimit -c unlimited" first before starting Asterisk |
| 12:18.26 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:19.18 | saftsack | did this. also i did starting with the g option but at all i cannot find a coredump file somewhere in my pc |
| 12:19.27 | saftsack | (it is an openwrt router) |
| 12:20.30 | jsmith | saftsack: Ah, it may not produce a core file then... you might look in /tmp if you're using the safe_asterisk script to start Asterisk |
| 12:20.45 | jsmith | saftsack: Otherwise, it'll probably be in the directory you started Asterisk from |
| 12:21.12 | saftsack | safe_asterisk doesnt work because i have just one tty |
| 12:21.22 | saftsack | no .... no coredumpfile at all |
| 12:21.24 | jsmith | saftsack: You can disable the tty in safe_asterisk |
| 12:22.27 | *** join/#asterisk rata (n=rcampos@princed/developer/rata) |
| 12:22.28 | s0ck | anyone having trouble with sipgate |
| 12:22.31 | rata | hi |
| 12:22.38 | Daviey | s0ck: not today |
| 12:22.44 | Daviey | sipgate.co.uk? |
| 12:23.15 | saftsack | jsmith, ok got it with safe_asterisk but no core file at all |
| 12:24.32 | jsmith | saftsack: It's kind of hard to debug without a core file, unfortunately |
| 12:24.45 | saftsack | thats true so i want to have this core file by now :( |
| 12:25.19 | s0ck | hmm |
| 12:25.21 | s0ck | changed box, can't seem to register |
| 12:26.04 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 12:28.46 | saftsack | OPTIONS="-DLOW_MEMORY -Dlinux" \ |
| 12:28.58 | robl^ | someone messed up the wiki! |
| 12:29.01 | saftsack | is this the killer for the debug option? |
| 12:38.40 | *** join/#asterisk jfitzgibbon (n=jfitzgib@64.72.237.130) |
| 12:38.55 | *** part/#asterisk jfitzgibbon (n=jfitzgib@64.72.237.130) |
| 12:41.07 | *** join/#asterisk hum711017 (n=humberto@200.55.132.205) |
| 12:41.51 | *** join/#asterisk lbow (n=lbow@dsl-241-31-00.telkomadsl.co.za) |
| 12:48.30 | *** join/#asterisk jfitzgibbon (n=NADT@64.72.237.130) |
| 12:49.41 | Wonka | [TK]D-Fender: i need to be able to do this automatic queue recording only if the caller has allowed it - my tries with Set(monitor-type=wav); have failed. any ideas? |
| 12:49.49 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
| 12:50.52 | [TK]D-Fender | Wonka: Easiest hack is to just make 2 queues with identical members & priority. One with recording, one without. |
| 12:53.07 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:54.07 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:54.31 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 12:55.02 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 12:55.02 | *** mode/#asterisk [+o file] by ChanServ |
| 12:58.56 | *** join/#asterisk dominei (n=nada@198-203-29-194.mtulink.net) |
| 12:59.03 | dominei | inovaphone..asterisk based |
| 13:00.25 | dominei | ? :) |
| 13:05.27 | DrAk0 | why i don't have `misdn` commands on asterisk but chan_misdn is loaded? |
| 13:06.37 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:06.53 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:07.19 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:08.18 | deeperror | when dialing out iax2 when the callee hangs up I hear fast busy. any idea what is throwing that or how to make it silence? |
| 13:08.54 | deeperror | i don't hear a busy signal when making outbound sip calls |
| 13:09.26 | Wonka | [TK]D-Fender: hehe. ugly, but works... |
| 13:09.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.06 | [TK]D-Fender | Wonka: I think you could also just variably put a Monitor before your call to Queue. |
| 13:10.29 | DrAk0 | what do i need to get misdn commands on the cli ? |
| 13:13.41 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 13:19.46 | DrAk0 | nvm |
| 13:19.47 | DrAk0 | i got it |
| 13:25.04 | Katty | datachomper: :> |
| 13:25.25 | Katty | i have a client that keeps emailing me 4 times a day asking stupid questions about their new server. |
| 13:25.45 | Katty | what does this silly message mean in event viewer, how do i get to x, where is y, how come i can't seem to get z to work like i want it. |
| 13:25.58 | Katty | and it's taking me at least 2 hours a day to do research and get back with her where i spend another hour on the phone |
| 13:26.02 | Katty | Grrr!! |
| 13:26.06 | Katty | and they won't bill my time. |
| 13:26.16 | Katty | meanwhile, my things keep stacking up and they wonder why. |
| 13:26.30 | s0ck | i dont seem to be getting anything back from a register command |
| 13:26.52 | s0ck | sip debug doesn't show any response |
| 13:26.55 | s0ck | can ping the sip gateway |
| 13:27.12 | s0ck | box is nat'd but shouldn't stop registration? |
| 13:27.56 | McDouglas | i put the sample cdr.conf into /etc/asterisk but it still does not log the dials |
| 13:27.59 | McDouglas | whats the problem? |
| 13:32.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:32.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:32.59 | Katty | anthm: :> |
| 13:33.22 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-31-215.w81-251.abo.wanadoo.fr) |
| 13:34.25 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 13:34.54 | ManxPower | For one thing, the sample config files are never met to be used, they are meant as an example of every available option |
| 13:35.20 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-4b719fc3490b6e9b) |
| 13:37.34 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 13:38.56 | jsmith | ManxPower: Amen, brother! |
| 13:40.00 | McDouglas | well, of course i read it and the comments |
| 13:40.07 | McDouglas | it should work |
| 13:41.12 | datachomper | Katty, send /them/ a bill for your time |
| 13:41.25 | [TK]D-Fender | McDouglas: perhaps you sould do a reload and see what module is loading for your CDRs..... |
| 13:43.23 | *** part/#asterisk rody (i=netstati@neptune.negativeblue.com) |
| 13:44.37 | McDouglas | <PROTECTED> |
| 13:44.37 | McDouglas | <PROTECTED> |
| 13:44.46 | McDouglas | do i need anything else beside this? |
| 13:46.16 | [TK]D-Fender | McDouglas: Ok, go show us your log file / folder |
| 13:46.20 | [TK]D-Fender | ~pb |
| 13:46.20 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 13:46.21 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 13:46.48 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 13:48.08 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:48.16 | [TK]D-Fender | ouch |
| 13:48.33 | McDouglas | well, there is nothing in log/asterisk/cdr-csv |
| 13:50.25 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:50.48 | [TK]D-Fender | McDouglas: And what about cdr-custom? |
| 13:51.45 | McDouglas | empty |
| 13:51.59 | McDouglas | content of cdr.conf: |
| 13:52.24 | McDouglas | [csv] |
| 13:52.24 | McDouglas | usegmtime=yes ; log date/time in GMT. Default is "no" |
| 13:52.24 | McDouglas | loguniqueid=yes ; log uniqueid. Default is "no |
| 13:52.24 | McDouglas | loguserfield=yes ; log user field. Default is "no |
| 13:53.34 | [TK]D-Fender | McDouglas: Copy the sample file, and simply rename it. |
| 13:54.16 | [TK]D-Fender | McDouglas: And then restart * and test. Check both of those folders for Master.csv |
| 13:54.34 | [TK]D-Fender | McDouglas: Before placing a test call, PB up the full module loadup output |
| 13:55.55 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 13:55.55 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 13:56.07 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:56.07 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:56.17 | McDouglas | ohh |
| 13:56.26 | McDouglas | it does work after restarting asterisk |
| 13:56.34 | McDouglas | module reload isnt enough then? |
| 13:57.33 | [TK]D-Fender | McDouglas: Apparently NOT :) |
| 14:00.45 | McDouglas | now, only one problem remains |
| 14:00.57 | McDouglas | why can i dial a number from a SIP phone and not from analog phone? |
| 14:00.59 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 14:01.00 | Yourname` | Hi, how can I use QUEUE_LOG to do this: Basically, I currently have an IVR. Where the first call goes out (callerid log), and then a message is played to a human (log it) . Hangs up if it's AMD (log it). Human presses 1 (log it), human hangs up (log it). After human presses 1, there's another message..and the human needs to press 1 again (log it) or hangs up (log it) .. then human waits in... |
| 14:01.01 | Yourname` | ...the queue (time waited log) before he/she gets to an agent (log agent username). Then log the talk time, and the total time from start to finish. |
| 14:01.14 | [TK]D-Fender | McDouglas: DETAILS might help. |
| 14:01.26 | [TK]D-Fender | McDouglas: And remember, PASTEBIN is your friend |
| 14:01.50 | McDouglas | http://pastebin.com/d79f5c530 |
| 14:02.08 | McDouglas | details: http://forums.digium.com/viewtopic.php?t=17743 |
| 14:02.28 | [TK]D-Fender | Yourname`: add commands to log this extra dialplan stuff IN your dialplan. |
| 14:02.47 | Yourname` | [TK]D-Fender: Going right to QUEUE_LOG? |
| 14:03.11 | *** join/#asterisk AsteriskProblems (n=pbarnsle@81.171.174.178) |
| 14:03.30 | AsteriskProblems | anyone ever had a problem where they can dial an extension, but they cant dial from that extension? |
| 14:04.56 | [TK]D-Fender | McDouglas: pastebin a failed call at verbose 10 |
| 14:05.08 | [TK]D-Fender | Yourname`: If you think thats an appropriate place. |
| 14:05.19 | Yourname` | [TK]D-Fender: Great, thank you, monsieur. |
| 14:05.30 | marc7 | AsteriskProblems: are you still having problems with asterisknow? :) |
| 14:06.01 | McDouglas | [TK]D-Fender: http://pastebin.com/d4ab224f0 |
| 14:06.09 | jsmith | AsteriskProblems: Sure... sounds like a problem with your contexts... are the two phones pointed at different contexts? |
| 14:06.16 | McDouglas | started the call from the analog line at line 17 |
| 14:06.29 | [TK]D-Fender | AsteriskProblems: You can't dial FROM an extension. An extension is a number, and numbers can't dial. DEVICES can dial. As for why you can't dial from your DEVICE, well that's most likely due to either * being mis-configured on where to send its calls. |
| 14:06.34 | McDouglas | made a successful call to the same number at line 12 with a sip phone |
| 14:06.47 | JT | AsteriskProblems: also, asterisk-gui is not supported here |
| 14:06.52 | [TK]D-Fender | McDouglas: Next! |
| 14:06.57 | s0ck | [TK]D-Fender: sip register showing 120 - request sent |
| 14:06.57 | *** join/#asterisk mog (i=mog@nat/digium/x-0ae2768876fc72db) |
| 14:06.57 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:07.06 | s0ck | sip debug shows nothing coming back from the sip gateway, ideas? |
| 14:07.21 | AsteriskProblems | jsmith - they are pointed at the same context, |
| 14:07.22 | *** join/#asterisk evangelion (n=manzy_ze@62.123.91.227) |
| 14:07.27 | McDouglas | [TK]D-Fender: next? |
| 14:07.28 | [TK]D-Fender | s0ck: Maybe you're not sending your register request to the right place <- |
| 14:07.35 | s0ck | i am |
| 14:07.36 | AsteriskProblems | whats the command to set maximum debug level on? |
| 14:07.38 | [TK]D-Fender | McDouglas: Now show me the FAILED call. |
| 14:07.45 | McDouglas | its there |
| 14:07.47 | [TK]D-Fender | s0ck: Show us something useful |
| 14:07.49 | McDouglas | at line 17 |
| 14:07.58 | evangelion | hello |
| 14:08.08 | kippi | hey |
| 14:08.13 | McDouglas | i picked up the phone and dialed |
| 14:08.24 | kippi | has anyone had any problems using SIP or RIP1? |
| 14:08.34 | [TK]D-Fender | McDouglas: And you get dialtone on your analog line? |
| 14:08.46 | AsteriskProblems | all the stuff ive done to my asterisknow install has been pure asterisk edits, and they have worked, so that is why i am here - plus no-one talks in the asterisknow channel :P |
| 14:08.47 | s0ck | this is the only useful thing i can find: Contact: <sip:1041780@192.168.1.222> |
| 14:08.49 | McDouglas | yes there is dialtone, i can dial any internal extension |
| 14:08.53 | [TK]D-Fender | McDouglas: Does it go bad after your first digit? |
| 14:09.07 | s0ck | box is nat'd but im lead to believe it should at least register |
| 14:09.22 | McDouglas | no, after the first digit the tone stops |
| 14:09.26 | McDouglas | i enter the full number |
| 14:09.33 | McDouglas | then after a while i get busy signal |
| 14:09.49 | jsmith | AsteriskProblems: Can you turn on some additional debugging to see why it's failing from the second phone? |
| 14:09.59 | [TK]D-Fender | McDouglas: increase your debug on CLI and retry |
| 14:10.05 | AsteriskProblems | yeh - whats the command for maximum debug? |
| 14:10.08 | McDouglas | set debug 10 ? |
| 14:10.17 | [TK]D-Fender | McDouglas: Sure, why not... |
| 14:10.20 | jsmith | core set debug 99 |
| 14:10.23 | AsteriskProblems | ok thanks |
| 14:10.36 | [TK]D-Fender | s0ck: Congratulations, you aren't sending the right RETURN address. |
| 14:10.43 | *** join/#asterisk the_esc (n=the_esc@adsl-76-222-205-102.dsl.ksc2mo.sbcglobal.net) |
| 14:10.56 | [TK]D-Fender | s0ck: Like calling someone, leaving a voicemail and telling the WRONG NUMBER to get back to you... |
| 14:11.05 | [TK]D-Fender | s0ck: Go read this now : |
| 14:11.07 | [TK]D-Fender | ~sipnat |
| 14:11.07 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:11.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 14:11.20 | [TK]D-Fender | ARE* |
| 14:11.51 | s0ck | i *may* have put externip in the wrong place |
| 14:11.52 | AsteriskProblems | "really destroying sip dialog... ACK / Register" is all im seeing |
| 14:12.05 | AsteriskProblems | when i dial from the phone literraly nothing shows on the asterisk box |
| 14:12.33 | AsteriskProblems | the phone has registered its account - i saw that show on the log, but when u dial its as tho it doesnt see the phones request |
| 14:13.02 | AsteriskProblems | i can call that phone from a softphone with no problem though |
| 14:13.02 | JT | sip debug |
| 14:13.07 | McDouglas | [TK]D-Fender: same output, even after debug 99 |
| 14:13.18 | Yourname` | In Ast 1.4, it would be Set(QUEUE_PRIO=10), right? Or SetVar(QUEUE_PRIO=10)? |
| 14:13.30 | JT | Set |
| 14:13.32 | JT | even in 1.2 |
| 14:13.32 | [TK]D-Fender | McDouglas: Show me a good call, then a bad one (intern, then ext, ALL zap) |
| 14:13.48 | [TK]D-Fender | s0ck: "May"? How about CERTAINLY? |
| 14:14.18 | McDouglas | i cant make a good external call from zap, but gonna make the others |
| 14:16.11 | *** join/#asterisk redbaron1973 (n=redbaron@host52-73.birch.net) |
| 14:16.15 | Yourname` | Thanks JT. |
| 14:16.26 | tzafrir | what was the issue, exactly? |
| 14:16.32 | redbaron1973 | utils.c: In function `vpoe_slprintf': |
| 14:16.33 | redbaron1973 | utils.c:186: error: invalid use of non-lvalue array |
| 14:16.37 | tzafrir | Do you actually *need* ppp with Zaptel? |
| 14:16.55 | redbaron1973 | I am trying to use a T411 with 4 T1's bonded from my isp for data |
| 14:17.05 | redbaron1973 | they claim to use MPPP |
| 14:18.10 | s0ck | [TK]D-Fender: :D |
| 14:18.14 | [TK]D-Fender | McDouglas: I want to see the good INTERNAL call then a bad external attempt immediately following |
| 14:18.15 | s0ck | my contact shizzle is looking good now |
| 14:18.17 | s0ck | still no reg |
| 14:18.33 | [TK]D-Fender | s0ck: SIP debug + pastebin <- |
| 14:18.41 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:18.50 | s0ck | sip debug shows nothing back from the itsp |
| 14:18.53 | s0ck | zero |
| 14:18.55 | flujan | morning all. |
| 14:18.57 | flujan | :) |
| 14:19.31 | [TK]D-Fender | s0ck: I want to see the OUTGOING register as well. |
| 14:19.41 | redbaron1973 | I haven't tried to compile zaptel yet with ppp enabled, but according to the docs, I need to get the patched ppp in place first |
| 14:19.43 | [TK]D-Fender | flujan: Good morning |
| 14:21.45 | tzafrir | Actually, I never tried to patch it |
| 14:21.56 | flujan | [TK]D-Fender: I think I have a memory leak on asterisk ... |
| 14:22.13 | flujan | I create a log of all actions and yesterday the bug appear again. |
| 14:22.24 | flujan | http://pastebin.com/df1a0945 |
| 14:22.37 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 14:22.44 | flujan | according to this error messages it is a problem allocating memory. |
| 14:22.45 | McDouglas | [TK]D-Fender: strange: i plugged in another analog telephone and made an extension and called: same error, then removed exten => _XXXXXXXXXXX,1,Dial(Zap/3/${EXTEN}) from my config (http://pastebin.com/d73189289) and i could make an internal call |
| 14:22.50 | redbaron1973 | Would I need hdlc for this? the provider says they use MPPP, so I assume that is all I would need |
| 14:22.55 | redbaron1973 | is just ppp |
| 14:22.59 | mattfletcher | hello, is it possible to force the use of ulaw for one particular outgoing extension |
| 14:23.34 | JT | if it has its own sip.conf entry, sure |
| 14:24.34 | mattfletcher | JT: it doesn't sadly. can it not be set within the dialplan? |
| 14:24.56 | dominei | is innovaphone asterisk? |
| 14:24.56 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 14:25.26 | tzafrir | redbaron1973, what distro? |
| 14:25.29 | [TK]D-Fender | mattfletcher: Go MAKE one for it. |
| 14:25.47 | redbaron1973 | centos 4.3 /c 2.6.20.2 kernel custom (make oldconfig) |
| 14:26.34 | mattfletcher | [TK]D-Fender: this is to dial a person using a 123456789@sip.gradwell.net syntax. not too sure how to define that in sip.conf. Are there examples out there? |
| 14:26.36 | flujan | the memory status: http://pastebin.com/d1550837a |
| 14:27.46 | s0ck | [TK]D-Fender: http://pastebin.ca/674749 |
| 14:27.56 | [TK]D-Fender | mattfletcher: Just make another peer like you did the first time. |
| 14:28.37 | mattfletcher | [TK]D-Fender: but this peer will not be registering or authenticating or anything with me, it's a remote sip phone hosted by gradwell |
| 14:29.33 | [TK]D-Fender | s0ck: Maybe they're down, and maybe you're sending it to the wrong place. |
| 14:29.41 | Yourname` | If QUEUE_PRIO is set for only maybe 1 extension out of 3-4, what will the priority be for the ones that don't have QUEUE_PRIO set? |
| 14:29.54 | s0ck | website clearly says sipgate.co.uk |
| 14:29.56 | s0ck | checked twice |
| 14:30.06 | [TK]D-Fender | mattfletcher: DOESN"T MATTER. You dial out of a a peer, and this has nothing to do with registering. |
| 14:30.12 | s0ck | im just wondering if firewall is blocking 5060 back in |
| 14:30.25 | [TK]D-Fender | s0ck: apstebin your configs, and link me to them. |
| 14:30.43 | [TK]D-Fender | s0ck: Well I think you should go CHECK, not shouldn't you? |
| 14:30.47 | [TK]D-Fender | now* |
| 14:30.56 | McDouglas | [TK]D-Fender: any idea about what i wrote? |
| 14:31.00 | *** part/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 14:31.05 | s0ck | i've added a global rule from sipgate to that box |
| 14:31.16 | s0ck | it SHOULD allow everything from that ip |
| 14:31.31 | [TK]D-Fender | McDouglas: Wheres that "good" followed by "bad" call CLI output I asked for? |
| 14:31.57 | McDouglas | [TK]D-Fender: strange: i plugged in another analog telephone and made an extension and called: same error, then removed exten => _XXXXXXXXXXX,1,Dial(Zap/3/${EXTEN}) from my config (http://pastebin.com/d73189289) and i could make an internal call |
| 14:32.10 | s0ck | i notice it tries nat and no nat in that pastebin |
| 14:32.30 | [TK]D-Fender | s0ck: You're assuming you know WHERE the response is coming from. And more poeple think they're "smart" about their firewall's usability, usually FARTER from the trusth they are. |
| 14:32.52 | s0ck | yeh |
| 14:32.53 | [TK]D-Fender | s0ck: Naturally my faith in your setup approaches the "0" mark... |
| 14:33.01 | s0ck | think you hit the nail on the head there |
| 14:33.05 | s0ck | could be coming from a diff ip ;/ |
| 14:33.40 | s0ck | i dunno about farting, seems over the top? |
| 14:34.06 | [TK]D-Fender | McDouglas: do a "dialplan show", pastebin it, and show me the output I asked for |
| 14:34.41 | [TK]D-Fender | s0ck: "farther from the truth" <- So I'm typing sloppy today... BIG DEAL :p |
| 14:35.39 | mattfletcher | [TK]D-Fender: no need to shout at me. did you get out of bed the wrong side this morning? we can't all be experts you know |
| 14:36.21 | McDouglas | [TK]D-Fender: if i put back that line i can't call internal either. If i take it out i can. thats all |
| 14:36.22 | [TK]D-Fender | mattfletcher: Not yelling, that's be the WHOLE think in caps. Thats just "point reenfocement" :p |
| 14:36.22 | s0ck | ;) |
| 14:36.38 | McDouglas | however if i take that line out i dont know how to make an external call.. |
| 14:36.43 | s0ck | mattfletcher: you will soon get used to tkd, if you want any help that is :P |
| 14:37.06 | [TK]D-Fender | thing* <- IRC dyslexia in full swing today. |
| 14:37.13 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 14:37.40 | s0ck | just tried another sip gateway and BAM |
| 14:37.43 | s0ck | it reg'd immediately |
| 14:37.49 | s0ck | sipgate is clearly replying on a diff ip |
| 14:39.04 | [TK]D-Fender | s0ck: or just broken, etc.... |
| 14:39.10 | *** join/#asterisk anthony[ (n=anthony@fl-71-49-118-147.dhcp.embarqhsd.net) |
| 14:39.19 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 14:39.21 | s0ck | could be |
| 14:39.34 | AsteriskProblems | right ive done a sip debug and tried placing a call from my faulty extension and now have a pastebin, i can see the error, but i dont know what it means - could someone cast their eye over it please? I believe the error is on line 93 |
| 14:39.40 | AsteriskProblems | http://pastebin.com/m13dfa83e |
| 14:40.00 | FlatFoot | OK who's ready for a daft question ????? Debian V 4.0 just can't get asterisk to start on boot . Keeps moaning about asterisk.ctl ( which does exist even though it says it does not ) HOW do i get it to work ??? |
| 14:40.32 | AsteriskProblems | FlatFoot, ensure your permissions are ok |
| 14:40.46 | AsteriskProblems | http://forums.digium.com/viewtopic.php?t=12559 |
| 14:41.03 | FlatFoot | AsteriskProblems: permissions on .ctl ??? |
| 14:41.15 | AsteriskProblems | i had the same problem when ssh'ing into the box |
| 14:41.22 | [TK]D-Fender | AsteriskProblems: Found no matching peer or user for '10.0.1.62:1026' <---------- first, * can't ID you PHONE. It (or *) is misconfigured, or simply don't match |
| 14:41.29 | AsteriskProblems | im guessing its a similar issue - tho ive not seen your error :P |
| 14:41.48 | AsteriskProblems | ok right thanks fender |
| 14:41.57 | [TK]D-Fender | AsteriskProblems: Looking for 6001 in bogon-calls (domain 10.0.1.2) <----- its naturally falling back to the context in [general] and NOT the one I'm sure you'd LIKE it to sue |
| 14:42.16 | [TK]D-Fender | AsteriskProblems: SIP/2.0 404 Not Found <--- and clearly the exten you are dialing is not IN that context. |
| 14:42.27 | AsteriskProblems | right |
| 14:42.33 | [TK]D-Fender | AsteriskProblems: So go fix you SIP entry |
| 14:43.09 | redbaron1973 | tzafrir: the patched ppp is required to make the zaptel.sso..finaly found out why I was doing that part |
| 14:43.33 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:44.17 | mattfletcher | [TK]D-Fender: I still can't work out what I'm meant to put into sip.conf. I've never seen an example of it being used this. I have entries for sip hardphones listed here, and SIP providers (eg voipcheap), but i can't see how to simply list "a person" that I might wish to call. |
| 14:45.14 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:46.42 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:47.11 | McDouglas | [TK]D-Fender: here: http://pastebin.com/d56898ab5 |
| 14:47.38 | luke-jr | anyone want to help me crack RC4? |
| 14:52.20 | *** part/#asterisk rata (n=rcampos@princed/developer/rata) |
| 14:52.56 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:53.02 | [TK]D-Fender | mattfletcher: You don't list a PERSON at your ITSP, you simply make another peer to dial your ITSP with a different codec list. Your Dial is in your dialplan like always and uses this OTHER peer, and not your usual one. |
| 14:54.28 | [TK]D-Fender | McDouglas: As soon as you reach the last digit, do you see the Hangup INSTANTLY? |
| 14:54.44 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:57.03 | mattfletcher | [TK]D-Fender: got it. my confusion was that i'd assumed a secret was necessary, and couldn't work out what to put. it dials now, but still i can't hear the other end, while they do hear me |
| 14:57.45 | [TK]D-Fender | mattfletcher: Sounding like NAT issues..... |
| 14:58.01 | [TK]D-Fender | mattfletcher: part of your setup behind NAT? |
| 14:59.49 | AsteriskProblems | D-Fender: - when i call between two softphones the call goes through with no problems. I have even used the same user credentials on a softphone as the sip phone and it works fine, it is only when i try to call from my aastra phone that i get this error, so i tend to think the server is configured correctly? |
| 15:01.22 | kippi | anyone got anyideas on this? http://www.pastebin.ca/674783 |
| 15:02.57 | [TK]D-Fender | AsteriskProblems: well I guess that kinda points out what side is wrong, now dowsn't it? |
| 15:03.16 | AsteriskProblems | D-Fender: Yes, but im totally at a loss as to what i could be doing wrong with the phone |
| 15:03.27 | jsmith | kippi: That's Asterisk's way of saying "Hey, the other side didn't respond to me and I need him to... so I'm hanging up the call, as he's gone away" |
| 15:03.34 | [TK]D-Fender | AsteriskProblems: Sorry, but you've got to learn how to set it up. |
| 15:03.48 | _Raptor_ | what can be the reason when i get 603 declined when i try to transfer a call? (and nothing happens) |
| 15:04.00 | AsteriskProblems | D-Fender: one thing i noticed is that the aastra phone sends to do an invite command with port 5060 on the end, whereas the x-lite does not add a port on the end? |
| 15:04.36 | [TK]D-Fender | AsteriskProblems: asterisk can handle both devices just fine, its your configuration of it thats off |
| 15:05.05 | [TK]D-Fender | AsteriskProblems: Go provide some SIP debug of it rebooting, and trying to register, etc, and pastbin your sip.conf masking only passwords |
| 15:05.16 | AsteriskProblems | Ok |
| 15:05.34 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 15:11.26 | FlatFoot | AsteriskProblems: cheers done the perms thing works a treat now |
| 15:11.38 | kippi | jsmith: what can cause this? |
| 15:13.24 | jsmith | kippi: Network problems, firewall problems, a phone being unplugged from the network... anything that would keep the phone from being able to talk to Asterisk |
| 15:13.40 | kippi | ok, i will look into it more |
| 15:15.24 | *** join/#asterisk pepo-- (n=pepOSX@200.90.126.74) |
| 15:15.30 | AsteriskProblems | D-Fender: This is the pastebin of the phone rebooting http://pastebin.com/m22f9073c |
| 15:15.58 | *** join/#asterisk Woifi1988 (n=anon@M1321P022.adsl.highway.telekom.at) |
| 15:16.07 | Woifi1988 | hi |
| 15:17.06 | [TK]D-Fender | AsteriskProblems: SIP/2.0 401 Unauthorized <- your suer or pass doesn't match. Go fix this. |
| 15:17.31 | [TK]D-Fender | *user |
| 15:17.48 | AsteriskProblems | thats weird... thanks |
| 15:18.05 | [TK]D-Fender | AsteriskProblems: Not "wierd", its normal. Just wrong. |
| 15:18.25 | pepo-- | why this exten => 30,2,voicemailmain(su${EXTEN}) dont work? and i need put the mailbox and password too |
| 15:18.28 | Woifi1988 | does asterisk different between a offline user and a busy user? |
| 15:18.45 | *** part/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 15:19.47 | jsmith | Woifi1988: Yes... if you try to dial to a user and then look at the ${DIALSTATUS} channel variable, you'll see different values for busy users than for offline users |
| 15:20.14 | Woifi1988 | jsmith how can i look for this variable? |
| 15:20.18 | jsmith | pepo--: That's the old syntax... try 30,2,VoiceMailMain(30@default,su) (where "default" is the voicemail context that contains the mailbox numbered 30) |
| 15:20.32 | jsmith | Woifi1988: Something like this: |
| 15:20.35 | pepo-- | jsmith, good |
| 15:20.42 | jsmith | exten =>123,1,Dial(SIP/somebody,20) |
| 15:20.56 | jsmith | exten => 123,2,SayAlpha(${DIALSTATUS}) |
| 15:21.05 | *** join/#asterisk fasgaroth (n=fasgarot@166.pool80-103-163.dynamic.orange.es) |
| 15:21.10 | jsmith | exten => 123,3,Verbose(0,Dialstatus is ${DIALSTATUS}) |
| 15:21.24 | jsmith | That should give you a couple ideas to play with |
| 15:21.32 | Woifi1988 | thank you! |
| 15:22.01 | pepo-- | jsmith, but still wanna mailbox and password |
| 15:22.16 | *** join/#asterisk fasgaroth (n=fasgarot@166.pool80-103-163.dynamic.orange.es) |
| 15:22.30 | jsmith | pepo--: Oh, you want it to prompt for the mailbox? |
| 15:22.46 | *** part/#asterisk fasgaroth (n=fasgarot@166.pool80-103-163.dynamic.orange.es) |
| 15:22.48 | pepo-- | jsmith, just i wanna put password |
| 15:23.07 | jsmith | pepo--: That should work then, unless you don't have a mailbox named "30" in the default voicemail context |
| 15:23.18 | pepo-- | yes |
| 15:23.23 | jsmith | pepo--: Oh, the "s" tells it not to prompt for the password |
| 15:23.32 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:23.38 | jsmith | pepo--: Take the "s" out of the second paramter if you want it to prompt for the password |
| 15:23.47 | pepo-- | let see |
| 15:24.30 | *** join/#asterisk Galeras (n=Galeras@201.244.242.191) |
| 15:24.49 | datachomper | So in my Perl AGI, it seems to be skipping my first stream_file()? |
| 15:24.50 | AsteriskProblems | D-Fender: thanks for your help, the password is definately wrong but im getting the same error. At least it gives me somewhere to start looking thanks |
| 15:25.03 | AsteriskProblems | *password is definately right ;) |
| 15:25.34 | pepo-- | nothings but if i dont put mailbox then i just put the password and work, but i need wait 10 second |
| 15:25.47 | [TK]D-Fender | AsteriskProblems: You've likely filled in the wrong blanks. please pastebin your SIP entry for the phone. |
| 15:26.33 | AsteriskProblems | D-Fender: it doesnt have a sip entry as it is asterisk now - its in a file called users.conf, but as i said I can log in to softphones using the same account so it must be the phone |
| 15:26.51 | Woifi1988 | jsmith: How can I handle these states? I'd like to respond to an chanunavail state for example! |
| 15:27.26 | Lucky7 | dropped calls blow |
| 15:27.32 | [TK]D-Fender | AsteriskProblems: There you have it. Go read the manual again :p |
| 15:27.49 | jsmith | Woifi1988: With GotoIf()... something like GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?context,extension,priority) |
| 15:28.34 | [TK]D-Fender | AsteriskProblems: I'll take it ;) |
| 15:28.55 | Daviey | AsteriskProblems: i have a few aastra phines.. they rock |
| 15:29.02 | Daviey | but price artifically high |
| 15:29.14 | jsmith | Their support rocks, fwiw |
| 15:29.47 | AsteriskProblems | as long as their support is good thats all that matters :P |
| 15:29.54 | coppice | i think if you need support for a phone, the phone sucks. different point of view, I guess |
| 15:30.07 | darkfires | b11d|bbl:28:19.932071 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 60) 10.1.0.1.19896 > 192.168.42.19.16414: [udp sum ok] UDP, length 32 |
| 15:30.07 | darkfires | b11d|bbl:28:19.952071 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 60) 10.1.0.1.19896 > 192.168.42.19.16414: [udp sum ok] UDP, length 32 |
| 15:30.17 | darkfires | asterisk is flooding the pap2 |
| 15:30.19 | darkfires | any ideas? |
| 15:30.24 | Daviey | coppice: I was pleased that the receptionist side panel.. "just works"TM |
| 15:30.34 | Daviey | no asterisk hacking really needed |
| 15:30.39 | AsteriskProblems | well the crazy thing is the phone used to work fine, i must have screwed something somewhere but its not like there are that many options to mess up ! |
| 15:30.48 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 15:31.12 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 15:32.07 | Woifi1988 | jsmith: but i get an "auto-falltrough" |
| 15:32.29 | Daviey | aastra default ringer sets are better than most |
| 15:32.47 | [TK]D-Fender | Polycom > ALL |
| 15:33.14 | Galeras | snom? |
| 15:33.34 | Daviey | Galeras: Who? ;) |
| 15:33.57 | jsmith | Woifi1988: An auto-fallthrough happens when you run out of priority numbers... are your priorities numbered correctly? |
| 15:33.58 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 15:34.07 | Daviey | Price Vs. Features = Linksys |
| 15:34.11 | Galeras | Daviey: a dream |
| 15:34.28 | jsmith | [TK]D-Fender: Except for reboot speed, ease of configuration, support department... |
| 15:34.38 | jsmith | I have yet to find the perfect phone. |
| 15:35.01 | Woifi1988 | jsmith : yes they are numbered correctly |
| 15:35.35 | jsmith | Woifi1988: Then it's up to you to figure out why Asterisk is running out of priorities for that extension |
| 15:35.59 | Woifi1988 | what does "run out of priorities" mean? |
| 15:36.44 | *** join/#asterisk gardo (n=gardo@121.97.213.202) |
| 15:36.58 | [TK]D-Fender | jsmith: Well to counter : Doesn't NEED to be constantly rebooted (mine stay up until I update firmwares every several months). Ease of configuration has a steeper initial learning curve. Afterwards it sets up real fast and easy. And Have never NEEDED any support. They WORK. I DID RMA one phone that fried though and my reseller did it fast and without a hitch |
| 15:37.27 | [TK]D-Fender | Woifi1988: Auto-fallthrough means its FINISH processing your exten. Go alook at what its DOING <- |
| 15:38.41 | Galeras | no votes for grandstream? |
| 15:39.13 | Woifi1988 | maybe this line is wrong? => exten => GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?022388761,1) |
| 15:40.26 | jsmith | [TK]D-Fender: Well, let me counter-counter with this... I deploy Polycom phones every month as part of my Asterisk Bootcamp classes. And since we've switched to Polycom phones, I've run into numerous issues. |
| 15:40.46 | [TK]D-Fender | Woifi1988: that has no exten or priority on it! |
| 15:40.51 | redbaron1973 | tzafrir: I have modified the patches for it to comple with CENTOS ppp-2.4.2-6.4.RHEL4.src.rpm |
| 15:41.04 | jsmith | [TK]D-Fender: Yes, about half of them are fixed by simply updating the firmware, but there are still a lot of quirky issues with the Polycom firmware |
| 15:41.08 | [TK]D-Fender | jsmith: Ok, what'd you run into? |
| 15:41.31 | Woifi1988 | it has 022388761 as extension and 1 as priority |
| 15:41.51 | [TK]D-Fender | jsmith: the only problem I've ever been able to replicate that I'd heard of is on 1 specifc version if you kept typing a ridiculous # of digits in an on-hook dial it'll lock. |
| 15:42.03 | [TK]D-Fender | Woifi1988: Pastebin your dialplan |
| 15:42.12 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 15:42.20 | [TK]D-Fender | Woifi1988: and as priority 1 WON'T WORK. |
| 15:42.34 | [TK]D-Fender | Woifi1988: You have to check DIALSTATUS ***AFTER** you try to dial. |
| 15:42.40 | jsmith | [TK]D-Fender: Phones that reboot every few minutes, phones that lock up on attended transfers, had a batch of phones that would mangle the SIP headers |
| 15:42.46 | darkfires | Asterisk is sending these packets at like 30 per second to an ATA.... IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 60) 10.1.0.1.19896 > 192.168.42.19.16414: [udp sum ok] UDP, length 32 |
| 15:42.46 | redbaron1973 | I am getting the error:Unable to put device '1' into HDLC mode |
| 15:43.04 | redbaron1973 | Do I need to also follow the docs instructions for enabling HDLC to get PPP to work? |
| 15:43.05 | [TK]D-Fender | jsmith: What firmwares & models? never heard of this before you now. |
| 15:43.05 | jsmith | [TK]D-Fender: Of course, the occasional phone that won't sync to an NTP server |
| 15:43.27 | Wonka | that sounds fun |
| 15:43.27 | [TK]D-Fender | jsmith: I usually reduce my NTP synch to 1 HR personally.... usually picks up. |
| 15:43.28 | jsmith | [TK]D-Fender: Most of the problems have been with IP320 and IP330 models (Yeah, I know.... don't bother) |
| 15:43.44 | Wonka | another reason to want linux on IP phones, IMO |
| 15:43.45 | [TK]D-Fender | jsmith: I haven't physcally worked with them, but have remote deployed. |
| 15:43.55 | Woifi1988 | ~pb |
| 15:43.56 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 15:43.57 | jsmith | [TK]D-Fender: I'm not a phone newbie... and I've been configuring Polycom phones for 5 years now... |
| 15:44.12 | jsmith | [TK]D-Fender: But Polycom phones still bug me |
| 15:44.18 | [TK]D-Fender | jsmith: I'm a little over 3 here myself.... |
| 15:44.27 | tzanger | jsmith: that's why you need to encrypt your voice |
| 15:44.51 | redbaron1973 | kernel: Zaptel: Zaptel PPP support not compiled in |
| 15:44.59 | Qwell[] | tzanger: the bug would be in the mic - before the encryption :p |
| 15:45.00 | jsmith | tzanger: I'm from Wyoming... even if I speak plain english, nobody can understand me! |
| 15:45.02 | Woifi1988 | [TK]D-Fender: pastebin.ca/674839 |
| 15:45.04 | redbaron1973 | I have modified ppp, but do I need to rebuild the kernel now? |
| 15:45.09 | tzanger | Qwell[]: I said voice, not SIP traffic :-) |
| 15:45.09 | [TK]D-Fender | jsmith: You must be like my other "virgin sacrifices" I keep around. People like you feel the karmic volcano so the rest of us live trouble-free :) |
| 15:45.16 | tzanger | jsmith: bwahahahahahaha |
| 15:45.27 | Qwell[] | jsmith: I don't think I've ever had trouble understanding you... |
| 15:45.32 | jsmith | [TK]D-Fender: I pride myself on being able to find bugs nobody else can |
| 15:45.36 | tzanger | [TK]D-Fender: I'll have to remember that line, that's awesome |
| 15:46.12 | [TK]D-Fender | tzanger: wunderkin is my other Polycom "sacrificial virgin" ;) |
| 15:46.23 | *** join/#asterisk lbow (n=lbow@dsl-241-31-00.telkomadsl.co.za) |
| 15:47.46 | wunderkin | we haven't had any problems on 2.1.2, at least nothing anyone has said to me.. i'm checking 2.2.0 release notes .. seem to remember something about attended transfers but not affecting us... |
| 15:47.58 | Woifi1988 | [TK]D-Fender: http://pastebin.ca/674839 |
| 15:49.05 | [TK]D-Fender | Woifi1988: ok, now pastebin a call at verbose 10 |
| 15:49.17 | AsteriskProblems | D-Fender: just to let you know - ive got it working now, i factory reset the phone (again!) and it seems to have sorted itself out this time |
| 15:49.18 | [TK]D-Fender | wunderkin: There you are :p |
| 15:49.23 | AsteriskProblems | (probably me being stupid) |
| 15:49.36 | [TK]D-Fender | AsteriskProblems: Inconceivable! |
| 15:49.53 | AsteriskProblems | well one thing - ive sure learnt a lot about asterisk :) thanks for everyones help |
| 15:50.04 | Woifi1988 | [TK]D-Fender: How can i configure "verbose 10"? |
| 15:50.17 | jsmith | Woifi1988: "core set verbose 10" at the Asterisk CLI |
| 15:51.43 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:52.07 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:52.37 | wunderkin | hmm dunno but there are some weird situation reboot fixes in 2.2.0.. a lot of 320/330 fixes in 2.1.2.. |
| 15:53.43 | Woifi1988 | [TK]D-Fender: The only Warning is pbx.c:1797 pbx_extension_helper: No application '123' for extension (test-telefone, 123, 4) |
| 15:54.10 | jsmith | Woifi1988: Another syntax error in your dialplan |
| 15:54.26 | jsmith | Woifi1988: pastebin the output of "dialplan show 123@test-telefone" |
| 15:55.02 | [TK]D-Fender | Woifi1988: clearly you made a typo and did not reload the dialplan you showed me. |
| 15:55.15 | [TK]D-Fender | jsmith: Error is apparent, no need to see yet |
| 15:55.47 | pepo-- | [angel] |
| 15:55.47 | pepo-- | exten => _XX,1,Dial(SIP/${EXTEN}|10) [TIME] exten => *60,2,playback(at-tone-time-exactly) |
| 15:55.47 | pepo-- | exten => *60,3,sayunixtime(,,IMp) |
| 15:55.58 | [TK]D-Fender | wunderkin: I'm running 2.2.0 on my IP 501 at home now, seems fine, doing my home 301 next, and then preppeing my office IP600 mass reconfig |
| 15:56.05 | jsmith | pepo--: Why are you starting your *60 extension with priority 3? |
| 15:56.16 | jsmith | 2, that is |
| 15:56.28 | Woifi1988 | ohh sorry i havent reloaded it... just for future needs: how can i save an output from the asterisk cli in a file? |
| 15:56.29 | [TK]D-Fender | pepo--: Asterisk will NOT jump from processing _XX to *60 <------ |
| 15:57.03 | [TK]D-Fender | pepo--: It doesn't care that you put it below. if you start with _XX it will ONLY continue on _XX extens. |
| 15:57.07 | pepo-- | jsmith, but if i write [TIME] i can listen SAYUNIXTIME |
| 15:57.19 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:57.19 | [TK]D-Fender | pepo--: And *60 doesn't even MATCH _XX |
| 15:57.27 | pepo-- | hmm |
| 15:57.51 | [TK]D-Fender | pepo--: "Put Down The Crack Pipe" (c) JerJer |
| 15:57.52 | tzanger | [TK]D-Fender: is there not an extensions.conf config option that has it fall thorugh to the next listed extension if you "fall off" the end of the current one? |
| 15:58.09 | Woifi1988 | ohh sorry i havent reloaded it... just for future needs: how can i save an output from the asterisk cli in a file? |
| 15:58.13 | [TK]D-Fender | tzanger: Nope. when an Exten is doe, its DONE |
| 15:58.26 | [TK]D-Fender | Woifi1988: copy & paste. |
| 15:58.42 | tzanger | I have never seen it, but I thought there was an option (which was ridiculous in my opinion) |
| 15:58.46 | pepo-- | jsmith, http://pastebin.com/m6736b3d9 |
| 15:59.04 | pepo-- | if i add [time] dont work |
| 15:59.05 | Woifi1988 | [TK]D-Fender: I use a non gui distribution |
| 15:59.51 | jfitzgibbon | tzanger: you're maybe thinking of autofallthrough=no, which kind of makes every extension have an implicit WaitExten() as it's last priority |
| 16:00.21 | tzanger | jfitzgibbon: aha! that was it |
| 16:00.22 | [TK]D-Fender | Woifi1988: Who said anything about a GUI? I sure didn't. |
| 16:00.42 | Woifi1988 | how can i do a copy an paste without a terminal? |
| 16:00.44 | [TK]D-Fender | Woifi1988: SSH to your sever. Conecct to your running * process. go through your scroll-back. Copy & paste |
| 16:01.15 | [TK]D-Fender | Woifi1988: Who said without a terminal? You are thinking to damn much. As I've said to countless others, this is a task best left to trained professionals! |
| 16:01.25 | jfitzgibbon | wolfi: you can also add the 'verbose' category to a destination in logger.conf and do a 'logger reload' |
| 16:01.52 | jfitzgibbon | wolfi: then make sure that you've done 'core set verbose 3' (or higher) |
| 16:02.11 | [TK]D-Fender | pepo--: Do it how you THINK it should be done, then we'll confirm where you went wrong |
| 16:02.26 | pepo-- | d |
| 16:02.28 | pepo-- | xD |
| 16:03.27 | Woifi1988 | jfitzgibbon: thats a good idea! |
| 16:03.47 | Woifi1988 | [TK]D-Fender: how can i connect to a running process? |
| 16:04.05 | Woifi1988 | i can't do a altF9 in ssh |
| 16:04.47 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 16:05.05 | Galeras | Can this work?: Put an * box in the middle of an E1/PRI and a panasonic PABX to use it as an "analog extensions bank" |
| 16:05.36 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 16:06.50 | jsmith | Galeras: Yes |
| 16:07.46 | Galeras | jsmith: thanks |
| 16:08.10 | pepo-- | [TK]D-Fender, if i delete * then i call t 60 unable to create channel of type SIP |
| 16:08.28 | darkfires | does anyone here work on the asterisk source? |
| 16:08.35 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:08.42 | Qwell[] | darkfires: I'm sure there is a developer or two in here |
| 16:09.12 | darkfires | http://pastebin.com/d6bf31550 |
| 16:09.28 | darkfires | im having that issue....seems like asterisk is dumping its stack to a ata over sip ? |
| 16:12.21 | Woifi1988 | [TK]D-Fender: i've got it: asterisk -rvvvv |
| 16:12.24 | Woifi1988 | thx |
| 16:12.41 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 16:13.17 | russellb | darkfires: wtf? |
| 16:13.20 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 16:13.21 | russellb | darkfires: that is quite bizarre |
| 16:13.39 | darkfires | you're telling me |
| 16:13.58 | russellb | does it happen reliably? |
| 16:14.02 | russellb | or random? |
| 16:14.13 | darkfires | i just had a problem with asterisk flooding an ATA so i updated the SVN ...and now i get that |
| 16:14.36 | russellb | latest 1.4? |
| 16:14.49 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 16:14.49 | *** join/#asterisk ptiggerdine (n=ptiggerd@123-243-144-208.tpgi.com.au) |
| 16:14.49 | *** mode/#asterisk [+o codefreeze] by irc.freenode.net |
| 16:15.15 | syzygyBSD | darkfires: actually, it looks like the ata is sending a stack to asterisk... |
| 16:15.18 | syzygyBSD | right? |
| 16:15.23 | darkfires | no |
| 16:15.30 | darkfires | pap2 doesn't have i686 tls libraries |
| 16:15.34 | *** join/#asterisk gmfm (n=hithere@216.161.142.20) |
| 16:16.05 | syzygyBSD | well, which direction do the messages start from? |
| 16:16.19 | darkfires | asterisk |
| 16:16.22 | russellb | it says ... 192.168.42.19.sip > pbx.jayrobinson.ca.sip: SIP, |
| 16:16.41 | darkfires | its replying |
| 16:16.43 | darkfires | one sec |
| 16:16.52 | darkfires | look at the 2nd packet |
| 16:16.58 | darkfires | i copied it wrong |
| 16:17.04 | darkfires | 2nd packet goes out, then the first one comes back... |
| 16:17.24 | syzygyBSD | um.. not as seen by the times... |
| 16:17.30 | russellb | that is asterisk just sending it back |
| 16:17.32 | syzygyBSD | but whatever |
| 16:17.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-79-180-14-100.red.bezeqint.net) |
| 16:18.11 | russellb | the ata is replying to a 489 Bad Event with a NOTIFY? |
| 16:18.21 | russellb | i don't think so ... |
| 16:18.28 | russellb | the timestamps say the first packet there came first |
| 16:18.30 | syzygyBSD | no, pap is sending notify, asterisk is sending back bad event.. |
| 16:18.32 | syzygyBSD | ya... |
| 16:18.34 | russellb | the ATA sent a bogus pcket to asterisk |
| 16:18.37 | darkfires | how would the ATA get ahold of asterisk stack data |
| 16:18.42 | russellb | asterisk is responding saying Bad Event |
| 16:18.47 | russellb | it's not asterisk stack data |
| 16:19.00 | russellb | it's the ATA sending screwed up stuff |
| 16:19.05 | darkfires | <PROTECTED> |
| 16:19.06 | russellb | and Asterisk saying "wtf?" back to it |
| 16:19.07 | syzygyBSD | darkfires: why does it have to be the asterisk stack? |
| 16:19.25 | darkfires | because why would the linksys pap2 have i686 TLS libraries |
| 16:19.37 | russellb | well in the trace you have, the ATA is sending that crap first |
| 16:19.38 | darkfires | those libraries are on the machine that asterisk is running on |
| 16:19.42 | syzygyBSD | I don't know, but it doesn't mean that it doesn't |
| 16:19.48 | russellb | so if asterisk sent it first, there was a packet before these |
| 16:20.13 | darkfires | it has only happened since i started running asterisk 1.4 svn |
| 16:20.36 | darkfires | patrick puttman @ digium said for me to run svn because of kernel panics with hpec |
| 16:20.43 | darkfires | which svn solved |
| 16:20.51 | darkfires | but now this is happening |
| 16:20.56 | syzygyBSD | darkfires: can you pastebin an entire sip debug? |
| 16:21.02 | darkfires | ya give me a minute |
| 16:21.03 | Nugget | life on the bleeding edge. :) |
| 16:21.04 | *** join/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 16:21.19 | darkfires | I don't want to be on bleeding edge, i just want it to work without kernel panicking ;) |
| 16:21.26 | syzygyBSD | starting right after boot of the pap2 |
| 16:21.42 | Aeudian | I just finished setting up my ITSP(SIP) for inbound callnig which works, but all phone calls inbound, shows the CID number but the name shows as "New User" why is this? |
| 16:22.42 | jsmith | Aeudian: Either something on your system is setting that, or your ITSP isn't passing along the name |
| 16:23.13 | Aeudian | jsmith: what passes the CID name? the dundi settings? |
| 16:23.17 | [TK]D-Fender | pepo--: You still haven't pastebinned how you THINK it should be done |
| 16:23.40 | jsmith | Aeudian: callerid= setting in sip.conf |
| 16:23.50 | jsmith | Aeudian: CallerID has *nothing* to do with DUNDi |
| 16:25.37 | syzygyBSD | unless you make it... |
| 16:26.56 | jfitzgibbon | got a wierd PRI problem if anyone can sanity check this debug output: http://pastebin.com/m43fec484 |
| 16:27.15 | jfitzgibbon | basically I get cause code 44 (channel or resource unavailable) whenever I attempt to call out on 4 of the 23 B channels of a PRI |
| 16:27.22 | jfitzgibbon | but the other 19 channels are fine |
| 16:27.27 | darkfires | http://pastebin.com/d2f8d22af sip debug |
| 16:27.39 | Aeudian | jsmith: Okay, i see that i can make a callerid="Name" under sip but, how to do I pass company callerid show that it shows on the phones? would callerid= with nothing after be like a wild card to accept from carrier? |
| 16:28.07 | jfitzgibbon | and when this happens, * sends the network a RESTART request for the B channel, but I never get a restart ack back, so the channel sits in "restarting" state until I restart * or chan_zap |
| 16:28.10 | darkfires | doesn't show anything useful though |
| 16:28.40 | Aeudian | jsmith: I know for a fact that my carrier supports CID naming, and our number shows up on inbound phone calls, but our name and every phone i use says "New User" |
| 16:28.43 | darkfires | oh i can do sip debug by ip only heh that makes it easier |
| 16:30.22 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 16:30.52 | Aeudian | jsmith: if i am reading it correclt the callerid in sip tab controls outbound callerid naming? my problem is inbound naming not being passed, |
| 16:31.23 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 16:31.32 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:31.36 | generalhan | hey all |
| 16:31.44 | *** join/#asterisk redbaron1973 (n=redbaron@host55-226.rancor.birch.net) |
| 16:32.07 | generalhan | anyone know what the latest firmware version is for the Cisco 7690? |
| 16:32.42 | redbaron1973 | tzafrir: I recompiled the kernel with ppp compiled in, not module, and it now appears to work. |
| 16:32.44 | redbaron1973 | thx! |
| 16:32.57 | uwe | hello, i would like to thank everyone who helps helpless people like me , and especially those who suggested a while ago to change my NIC card and not use the builtin Gigabit interface, an old 3com did fix all my problems so far, which were really ugly ones ... for three days now im facing no trouble at all!! i hope it will go on this way, thank you again :) |
| 16:33.53 | uwe | and of course [TK]D-Fender , who is always helping :D |
| 16:35.22 | Nugget | latest 7960 firmware is 8.3.2 released 10-Aug-2007. |
| 16:36.25 | generalhan | hmm ... im using 08-6-00, and it has been working perfectly for some time now. but all the sudden one user's phone just keeps rebooting itself at random times :( |
| 16:36.26 | darkfires | <PROTECTED> |
| 16:36.26 | darkfires | ber123:36:07.739396 IP (tos 0x0, ttl 249, id 4016, offset 0, flags [none], proto: ICMP (1), length: 56) 192.168.42.19 > 10.1.0.1: ICMP 192.168.42.19 udp port 16406 unreachable, length 36 |
| 16:37.08 | generalhan | so i changed the power brick thinking that was it (3 times) and it still does it. so i replaced the phone itself now, 3 times. and it STILL does it. its sooo strange |
| 16:37.33 | generalhan | and all the other 14 cisco phones have the same firmware and are just fine |
| 16:38.13 | generalhan | even the ones that got the replaced phone in return for their good phone ... its like this issue only follows this one extension ! |
| 16:38.24 | dlynes_home | generalhan: maybe there's a lose connection on the dataport for that phone and it keeps rebooting, because the dhcp address keeps changing |
| 16:38.36 | generalhan | hmmm |
| 16:38.42 | generalhan | never thought of that |
| 16:38.59 | generalhan | maybe i should setup a IP reservation for that MAC and see if that solves anything |
| 16:40.19 | generalhan | but i really should get another support contract so that i can get the newest firmware too. my contract ran out some time ago, but everything was working so great that i didnt think i would need it again ! |
| 16:41.43 | syzygyBSD | darkfires: were there any of the stack dump in the time the sip debug was taken? |
| 16:46.19 | darkfires | syzygyBSD yes but the sip debug isn't showing it... i guess i will have to do an strace on asterisk to see ? |
| 16:53.05 | krdian_ | hmmm, is there any way to cancell restart ? |
| 16:53.26 | krdian_ | restart when convenient |
| 16:53.29 | krdian_ | ? |
| 16:57.24 | syzygyBSD | krdian_: why do you wnat to cancel it? |
| 16:58.53 | krdian_ | syzygyBSD: i have to go out and i don't know what happen after restart coz i made some changes :) |
| 16:59.01 | tzafrir_laptop | kaldemar, "no" or "cancel" |
| 16:59.08 | tzafrir_laptop | or something similar |
| 16:59.48 | krdian_ | tzafrir_laptop: nope |
| 17:00.35 | krdian_ | tzafrir_laptop: my * doesn't have sth like that |
| 17:01.44 | krdian_ | ok, never mind, its restarted already |
| 17:02.29 | krdian_ | but ... maybe its not so bad idea to have possibility to cance restart ? |
| 17:04.09 | krdian_ | particularly for restart gracefuly |
| 17:04.18 | dlynes_home | generalhan: that's just a gotcha built in to cisco stuff, as an incentive to keep renewing your support contract |
| 17:05.38 | *** join/#asterisk jsmith (n=jsmith@000-143-916.area3.spcsdns.net) |
| 17:05.38 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 17:06.33 | *** join/#asterisk RU (n=ru@85.15.191.66) |
| 17:07.18 | RU | Hi, there! Please, tell me where can I get RPM packages for RedHat EL5? |
| 17:07.23 | coppice | A 7960, sir, and would you like to take advantage of our extended warranty package? :-) |
| 17:07.28 | RU | Asterisk RPM I mean |
| 17:07.56 | generalhan | bah i dont do warranties ! lol. but the support contract i needed for the firmware downloads. |
| 17:08.59 | generalhan | but i just went through CDW and got like a $5 one or something like that ... ill have to call them up again and see if i can still do that same thing. |
| 17:09.00 | generalhan | all i need are the DLs its not like im calling them up every day for issues |
| 17:09.00 | dlynes_home | generalhan: support contract, extended warranty...are they not one and the same...just horses of a different color? |
| 17:09.11 | errr | I have someone who is calling us saying that they hear static when they call us. Is there a way to find out if its our system thats causing the static? Would a recording of the call help with this? |
| 17:09.40 | generalhan | dlynes_home: nah, warranty would be for the hardware ... to replace it if it breaks. support contract would be for me to get support on the phone, but i would still need to buy a new phone if it broke ! |
| 17:10.05 | krdian_ | <PROTECTED> |
| 17:10.14 | generalhan | its the difference of paying $5/yr and $200(dont really know the cost)/yr |
| 17:10.15 | tzafrir_laptop | RU, check #centos. but don't tell them it's called "redhat", as they won't believe you |
| 17:10.42 | tzafrir_laptop | ah, asterisk packages? check atrpms.net |
| 17:10.42 | dlynes_home | generalhan: yeah...i was just getting at, a support contract is a different way of saying 'limited extended warranty, with no phone support' |
| 17:10.54 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 17:10.56 | generalhan | dlynes_home: lol, ok ! |
| 17:11.07 | generalhan | ill accept that definition ! |
| 17:11.15 | *** join/#asterisk awellssjtg (n=awellssj@adsl-070-155-079-003.sip.asm.bellsouth.net) |
| 17:12.16 | RU | thank a lot |
| 17:12.21 | RU | I'll try |
| 17:12.38 | generalhan | either way i need to get the newest firmware, and my link isnt working (i dont think). ususally when i go there to DL the firmware it has pages and pages of downloads. i go there now and there is only 2. Release nots for, and the actual download for, SIP version 8-2-00 |
| 17:12.48 | generalhan | which is a version ive had for a VERY long time |
| 17:21.32 | *** join/#asterisk Op3r (n=Op3r@121.97.247.27) |
| 17:22.33 | RU | Thanks people! I found Asterisk RPMs for RedHat-EL-4/5 and CentOS 4/5 at http://www.laimbock.com/asterisk/ |
| 17:23.09 | errr | RU: IMO you shouldnt use an rpm for it, You should really build it from source |
| 17:23.25 | RU | errr> why? |
| 17:23.36 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 17:23.42 | RU | errr> It's much easier. |
| 17:24.04 | errr | RU: if you say so |
| 17:24.13 | *** join/#asterisk eonz (n=Icarus@irc.americatelnet.com.pe) |
| 17:26.24 | tzafrir_laptop | because installing from source is cool |
| 17:32.09 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:44.32 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 17:46.33 | nDuff | Are there any existing 802.11-based phones with encryption support (be it OpenVPN, IPsec or TLS/SRTP)? |
| 17:52.54 | [TK]D-Fender | nDuff: Snom 360 does, |
| 17:53.08 | nDuff | [TK]D-Fender: that's where the "802.11" bit comes in. |
| 17:53.29 | [TK]D-Fender | nDuff: Ok, so its a rounding error! :p |
| 17:55.26 | [TK]D-Fender | nDuff: problem is the CPU power needed to encode.... Dunno if there's one out that'll handle it... |
| 17:56.44 | [TK]D-Fender | nDuff: None of the usual retailers has anything if one even exists. |
| 17:56.53 | [TK]D-Fender | nDuff: Wait for MokoIAX ;) |
| 18:02.11 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 18:02.29 | nDuff | [TK]D-Fender: that *does* look sweet. if it's relying on IAX's encryption, though -- I thought there was an analysis indicating that there's a possible keying vulnerability which hasn't been answered yet. |
| 18:03.30 | *** join/#asterisk malph_work (n=chatzill@66-231-0-194.hosts.sdnet.net) |
| 18:04.03 | [TK]D-Fender | nDuff: Clearly no phone out there will make you happy... |
| 18:04.12 | [TK]D-Fender | nDuff: Your "Cone Of Silence" is in the mail :p |
| 18:04.36 | nDuff | heh. if it's fast enough, though, OpenVPN might be usable on the OpenMoko |
| 18:04.58 | elixer | so i want to test some usage scenarios on a new PBX i am building and i want to have a few dozen SIP phones registered for the test. i want to be able to "ring" individual extens, put a couple in a queue, and just have other sitting there idle. is 'sipp' what i am looking for? or are there other tools? |
| 18:05.00 | [TK]D-Fender | nDuff: Clearly THIS is the more elegant solution! http://en.wikipedia.org/wiki/Cone_of_Silence |
| 18:05.13 | nDuff | [TK]D-Fender: I'm not *that* young. |
| 18:05.36 | *** join/#asterisk drwelby (n=mpfister@68.186.35.242) |
| 18:06.15 | [TK]D-Fender | nDuff: Would you believe.... |
| 18:06.20 | [TK]D-Fender | nyuk! |
| 18:06.34 | datachomper | Can anybody reccomend a good book for learning about the telecom networks? How ss7 works, how to hook into carriers, etc ... |
| 18:07.27 | generalhan | [TK]D-Fender: hahaha "Would you believe ... 2" hahaha i havnet though about that in sooo long ! |
| 18:08.06 | [TK]D-Fender | ~telephony101 |
| 18:08.14 | [TK]D-Fender | ~telecom101 |
| 18:08.24 | generalhan | man i miss that show ... i wonder if they have, or plann to release the DVD seasons of that ! i would soo buy them ! |
| 18:08.26 | [TK]D-Fender | Strom_M: Whats that link of yours again? |
| 18:08.54 | [TK]D-Fender | datachomper: http://www.stromcarlson.com/docs/ |
| 18:09.25 | [TK]D-Fender | generalhan: Already released most likely, and they're making a NEW movie :) |
| 18:09.44 | [TK]D-Fender | generalhan: http://www.imdb.com/title/tt0425061/ |
| 18:09.58 | [TK]D-Fender | generalhan: Steve Carell .... perfect choice. |
| 18:10.07 | Strom_M | [TK]D-Fender: |
| 18:10.11 | Strom_M | ~101 |
| 18:10.12 | jbot | from memory, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 18:10.21 | generalhan | YES !!! this is gonna be GREAT ! |
| 18:10.27 | [TK]D-Fender | Strom_M: thanks... Googled you up FAST though :) |
| 18:11.18 | *** join/#asterisk cirgal (i=robert@216.193.203.2) |
| 18:11.35 | Strom_M | :0 |
| 18:11.37 | Strom_M | er :) |
| 18:12.15 | generalhan | http://www.wouldyoubelieve.com/dvd.html <-- there they are ! |
| 18:13.03 | [TK]D-Fender | ~[TK]D-Fender |
| 18:13.03 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 18:13.09 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 18:13.31 | elixer | ... and team killer |
| 18:13.50 | elixer | whenever i see "TK" i think team killer. old habits. |
| 18:14.22 | [TK]D-Fender | elixer: My old clan :) |
| 18:14.29 | cirgal | silly question - have mercy: my assumption is that dialplan applications like, er, Dial fall through to the next part of the dialplan immediately when they run? |
| 18:14.38 | elixer | [TK]D-Fender: ahhh :-) CS? |
| 18:14.39 | Strom_M | cirgal: no |
| 18:14.48 | [TK]D-Fender | elixer: Action:Half-Life |
| 18:14.52 | elixer | nice |
| 18:15.00 | cirgal | Strom_M: can i give you a specific example? |
| 18:15.14 | Strom_M | Dial() for example will execute until the call is torn down |
| 18:15.27 | [TK]D-Fender | cirgal: Dialplan apps do what the do and things continue (if even applicable) AFTER they are finished. |
| 18:15.48 | [TK]D-Fender | cirgal: there is no "background" processing |
| 18:16.45 | cirgal | so if I have exten => ...,Dial(SIP/etc) on one line, and another dialplan app on the next, the next won't pick up with the channel created w/Dial. |
| 18:17.01 | cirgal | basically it's sequential and synchronous. |
| 18:17.19 | [TK]D-Fender | cirgal: Correct. |
| 18:17.30 | cirgal | Thanks folks. |
| 18:17.36 | [TK]D-Fender | cirgal: Things continue when you call is completed or fails |
| 18:17.44 | [TK]D-Fender | cirgal: "normally" |
| 18:17.47 | cirgal | perhaps it wasn't such a silly question ;) |
| 18:17.52 | [TK]D-Fender | cirgal: What would you LIKE to have happen? |
| 18:18.01 | cirgal | Let me explain what I'm trying for. |
| 18:18.07 | [TK]D-Fender | cirgal: This is an amount of cheating that can be done |
| 18:18.11 | [TK]D-Fender | there* |
| 18:18.43 | cirgal | I would like to dial out and when the other endpoint picks up, do some processing on the channel with my own custom dialplan application. |
| 18:19.06 | cirgal | Said processing is listening for some specific tones and replying with some other specific tones, somewhat like DTMF |
| 18:19.15 | [TK]D-Fender | cirgal: Sounds like you want ASTERISk to make the call out, and not be like using 1 phone to dial another. |
| 18:19.22 | cirgal | yes. |
| 18:19.31 | cirgal | Exactly correct. |
| 18:19.32 | [TK]D-Fender | cirgal: Ah, that IS different |
| 18:19.46 | [TK]D-Fender | cirgal: Lookup "call files", "AMI originate" on the WIKI |
| 18:19.53 | [TK]D-Fender | cirgal: This is entirely doable. |
| 18:19.59 | cirgal | <PROTECTED> |
| 18:20.09 | *** join/#asterisk haystack (n=bve@ip5457284a.direct-adsl.nl) |
| 18:20.15 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 18:20.16 | [TK]D-Fender | cirgal: these are 2 ways to have * dial out and dump the caller into the dialplan upon answering |
| 18:20.39 | [TK]D-Fender | cirgal: From there its as though they called in instead. |
| 18:20.41 | malph_work | I was trying to find a way to execute an agi script when a call ended. what should I be searching for? |
| 18:21.04 | jsmith | malph_work: You want to look for DeagAGI and the magic 'h' extension |
| 18:21.05 | [TK]D-Fender | malph_work: the "g" option for Dial, and/or the "h" standard extension. |
| 18:21.07 | many | 00,h, |
| 18:21.07 | cirgal | <PROTECTED> |
| 18:25.58 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 18:28.58 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 18:30.32 | *** join/#asterisk pgarcia (n=pedgarci@c934ac83.virtua.com.br) |
| 18:30.58 | cirgal | [TK]D-Fender, I think I said yes when I meant no. :) |
| 18:32.21 | [TK]D-Fender | cirgal: Whatever! We're clear now and it didn't take to 50 million question so twits around here require :p |
| 18:32.23 | cirgal | This is what I want to happen: 1) a call comes into *. 2) * answers and does brief handshaking, then puts it on hold. 3) * dials out to another endpoint (originating a call), and does brief handshaking. 4) Upon completion of handshaking, 8 _bridges_ the two calls. |
| 18:32.50 | [TK]D-Fender | cirgal: No. What I described is * INITIATING the call. |
| 18:32.59 | cirgal | Got it, sorry for the misunderstanding. |
| 18:33.18 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 18:33.25 | cirgal | In this case someone else initiates the call, * initiates a 2nd call, does some handshaking, then bridges the two. |
| 18:33.34 | cirgal | Perhaps queues or conferencing is the way to go. |
| 18:33.36 | [TK]D-Fender | cirgal: So your server (due to a trigger of your determining) is made to call out and the caller is dumped into the dialplan where you can do whatever you would like. |
| 18:33.57 | cirgal | [TK]D-Fender: Yes. |
| 18:34.10 | cirgal | [TK]D-Fender: It looks like call files can do this nicely. |
| 18:34.11 | [TK]D-Fender | cirgal: So you want someone to make the call out and THEN pass it off for processing? |
| 18:34.23 | [TK]D-Fender | cirgal: you are starting to head in circles... |
| 18:34.29 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 18:34.43 | cirgal | [TK]D-Fender: Let me try to clarify, but my last 1) 2) ... is the sequence I want. |
| 18:35.05 | [TK]D-Fender | cirgal: clarity = good |
| 18:35.54 | cirgal | [TK]D-Fender: A call comes in to *. * puts it on hold. * _dials out_ on a _new outbound call_. On this new call, * needs to do some processing on the line, suffice it to say it listens for DTMFs and sends some DTMFs. |
| 18:36.30 | cirgal | [TK]D-Fender: Then, * bridges the original call that came in to the _new outbound call_. |
| 18:36.52 | cirgal | [TK]D-Fender: Then, * is done. :) I hope that's clear. |
| 18:37.37 | cirgal | (btw, the expertise is much appreciated :) ) |
| 18:38.08 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 18:39.21 | [TK]D-Fender | cirgal: Ok, THIS is hard... |
| 18:40.23 | *** part/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 18:40.42 | syzygyBSD | cirgal: I think AGI would be the easiest way to accomplish that |
| 18:41.18 | [TK]D-Fender | cirgal: I can picture a way.... you would use a specail Parking app called "ValetParking" to "park" your first caller into a dedicated slot. It would them pass that info on to a newly originated call-file generated call that would do all of this work. Then upon reaching that special point in your dialplan, it would bridge to that parked call. |
| 18:41.31 | *** join/#asterisk nirz (n=nirz@bzq-79-179-88-51.red.bezeqint.net) |
| 18:41.44 | [TK]D-Fender | syzygyBSD: Not AGI..... actually deserving of a full "C" app.... |
| 18:41.57 | [TK]D-Fender | syzygyBSD: But potentially doable with some hacks. |
| 18:42.01 | syzygyBSD | really? I have done things like that with agi |
| 18:42.21 | syzygyBSD | sure it might deserve a c app, but I still did it in agi |
| 18:44.21 | cirgal | [TK]D-Fender: It's the "all of this work" part that seems to be the difficult part. If I can't pass the channel to a function or dialplan app to do that processing until Dial() comes back ... |
| 18:44.50 | rudholm | anyone having problems getting calls out through teliax at the moment? |
| 18:45.50 | [TK]D-Fender | cirgal: What you are missing is that these are 2 DIFFERENT calls. they ge rebridged by ValetParking AFTER that "new leg" is satisfied |
| 18:46.14 | [TK]D-Fender | cirgal: it is ugly and a little painful, but possible |
| 18:48.12 | cirgal | [TK]D-Fender: Ok, I know pain :) |
| 18:48.45 | cirgal | [TK]D-Fender: If the bridging part is doable, then it's really the "dial out, and then do processing on the line before bridging" part that's troubling me. |
| 18:48.52 | nDuff | Anyone know about what I should expect to pay for a voice-only PRI? I'm looking at getting a very general price range to run 1-4 PRIs to our colo facility, but our sales rep with Time Warner Telecom isn't answering their phone right now; our current service is a funky voice/data 3-T1 package, so I don't have any clue as to voice-only pricing. |
| 18:48.57 | [TK]D-Fender | cirgal: This will be like an Oprah Exclusive then :p |
| 18:49.14 | [TK]D-Fender | cirgal: actually... thats not THAT hard..... |
| 18:49.33 | [TK]D-Fender | nDuff: Extremely dependent on location and company |
| 18:49.36 | cirgal | [TK]D-Fender, syzygyBSD: (first, hahahah Oprag). Then, I considered EAGI at first but I'd like this to be 'fast', so I thought a dialplan app would be better. |
| 18:50.22 | [TK]D-Fender | syzygyBSD: what do you think about the Macro function in Dial for this? like privacy effectively... |
| 18:50.56 | syzygyBSD | cirgal: how fast is fast? |
| 18:51.21 | syzygyBSD | ie, it will be dialing another system.. |
| 18:51.36 | cirgal | syzygyBSD: truthfully it's a vague notion at this point. |
| 18:52.13 | [TK]D-Fender | cirgal: No.. it was a vague notion THEN too :p |
| 18:52.26 | cirgal | [TK]D-Fender: indeed :) |
| 18:52.38 | [TK]D-Fender | We don't neeeed no steeeeeeenking deeettttaaails |
| 18:53.23 | cirgal | heh |
| 18:57.17 | *** join/#asterisk mtaht4 (n=m@42-109-62-200.enitel.net.ni) |
| 18:57.38 | cirgal | [TK]D-Fender: I think I'll experiment with a Parking app to park the first call, and a .call file to do the outbound call processing and also to dump the outbound call into a dialplan that will then bridge the two calls. |
| 18:57.48 | cirgal | [TK]D-Fender: does that sound sane? |
| 18:58.11 | [TK]D-Fender | cirgal: nO, BUT IT MIGHT JUST WORK! |
| 18:58.20 | *** join/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 18:58.40 | cirgal | [TK]D-Fender: outstanding. i don't need sane. i need to pay the billz. |
| 19:01.09 | [TK]D-Fender | cirgal: You learn quickly young Jedi..... |
| 19:01.39 | cirgal | [TK]D-Fender: I don't know why everyone says you guys are mean. This is nothing compared to my regular haunt. |
| 19:02.53 | cirgal | I'd be in tears by now. |
| 19:04.06 | [TK]D-Fender | cirgal: We don't need to give you pain.... you're evidently more than capable of finding it yourself. |
| 19:04.32 | cirgal | Pity me. |
| 19:08.21 | *** part/#asterisk mtaht4 (n=m@42-109-62-200.enitel.net.ni) |
| 19:09.26 | CoolGuy21 | is there anything that stops 1 device from registering 2 extensions? i have a Cisco 7940 but i cannot register 2 extensions, the second one fails. |
| 19:10.39 | [TK]D-Fender | cirgal: "If it is weak kill it, or ignore it. Anything else honours it" |
| 19:10.57 | [TK]D-Fender | CoolGuy21: only a bad config |
| 19:14.58 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 19:16.03 | datachomper | Dumb question, but as I understand a T1 can provide 24 channels of voice, is that duplex or one way? |
| 19:16.03 | *** part/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 19:16.46 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:17.23 | *** join/#asterisk kkn088 (n=kikoun@88-136-56-85.adslgp.cegetel.net) |
| 19:21.19 | [TK]D-Fender | datachomper: 24 bidirection channels. But then again we highly recommend PRI signalling which only leaves you 23 |
| 19:25.41 | datachomper | So just doing the math ... 1 voice channel is sampled at 8,000bits/s * 8 bit resolution = 64Kb per channel. Then 64Kb x 24 = 1.536Mb/s which is the line speed of T1, right? |
| 19:25.53 | holiday_42 | heh, yep |
| 19:26.47 | holiday_42 | north american t1 anyway |
| 19:26.54 | datachomper | But I can only see that being one way? Unless a T1 setup consists of a pair of wires. |
| 19:27.21 | holiday_42 | two pairs |
| 19:27.37 | datachomper | So, 4 wires? |
| 19:27.42 | [TK]D-Fender | datachomper: mATH HAS not FAILED YOU. |
| 19:28.00 | [TK]D-Fender | datachomper: Yes, 4 wires |
| 19:28.08 | [TK]D-Fender | datachomper: 2 pairs on an RJ-48 |
| 19:28.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:28.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:28.50 | datachomper | So each pair carries half the traffic of a T1, just to be clear? |
| 19:29.00 | Qwell[] | there are rx and tx pairs |
| 19:29.06 | [TK]D-Fender | datachomper: 1 pair = TX, the other RX |
| 19:29.06 | holiday_42 | eh, no one pair receive, one send |
| 19:30.24 | datachomper | And what type of wire is it transmitted over, generally? |
| 19:30.30 | Qwell[] | copper? |
| 19:31.10 | [TK]D-Fender | datachomper: CONDUCTIVE ;) |
| 19:31.13 | datachomper | Like, cat5? |
| 19:31.26 | [TK]D-Fender | datachomper: For certain lengths of it, sure |
| 19:31.38 | datachomper | So it varies, gotcha |
| 19:33.37 | elixer | its possible to share a single d chan for multiple T1s, yeah? |
| 19:33.43 | Qwell[] | elixer: NFAS |
| 19:34.30 | elixer | Qwell: danke |
| 19:37.53 | *** join/#asterisk jcaceres (n=jcaceres@190.41.82.1) |
| 19:39.20 | holiday_42 | lol |
| 19:41.37 | CoolGuy21 | is there anything that stops 1 device from registering 2 extensions? i have a Cisco 7940 but i cannot register 2 extensions, the second one fails. |
| 19:42.14 | Qwell[] | CoolGuy21: using users.conf? |
| 19:42.35 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 19:42.45 | CoolGuy21 | let me check |
| 19:42.51 | Qwell[] | asterisk-gui? |
| 19:42.55 | jcaceres | hello i have a doubt i have successfully loaded zaptel an chan_zap.so, but when i do zap show channels i do not get any of those |
| 19:43.15 | CoolGuy21 | can i do authentication using the mac address ? |
| 19:43.17 | jcaceres | i have tried to reloaded chan_zap.so but |
| 19:43.35 | CoolGuy21 | no users.conf |
| 19:43.46 | Qwell[] | CoolGuy21: how are you configuring it? |
| 19:43.52 | jcaceres | i get this error ""No category context for line 10 of /etc/asterisk/zapata.conf"" |
| 19:43.55 | CoolGuy21 | using asterisk gui |
| 19:44.18 | jcaceres | my zapata.conf can be seen in http://pastebin.com/d331e5556 |
| 19:44.32 | jcaceres | any idea? |
| 19:47.07 | [TK]D-Fender | jcaceres: yeah... you don't have [channels] at the top of your zapata.conf |
| 19:47.11 | tzafrir_laptop | jcaceres, is that your complete zapata.conf? |
| 19:47.33 | jcaceres | thnkls |
| 19:47.39 | tzafrir_laptop | you shouldn't copy zapata-channels.conf instead of zapata.conf |
| 19:47.50 | tzafrir_laptop | you should add it to the end |
| 19:47.54 | jcaceres | i realise now |
| 19:48.07 | tzafrir_laptop | or #include zapata-channels.conf |
| 19:48.13 | tzafrir_laptop | in the end of zapata.conf |
| 19:50.00 | Qwell[] | Strom_M: ping |
| 19:50.57 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 19:51.33 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:52.47 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 19:54.39 | *** part/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 19:55.08 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 19:55.36 | chemikk | [Aug 29 16:53:16] WARNING[12799]: pbx.c:1797 pbx_extension_helper: No application 'SetGroup' for extension (trymat, 800123456, 1) |
| 19:56.05 | chemikk | SetGroup nonexists? |
| 19:56.31 | JunK-Y | chemikk: core show functions like GROUP |
| 19:56.46 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:57.03 | jsmith | chemikk: It was deprecated in favor the GROUP() dialpllan function |
| 19:58.37 | chemikk | ok thanks |
| 20:02.29 | CoolGuy21 | one of my sip extensions arnt registering, how can i check? im in CLI verbose 8 ad cant see anything |
| 20:03.06 | jfitzgibbon | coolguy21: sip debug peer <peername>, restart the useragent, and pastebin the CLI output |
| 20:03.10 | [TK]D-Fender | CoolGuy21: enable SIP DEBUG in cli. |
| 20:03.17 | CoolGuy21 | k one sec |
| 20:03.27 | CoolGuy21 | Unable to get IP address of peer '1003' |
| 20:03.28 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:04.04 | jfitzgibbon | coolguy21: then do sip debug ip x.x.x.x |
| 20:04.15 | CoolGuy21 | whos ip do i put? |
| 20:04.29 | jfitzgibbon | coolguy21: 127.0.0.1 works, or you could put - you know - the IP of the user agent |
| 20:07.28 | *** join/#asterisk Weezey (n=ohno@wan.iasloffice.iasl.com) |
| 20:07.33 | CoolGuy21 | nope too many extensions |
| 20:07.37 | CoolGuy21 | cant seee a thing |
| 20:07.53 | jfitzgibbon | coolguy21: you mean you don't know the IP of your user agent? |
| 20:08.20 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 20:08.28 | CoolGuy21 | no i have multiple extensions from that one ip |
| 20:08.30 | CoolGuy21 | like 10 |
| 20:09.24 | Weezey | I'm using a Sangoma card with hw echo cancellation and incoming calls, it works fine, outgoing calls the echo is not enabled for some reason. a zap show channel 9 gives me: Echo Cancellation: 128 taps, currently OFF but during incoming I get Echo Cancellation: 128 taps, currently ON |
| 20:09.31 | jfitzgibbon | coolguy21: then you're probably best trying to capture a trace from behind whatever is doing NAT for that user agent |
| 20:10.03 | jfitzgibbon | coolguy21: or get a bigger scrollback buffer and search for the username of that agent |
| 20:11.07 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:12.42 | *** join/#asterisk cirgal (n=robert@wsip-70-169-190-173.sb.sd.cox.net) |
| 20:16.36 | [TK]D-Fender | Weezey: are you using echo-training? |
| 20:17.02 | CoolGuy21 | hummm |
| 20:17.08 | CoolGuy21 | this is so weird i wish i can fix this |
| 20:17.23 | [TK]D-Fender | CoolGuy21: go get us some output and maybe we can help. |
| 20:17.40 | CoolGuy21 | ok tell me what u want me to do i will gladly do it |
| 20:18.19 | jfitzgibbon | coolguy21: the register only takes a few seconds, so just capture the sip debug output for the NAT IP during the attempt to register and pastebin it |
| 20:18.21 | jfitzgibbon | ~pb |
| 20:18.22 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 20:18.28 | [TK]D-Fender | CoolGuy21: I've asked several times and my answer isn't changing.. |
| 20:18.44 | CoolGuy21 | k will do right now |
| 20:18.47 | *** join/#asterisk DeepY0X (n=DeepY0X@201.240.54.73) |
| 20:20.45 | jfitzgibbon | coolguy21: also include your sip.conf with the user definition for the thing trying to register |
| 20:22.36 | *** join/#asterisk crichardson (n=crichard@38.113.5.185) |
| 20:24.04 | CoolGuy21 | that extension isnt comming up at all |
| 20:24.22 | Weezey | [TK]D-Fender: no, should I be? |
| 20:25.16 | jfitzgibbon | coolguy21: then like I suggested before, you need to watch the traffic behind whatever is doing NAT |
| 20:25.18 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-14-82.lns4.syd7.internode.on.net) |
| 20:25.25 | CoolGuy21 | k |
| 20:25.27 | jfitzgibbon | coolguy21: you've got one of two problems (if you're lucky) |
| 20:25.33 | [TK]D-Fender | Weezey: Nope. pastebin your wanpipe1.conf (or whichever is appropriate), and your zapata.conf |
| 20:25.35 | [TK]D-Fender | FAST |
| 20:25.39 | [TK]D-Fender | I'm here for 5 mins :) |
| 20:25.44 | Weezey | k |
| 20:25.44 | CoolGuy21 | jfitzgibbon which are? |
| 20:25.46 | jfitzgibbon | coolguy21: either the REGISTER isn't getting to your * or the REGISTER contains invalid info |
| 20:26.19 | *** join/#asterisk havarian (n=amr_emam@205.189.149.240) |
| 20:26.46 | [TK]D-Fender | jfitzgibbon: No, invalid should show up, even as crap. Its jsut NOT THERE. |
| 20:27.25 | Weezey | [TK]D-Fender: http://www.pastebin.ca/675091 |
| 20:28.14 | jfitzgibbon | [TK]D-Fender: I was allowing for the search to have failed for various reasons. But we're all assuming, but there has been no pastebin of the debug output |
| 20:28.22 | Weezey | [TK]D-Fender: I tried adding line 91 last time because that's the channel I was testing on |
| 20:28.25 | [TK]D-Fender | Weezey: Ok, looks perfectly legit. Now aside from * reporting no EC... are you getting echo? |
| 20:28.32 | Weezey | yes |
| 20:28.39 | Weezey | when I call out only |
| 20:28.47 | Weezey | when I receive a call in, it's on and there's no echo |
| 20:28.51 | [TK]D-Fender | Weezey: I would suggest checking with Sangoma support on this one... |
| 20:28.58 | Weezey | thanks |
| 20:29.33 | [TK]D-Fender | ok, checkout time |
| 20:29.36 | [TK]D-Fender | BBAIB |
| 20:32.30 | CoolGuy21 | how can i do authentication of extension using mac address? |
| 20:34.57 | jsmith | You can't |
| 20:35.21 | *** join/#asterisk galeras (i=galeras@200.21.36.237) |
| 20:36.00 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 20:36.11 | galeras | Please, someone can tell me if 2.0.1.0291 is the last sip firmware version for Polycom? |
| 20:36.26 | kolian123 | Russellb, hi |
| 20:36.35 | wunderkin | galeras, no... way off, www.polycom.com can tell you |
| 20:39.05 | galeras | wunderkin: sorry, i mean if is the "latest version" this information isn't at polycom site |
| 20:39.33 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 20:39.49 | kombi | how does one see all the CLI commands again..? |
| 20:39.51 | lirakis | l8r all |
| 20:39.55 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:39.59 | wunderkin | galeras, BS |
| 20:40.42 | russellb | kolian123: hello |
| 20:40.53 | kombi | core show cli commands? |
| 20:41.10 | kolian123 | Russel, would you like to take a look at core? |
| 20:41.31 | russellb | kolian123: gdb /usr/sbin/asterisk core.12345 .... post it to pastebin.ca |
| 20:41.43 | kolian123 | one sec |
| 20:41.57 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 20:42.47 | kombi | can you kill a conference from manager or cli? |
| 20:44.53 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:45.04 | kolian123 | Russellb, http://www.pastebin.ca/67510 |
| 20:45.09 | kolian123 | thank you |
| 20:45.23 | _mm_ | z/wi _mm_ |
| 20:45.34 | Qwell[] | kolian123: wrong one |
| 20:45.48 | *** part/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) |
| 20:46.06 | kolian123 | one sec |
| 20:47.50 | pnlarsson | If i want to cp my menuselect options from one build to another - which file do i cp? |
| 20:48.09 | kombi | your dialplan? |
| 20:48.25 | Qwell[] | pnlarsson: menuselect.makeopts |
| 20:49.37 | kolian123 | Qwell, http://www.pastebin.ca/675116 |
| 20:49.42 | kolian123 | thanks |
| 20:49.56 | kolian123 | sorry, missed a number |
| 20:50.31 | kombi | to kick everyone off a conference via manager, what does one do? |
| 20:50.34 | pnlarsson | Qwell[], thanks |
| 20:53.35 | jfitzgibbon | I just had nearly all my queue members go into a status of "Unknown" (in the output of 'queue show'). http://pastebin.com/m201b2c49. I had to restart *, dumping all my waiting callers. Anyone ever seen this before? I have once, but I couldn't reproduce it in the lab. |
| 20:54.57 | kolian123 | kombi, maybe hangup? |
| 20:56.26 | *** join/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) |
| 20:57.38 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 20:57.44 | cirgal | with respect to parking a call: does anyone know whether the slot the call got parked into is set into some variable or is it the case that the digits are spoken only? |
| 20:59.35 | chemikk | 1 > 1 is false no? :) |
| 21:00.33 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 21:01.03 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.183) |
| 21:01.11 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
| 21:01.18 | Qwell[] | chemikk: Unless you redefine 1 mid-check |
| 21:02.41 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 21:02.54 | luke-jr | chemikk: not necessarily if you're using floats ;) |
| 21:03.06 | mvanbaak | redefine!!!!!!!!!!!!!! |
| 21:03.09 | mvanbaak | runkit_constant_redefine - Redefine an already defined constant |
| 21:03.21 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 21:04.15 | luke-jr | well this sucks-- looks like the PAP2-NA is impossible to unlock |
| 21:05.02 | mvanbaak | that will unlock it |
| 21:05.10 | chemikk | <PROTECTED> |
| 21:05.13 | chemikk | <PROTECTED> |
| 21:05.43 | chemikk | why goto to 20? , 1 > 1 is false no? |
| 21:05.50 | mvanbaak | I bet the first 1 is 1.something |
| 21:05.51 | mvanbaak | ;) |
| 21:06.11 | luke-jr | 20 is after 10 |
| 21:06.37 | mvanbaak | 1+1 = 3 for big values of 1 |
| 21:06.46 | chemikk | i dont understand |
| 21:06.58 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 21:07.11 | Qwell[] | "[ 1 > 1 ]" is a string, and will always evaluate to non-zero |
| 21:07.23 | Qwell[] | perhaps you mean $[ 1 > 1 ] ? |
| 21:08.04 | chemikk | yes |
| 21:08.04 | mvanbaak | what Qwell[] said |
| 21:08.04 | chemikk | thanks |
| 21:13.45 | *** part/#asterisk pgarcia (n=pedgarci@c934ac83.virtua.com.br) |
| 21:17.38 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.105.176) |
| 21:22.01 | lbow | corydon: are you there? |
| 21:25.35 | lbow | corydon: http://bugs.digium.com/view.php?id=10549 (again, I know). Put up a patch to make an MSet app with the semantics of Set as it was before |
| 21:25.37 | *** join/#asterisk klictel (n=klictel@atelka.info) |
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| 21:55.38 | cirgal | so, anyone experts on call parking? |
| 21:59.36 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 22:00.59 | *** join/#asterisk bacs (n=bacs@flunge.gladserv.com) |
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| 22:03.57 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 22:05.35 | *** join/#asterisk nephfl (n=no@wsip-68-110-130-57.ga.at.cox.net) |
| 22:06.29 | nephfl | im wondering, why cant a polycom "buddy watch" plus a command to zap barge with the key next to the buddy simulate a key system? |
| 22:11.01 | *** part/#asterisk havarian (n=amr_emam@205.189.149.240) |
| 22:12.29 | elixer | is it common to see "-- B-channel 0/1 successfully restarted on span 1" periodically in your console output? |
| 22:13.33 | fujin | anyone familiar with 'in use' detection |
| 22:13.55 | blitzrage | elixer: yes, that is normal |
| 22:14.01 | elixer | blitzrage: thanks. |
| 22:14.27 | nephfl | i can detect "in use" |
| 22:19.15 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-48-125.dhcp.insightbb.com) |
| 22:25.34 | *** join/#asterisk JoseBravo (n=jbravo@190.156.225.15) |
| 22:26.13 | JoseBravo | I have analog card, and asterisk didn't determine when the caller hangup. Any idea? |
| 22:26.16 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 22:26.44 | *** join/#asterisk particle (n=alex@user-0cev8be.cable.mindspring.com) |
| 22:27.42 | *** join/#asterisk powerkill (n=powerkil@84.205.154.247) |
| 22:27.55 | Nugget | welcome to the world of analog PSTN. |
| 22:28.09 | particle | analog? eh! |
| 22:28.16 | powerkill | hi |
| 22:28.23 | powerkill | anthm are you there ? |
| 22:30.42 | elixer | could someone explain overlap dialing to me like i'm a 5 year old? |
| 22:32.22 | fujin | that depends |
| 22:32.24 | fujin | what's overlap dialing? |
| 22:32.26 | elixer | heh |
| 22:32.56 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 22:33.27 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 22:36.02 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 22:36.08 | cirgal | anyone know why ast_waitfor() might wait 0 ms instead of what i tell it to wait? |
| 22:36.40 | cirgal | is this the right channel for coding questions as well as config? :) |
| 22:37.27 | cirgal | ........... silence ............. |
| 22:37.34 | elixer | cirgal: i think there is an #asterisk-dev channel as well? |
| 22:37.40 | cirgal | ah |
| 22:37.47 | cirgal | elixer: thanks :) |
| 22:37.52 | elixer | np |
| 22:41.32 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 22:42.41 | [hC] | Im curious, when i have an exten that does a Goto() to a new context, and in that context there is an 'h' exten, why would it not execute once the call is hung up on? |
| 22:43.25 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 22:44.26 | generalhan | hey guys ... im running a zttest right now and im seeing a lot of 99.975 and 99.963 ... how, not good is that ? |
| 22:45.26 | [hC] | generalhan: ive seen that on boxes that worked fine, ive also seen it on boxes that were really bad. im not sure how much i trust zttest to be relative to results. |
| 22:45.34 | [hC] | generalhan: then again i was using sangoma. |
| 22:45.41 | [hC] | but, it shouldnt matter. |
| 22:45.50 | generalhan | hmm |
| 22:46.09 | generalhan | well i have a lot of users comming up to me right now, saying that people are complaining that they are "cutting in and out" |
| 22:46.28 | generalhan | so thats when i ran the zttest. typically the lowest ill see all day is a 99.987 |
| 22:47.54 | [hC] | check irq usage/io on the box |
| 22:47.58 | [hC] | maybe move it to a new irq/slot |
| 22:48.14 | generalhan | hmm, dont think i know how to do that ! |
| 22:49.57 | [hC] | nobody knows about this 'h' extension stuff, when using goto/ |
| 22:49.59 | *** join/#asterisk remmo (n=junk@203.25.123.250) |
| 22:54.00 | SplasPood | generalhan: cat /proc/interrupts |
| 22:56.39 | *** join/#asterisk didge (n=mcveighj@bas2-barrie18-1242454602.dsl.bell.ca) |
| 22:56.43 | didge | hi. |
| 22:57.16 | didge | i want to connect a plain old telephone line to my asterisk computer. what hardware do i need ? |
| 22:57.43 | generalhan | well i just took a look at it ... and although im not an expert, by any means, i dont see any issues: http://generalhan.pastebin.ca/675256 |
| 22:58.08 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:59.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:01.01 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 23:01.52 | [hC] | Hey guys... http://pastebin.ca/675255 - know why 'h' would never be reached here? |
| 23:07.39 | flenders | didge: you need an analogue card, like digium's TDM400P or sangoma's A200 |
| 23:08.30 | flenders | didge: if it's a single line, you could also use an ATA with FXO port, like the linksys/sipura SPA3000 or SPA3102 |
| 23:10.05 | didge | flenders; thank you. |
| 23:11.08 | *** join/#asterisk nichtwirklich (n=guess@88.134.54.113) |
| 23:11.13 | nichtwirklich | hi all |
| 23:12.23 | JoseBravo | I have analog card, and asterisk didn't determine when the caller hangup. I can do something? |
| 23:14.18 | nichtwirklich | I am hanging with dtmf from sip phone through asterisk with an isdn adapter to the outside world |
| 23:15.23 | nichtwirklich | dtmf works intern on the asterisk box, and dtmf works from an isdn phone through asterisk (2nd adapter in nt mode) through to world via isdn te |
| 23:15.59 | nichtwirklich | but from my snom 360 I cannot send dtmf to the world, I use inband and alaw/ulaw, any ideas? |
| 23:18.40 | *** part/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com) |
| 23:19.07 | flenders | nichtwirklich: try dtmfmode rfc2833 |
| 23:19.35 | nichtwirklich | flenders, I give it a try, but I used auto already where rfc should be the first |
| 23:21.48 | nichtwirklich | flenders: no |
| 23:22.19 | flenders | nichtwirklich: no what? |
| 23:23.36 | nichtwirklich | flenders: doesn't work |
| 23:23.38 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:24.00 | nichtwirklich | flenders: do you have a similar config? |
| 23:24.38 | flenders | I don't know what your config is |
| 23:25.10 | nichtwirklich | flenders: I mean the hardware, ISDN line from your provider, asterisk, sip phone |
| 23:26.05 | flenders | what card do you have? |
| 23:26.11 | flenders | I have a similar setup yes |
| 23:26.28 | flenders | but I have a PRI, and a bunch of linksys phones |
| 23:26.49 | *** join/#asterisk grimsy (n=chatzill@203.14.171.102) |
| 23:27.19 | nichtwirklich | freebsd, chan_capi, 2 x hfc adapter (nt and te), some different voip phones and one isdn phone |
| 23:28.10 | flenders | sorry, our setups are very different. |
| 23:28.16 | nichtwirklich | so it's no particular snom problem, it doesn't work with an nokia e61 or an elmeg as well |
| 23:29.15 | nichtwirklich | I dont think so, actually the problem seems to be in forwarding dtmf from sip to a "real" phone line, pri or bri shouldn't matter |
| 23:30.24 | flenders | can you send DTMF using the ISDN phone? |
| 23:31.05 | nichtwirklich | yes |
| 23:31.26 | nichtwirklich | and intern with the sip phone works too (mailbox) |
| 23:32.06 | nichtwirklich | and from extern via isdn works also (mailbox / callthrough from cell phone) |
| 23:36.04 | flenders | nichtwirklich: can you pastebin your sip.conf |
| 23:36.25 | nephfl | anybody very familiar with polycom and the auto answer feature? |
| 23:37.44 | nichtwirklich | disallow=all ; First disallow all codecs |
| 23:37.45 | nichtwirklich | allow=alaw |
| 23:37.45 | nichtwirklich | allow=ulaw |
| 23:37.56 | nichtwirklich | dtmfmode = rfc2833 |
| 23:38.40 | [TK]D-Fender | nephfl, Yes, its all very well documented on the WIKI. What about it? |
| 23:39.03 | nichtwirklich | I think thats it for this problem |
| 23:39.42 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 23:39.43 | nephfl | i wouldnt say it is well documented anywhere...but if you know a wiki i missed...please help me out, because google isnt helping much |
| 23:40.10 | heelios | nephfl: http://www.voip-info.org/wiki/ |
| 23:42.52 | [TK]D-Fender | nephfl, So what part are you stuck on? |
| 23:44.21 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 23:45.38 | quah | anyone here use "announce-holdtime"? I have it turned on,but it only announces "less than 2 minutes" if hold was less than 2 minutes. Anyway to make is more granular - like say 15 seconds if holdtime was 15 seconds? |
| 23:45.43 | flenders | nichtwirklich: I asked you to pastebin your sip.conf, not just a few lines of it |
| 23:45.53 | flenders | ~pn |
| 23:45.53 | jbot | Probably Never |
| 23:45.55 | flenders | ~pb |
| 23:45.56 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 23:47.48 | nichtwirklich | flenders: here it comes |
| 23:48.11 | nichtwirklich | [general] |
| 23:48.11 | nichtwirklich | context=sipintern ; Default context for incoming calls |
| 23:48.11 | nichtwirklich | bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) |
| 23:48.11 | nichtwirklich | bindaddr=192.168.8.9 |
| 23:48.11 | nichtwirklich | srvlookup=yes ; Enable DNS SRV lookups on outbound calls |
| 23:48.20 | nichtwirklich | autodomain=yes |
| 23:48.27 | nichtwirklich | checkmwi=10 |
| 23:48.33 | nichtwirklich | disallow=all |
| 23:48.40 | nichtwirklich | allow=alaw |
| 23:48.40 | nichtwirklich | allow=ulaw |
| 23:48.46 | nichtwirklich | musicclass=default |
| 23:48.46 | jsmith | nichtwirklich: Use the pastebin! Don't flood the channel! |
| 23:48.47 | flenders | nichtwirklich: PASTEBIN! |
| 23:49.09 | nichtwirklich | oh sorry |
| 23:49.10 | flenders | read what jbot says! |
| 23:49.13 | flenders | ~pb |
| 23:49.14 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 23:49.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:50.15 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 23:51.35 | fujin | anyone run SPA942's? |
| 23:51.44 | fujin | need to know if they can subscribe to multiple mailboxes |
| 23:51.51 | fujin | or if asterisk can generate a mwi to the phone |
| 23:52.07 | flenders | what's a mwi? |
| 23:52.32 | Nugget | message waiting indicator |
| 23:52.53 | nichtwirklich | flenders: sorry, here it is: http://pastebin.com/d3fe98409 |
| 23:52.54 | fujin | I can see asterisk sending NOTIFY messages to my phone |
| 23:53.04 | fujin | so I assume this should be customizable (for multiple phones) |
| 23:53.08 | CrashSys | I'm amazed that shoretel has no information regarding their licenses on their website. |
| 23:54.22 | fujin | can you have dual 'mailbox' settings in sip.conf for a device? |
| 23:54.30 | CrashSys | No |
| 23:54.54 | fujin | hrm |
| 23:54.59 | fujin | looks like I can mailbox=x, x |
| 23:55.29 | *** join/#asterisk Rospo (i=Geo@202.189.78.66) |
| 23:55.49 | fujin | hmm, that'll work |
| 23:55.49 | fujin | :] |
| 23:55.53 | CrashSys | New one on me |
| 23:56.18 | fujin | awesome |
| 23:56.22 | fujin | getting an MWI for both of my accounts |
| 23:56.34 | fujin | now if only the phone could display *which* mailbox it was for. |
| 23:57.19 | CrashSys | Maybe the phone supports dual-registrations? |
| 23:57.20 | flenders | nichtwirklich: did you try other dtmf modes too? |
| 23:57.22 | CrashSys | might work that way |
| 23:58.05 | CrashSys | Does a shoretel license have a renewal or is it more like microsoft's license? |
| 23:58.17 | CrashSys | Trying to put together a bid against a shoretel guy. |
| 23:58.44 | fujin | CrashSys: it does, but I don't plan on making the phones register to a "group" sip account. |
| 23:59.04 | fujin | I have group mailboxes setup for queues, and want to make it so that the devices have visibility of their own mailbox and the group mailbox |
| 23:59.07 | [TK]D-Fender | Polycom > ALL |
| 23:59.39 | *** join/#asterisk mjmarrio (n=mike@219-90-205-152.static.adam.com.au) |
