| 00:15.18 | *** join/#asterisk danfromuk (~IceChat77@2.27.1.186) |
| 00:15.55 | danfromuk | Hi. When asterisk uses odbc for voicemail, does it store the voicemail greetings in odbc too? |
| 00:16.05 | danfromuk | the answer seems to be no, but is that correct? |
| 00:19.17 | Nugget | it does store the greetgins in odbc too |
| 00:19.59 | danfromuk | ah, i see the problem. i just migrated to a new sql server and it didnt take the records with it. |
| 00:20.04 | danfromuk | thanks |
| 00:25.38 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 00:26.57 | danfromuk | ok, now i've got a problem. some users have already received voicemail, so the new server already has some records. i noticed there is a msgnum column. any idea what would happen if there are two records with the same msgnum? |
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| 01:00.09 | pcangel | I keep getting retransmissions and then sip dialog distructions, with sipdebug on I noticed this "To: <sip:9001@192.168.0.18:5060>;tag=as11f75136" -- I think it's sending the LAN IP address of my server to the remote server, even though I have localnet=192.168.0.0/255.255.255.0 and externip=my.external.ip |
| 01:00.17 | pcangel | can anybody here help me resolve this? |
| 01:06.01 | thecardsmith | pcangel: do you have a "nat=yes" under your [client_device] section in sip.conf? |
| 01:06.55 | thecardsmith | if it's behind a nat... i assume too much |
| 01:07.47 | thecardsmith | (i just have been in a nat hell, so, yeah, that's what i'm thinking about, hah) |
| 01:10.21 | pcangel | I had it in [general] in sip.conf, does that apply it to all inbound? |
| 01:10.28 | pcangel | this is on accepting an inbound fax |
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| 01:11.34 | pcangel | I commented out the device entry for the remote client that is initiating the session, in [general] I have allowguests=yes canreinvite=yes qualify=yes insecure=very directrtpsetup=yes |
| 01:11.51 | pcangel | and nat=yes |
| 01:19.14 | thecardsmith | in the general section should be good, hrmmm |
| 01:21.18 | pcangel | I put on sipdebug on the other server (the client in this case) and it's not receiving any messages regarding 192.168.0.18 - but it is having the same retransmitting messages (#1-#5) |
| 01:21.53 | pcangel | on the other server there is a NAT, but it is behind a different interface, the interface asterisk is communicating out on is not NAT'd |
| 01:22.24 | pcangel | I still have canreinvite=yes, nat=yes, qualify=yes on it |
| 01:22.34 | pcangel | because I didn't think it'd hurt to keep those enabled |
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| 01:49.31 | *** join/#asterisk boch (~boch@190.174.9.252) |
| 01:49.34 | boch | hi all |
| 01:50.03 | boch | do you know how to make Asterisk reply to Contact or Via header instead address in From header? |
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| 03:11.44 | AlexNeuman | Good Evening! / Buenas Noches! |
| 03:17.36 | AlexNeuman | Does anybody know if the "attended transfers where the third party hangs up but instead of getting the first and second party to talk again the call gets hung up" problem present in some versions of asterisk has been fixed or a workaround? :-D |
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| 03:35.15 | *** join/#asterisk BeeBuu (3afc49a4@gateway/web/freenode/ip.58.252.73.164) |
| 03:36.00 | BeeBuu | any one teach me how can i get "Event: ExtensionStatus" in AMI? |
| 03:46.36 | LiuYan | do you mean how to get the event or the string? |
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| 03:55.27 | BeeBuu | no, |
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| 03:56.07 | BeeBuu | when i send a command like follow: "action: ExtensionState context: from-internal exten: 800" |
| 03:56.20 | BeeBuu | i get -1 |
| 03:56.33 | BeeBuu | but my extensions.conf get 800 exten |
| 03:56.56 | BeeBuu | i don't know what's wrong this |
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| 04:10.02 | p3nguin | If there are no channels using extension 800, what would you expect the state of extension 800 to be? |
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| 04:26.13 | BeeBuu | p3nguin: i had tried ring it |
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| 04:28.22 | BeeBuu | LiuYan: 刘燕，你知道吗？ |
| 04:31.29 | drmessano | {} }{ ]{ {} }{ ]{ |
| 04:34.24 | BeeBuu | drmessano: can you help me? |
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| 04:43.37 | din3sh | mrning all |
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| 05:25.19 | mattems | hey all! |
| 05:26.57 | din3sh | hey |
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| 06:03.47 | jaxon007_ | I have problem in asterisk PRI's, I have found another which has exactly same problem like me. But his post is yet unanswered on asterisk forum. I am pasting link here http://forums.digium.com/viewtopic.php?f=1&t=82827 |
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| 06:14.25 | drmessano | jaxon007, you are using the FoneBridge's as well? |
| 06:14.50 | jaxon007 | drmessano: yes |
| 06:16.11 | drmessano | Regardless of the other pieces here, this seems to boil down to "I have FoneBridges and they drop at random".. Have you tried the manufacturer? |
| 06:17.10 | drmessano | Check this out.. |
| 06:17.11 | drmessano | http://lists.digium.com/pipermail/asterisk-users/2008-May/211365.html |
| 06:17.16 | drmessano | jaxon007, ^^ |
| 06:18.36 | drmessano | This as well.. Same thread.. http://web.archiveorange.com/archive/v/8Et0ZjtN0VA9zsS8Y5zN |
| 06:19.28 | drmessano | jaxon007, near the bottom of that last link... Thank you very much for your suggestions. Yesterday evening, the telco |
| 06:19.30 | drmessano | technician disabled a feature they call NT1 on their side, which is |
| 06:19.30 | drmessano | supposed to be a protocol for line monitoring. We tested again and were |
| 06:19.30 | drmessano | unable to reproduce the call dropping problem. |
| 06:19.30 | drmessano | OOPS |
| 06:24.46 | jaxon007 | drmessano: I called my TELCO operator to check the NT1 settings on thier side, But same setup is running with another asterisk box running in good condition |
| 06:27.36 | drmessano | At the same location? |
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| 06:35.44 | pbxMAN | hello |
| 06:36.47 | drmessano | olleh |
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| 06:38.31 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:38.35 | schmidts | good monring |
| 06:44.06 | kleszcz | moring |
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| 06:46.19 | *** join/#asterisk din3sh (~din3sh@41.212.203.245) |
| 06:46.26 | din3sh | hey all again |
| 06:46.35 | jacc0 | morning |
| 06:50.24 | jaxon007 | drmessano:Yes, At same location and sorry for my late response :) |
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| 06:55.04 | din3sh | Jacc0: have you deployed polycom ip 330 phones ? |
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| 06:55.49 | jacc0 | nope |
| 06:56.08 | jacc0 | I've got this message: |
| 06:56.10 | jacc0 | [May 24 14:26:47] WARNING[25758]: channel.c:3578 ast_waitfordigit_full: write() failed: Resource temporarily unavailable |
| 06:56.23 | jacc0 | when my agi is waiting for incomming dtmf |
| 06:56.38 | jacc0 | any clue what this is about? |
| 06:56.51 | jacc0 | I don't get it all the time |
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| 06:59.36 | jacc0 | I'll have a look at ; channel.c:3578 |
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| 07:09.44 | jacc0 | hmm, still no clue |
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| 07:15.12 | din3sh | jacc0: sorry no clue either with that one |
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| 07:32.54 | din3sh | jacc0: is it possible to dial alphabets from a sip phone? |
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| 07:38.01 | drmessano | Depends on the phone |
| 07:38.38 | drmessano | Most softphones will let you dial a URI like sip:yomomma@myhouse.com |
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| 07:52.49 | jacc0 | din3sh: not only is it posible to dial alphabets; you can also dial all chars needed to perform SQL injection :) |
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| 07:57.29 | mikunos | hi guys which kind of modem fax device have I to buy to use asterix? |
| 07:57.50 | mikunos | I am new of Asterix |
| 08:00.01 | kaldemar | you don't need a fax modem to use asterisk. actually, you better not try to use a fax modem for anything. |
| 08:00.23 | kaldemar | all you need is a NIC to use VoIP. |
| 08:00.47 | mikunos | kaldemar so, how have I to connect the telephone line to the pc? |
| 08:03.47 | jacc0 | you could find a SIP-provider that will take care of that or you could use a digium card |
| 08:06.11 | mikunos | http://store.digium.com/productview.php?product_code=1TDM410PLF |
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| 08:20.10 | jacc0 | mikunos: yes, something like that |
| 08:22.52 | roswell | hi channel. is there a way of setting TIMEOUT(digit) globally so * would use its value in every extension? |
| 08:24.10 | kaldemar | roswell: set it under [globals] in extensions.conf. |
| 08:24.46 | roswell | kaldemar, i've tried that, somehow it doesn't help |
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| 08:54.39 | shadebob | A user of the Vestec ASR in the room? |
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| 09:56.30 | CommaCrazy | how many simultaneous calls can ISDN pri handle |
| 09:56.34 | CommaCrazy | does any one know |
| 09:57.24 | bulkorok | depends on your connection... |
| 09:58.16 | bulkorok | standard ISDN has 2 lines... PRI depends on country... US has 24 europe has 30 lines |
| 09:59.11 | CommaCrazy | 30 lines |
| 09:59.19 | kaldemar | 2 channels for BRI and 24 for PRI over T1 and 30 for PRI over E1. |
| 09:59.41 | CommaCrazy | so 30 simultanious calls |
| 10:00.11 | CommaCrazy | thank you :) |
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| 10:13.41 | WIMPy | But you can have more calls than you have channels. |
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| 11:52.27 | danfromuk | Hi, I know this is a little OT but if I have a 2Mb upload speed, can I run 13 concurrent calls? |
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| 11:52.47 | danfromuk | Using the GSM codec |
| 11:53.11 | danfromuk | How can I calculate the number of calls I can successfully make without running into problems? |
| 11:53.47 | WIMPy | ask google for a voip bandwidth calculator |
| 11:55.33 | danfromuk | What codec is GSM listed as? I cant find it in the list on the calculators online. |
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| 11:57.31 | WIMPy | The dusty GSM one ia 13kbit. |
| 11:57.45 | Gugge | http://www.asteriskguide.com/bandcalc/bandcalc.php <- i can see gsm there |
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| 12:03.29 | danfromuk | Dusty? What codec would you recommend? With asterisk 1.8, we seemed to have problems using other codecs when using voicemail. The final beep was getting 'stuck' and continued to play causing the voicemessages to be distorted. |
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| 12:05.14 | schmidts | danfromuk do you store voicemails over nfs? |
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| 12:08.20 | WIMPy | Well, the GSM codec you find in Asterisk and in phones is the one that probaly hasn;t been used in the GSM for over 10 years. |
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| 12:13.17 | danfromuk | No. ODBC |
| 12:13.40 | schmidts | danfromuk i mean the recored file ;) |
| 12:13.43 | danfromuk | I tried alaw which worked find in 1.4 but when i upgraded to 1.8, it caused problems. |
| 12:14.00 | danfromuk | the voicemessages are also stored in odbc |
| 12:14.07 | schmidts | ok |
| 12:14.31 | schmidts | do you start asterisk with the "-t" option? |
| 12:15.02 | schmidts | cause it sounds like the typical problem you will have when storing voicemails over nfs, that they get disordered (speed up) after 30 seconds |
| 12:15.29 | schmidts | when using the -t option, asterisk will store them in the tmp dir until its finished and copy it then to the final directory |
| 12:15.50 | danfromuk | However that wont resolve the continious beep that is left in the audio stream. |
| 12:16.00 | danfromuk | The caller doesnt know when to start speaking |
| 12:16.24 | schmidts | danfromuk ok sorry didnt understood your problem right, i thought you only have these beeps in the recording |
| 12:30.19 | *** join/#asterisk samba35 (~shrikant@unaffiliated/samba35) |
| 12:30.50 | samba35 | is there any free sip service provider or where i can do testing of my test voip setup |
| 12:31.10 | samba35 | i am sorry i am new to voip and asterisk |
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| 12:34.07 | CommaCrazy | WIMPy, yeah you could send over 30 simultaneous calls over ISDN PRI (E1) but only on certain codec compressions |
| 12:35.04 | WIMPy | CommaCrazy: No. Unless you want to do something like VOIP over dialup. |
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| 12:35.25 | WIMPy | But If you put a call on hold, you can place another call on the same channel. |
| 12:35.36 | CommaCrazy | smart man :) |
| 12:38.49 | WIMPy | Probaly the number of calls is only limited by the range of the call reference. |
| 12:39.35 | WIMPy | That should make 127 for BRI and probably 16383 for PRI, each coming and going. |
| 12:39.40 | WIMPy | Unlikely to be usefull :-) |
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| 12:44.36 | [TK]D-Fender | samba35, You don't need an external service to "test" your system. Even a softphone is fine. just point it to where your incoming calls would go instead of wherever you'd point it normally |
| 12:44.56 | samba35 | ok |
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| 14:00.08 | pabelanger | Working with a Bell tech to test an Avaya IPO and asterisk |
| 14:00.12 | pabelanger | should be interesting |
| 14:01.06 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:03.47 | [TK]D-Fender | pabelanger, Connected how? |
| 14:03.56 | pabelanger | via SIP |
| 14:04.10 | [TK]D-Fender | Cool, I'd like to know what setting end up working for you... |
| 14:04.19 | [TK]D-Fender | my ehad office has an IPO |
| 14:04.22 | [TK]D-Fender | head* |
| 14:05.04 | pabelanger | Right now I'm just creating a basic SIP peer, no authentication to start. Waiting for the bell guy to send a call down |
| 14:05.23 | pabelanger | SIP OPTIONS messages are being passed though |
| 14:08.00 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:08.00 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:18.16 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust1000.sgyl.cable.virginmedia.com) |
| 15:19.09 | Greenlight | Howdy folks. Seem to be having some issues with DTMF - my voip provider claims asterisk is broken and doesn't correctly sent DTMF as payload type 96, but uses 101 instead. Is this correct, and is there a work around? |
| 15:20.02 | [TK]D-Fender | 101 is the standard for RFC2833 |
| 15:20.33 | Greenlight | So why would they only accept payload type 96, seems backwards? |
| 15:21.16 | Qwell | wow |
| 15:21.50 | leifmadsen | Greenlight: probably their old and busted software/hardware they are using |
| 15:22.15 | leifmadsen | and of course they will "blame" the other software |
| 15:22.19 | Greenlight | They claim their hardware provider wont alter cause it's a known bug with asterisk ;/ |
| 15:22.29 | *** join/#asterisk angryuser_laptop (~angryuser@83-230-135-95.pool.ukrtel.net) |
| 15:22.37 | leifmadsen | "known bug" |
| 15:22.45 | leifmadsen | easy way to get out of doing work |
| 15:23.09 | Greenlight | I can't see one way or the other in http://www.ietf.org/rfc/rfc2833.txt |
| 15:23.38 | Greenlight | 96 is an "Acceptance tone", 101 is "Line lockout tone" |
| 15:24.06 | leifmadsen | just alter the code to send 96 instead of 101 :D |
| 15:24.14 | leifmadsen | has no idea if that'll work at all |
| 15:24.19 | Greenlight | ;) |
| 15:24.25 | leifmadsen | I'm not an rfc2833 expert |
| 15:24.26 | Greenlight | And to receive as well :) |
| 15:24.50 | Greenlight | I'm always hesitent to alter code, as next time i upgrade i'll likely forgot what I altered |
| 15:25.07 | leifmadsen | that's why you keep a separate repo with your patches to apply |
| 15:25.22 | leifmadsen | check out new code, apply patches, deploy |
| 15:25.26 | Greenlight | Hmm... That would be very cool |
| 15:25.33 | leifmadsen | that's the development way... |
| 15:25.59 | leifmadsen | just keep your changes somewhere and know you need to apply them before upgrading your test environment |
| 15:26.04 | Greenlight | I've a ton of custom modifications I'd like to apply as well |
| 15:26.15 | leifmadsen | so just keep them somewhere and apply them each time |
| 15:26.31 | Greenlight | ANy decent guides on how to set such a thing up? |
| 15:26.42 | leifmadsen | however you want to manage your files |
| 15:26.53 | leifmadsen | keep them in subversion, or a folder on the system |
| 15:26.56 | leifmadsen | back them up somewhere |
| 15:26.59 | leifmadsen | that's about it |
| 15:27.10 | leifmadsen | a guide isn't really mandated for this |
| 15:27.49 | Greenlight | Sorry I'm a bit slow at this - it's not something I've worked with before. So say I take a source file, then make a change, how to I make that into a 'patch' file ? |
| 15:28.17 | leifmadsen | if it's subversion, just: svn diff > /tmp/my-new-patch.diff |
| 15:28.26 | Greenlight | I think I've been spoiled with VS and TFS |
| 15:28.36 | leifmadsen | then to apply back to an unaltered code, just do: patch -p0 < /tmp/my-new-patch.diff |
| 15:28.36 | Greenlight | wow - that's easy |
| 15:28.51 | leifmadsen | ya, that's why I said a guide wasn't really necessary |
| 15:28.55 | Greenlight | Indeed :) |
| 15:28.57 | leifmadsen | you're just creating and apply patches |
| 15:29.15 | leifmadsen | just look on google how to do that with whatever system you use (with asterisk, we use subversion) |
| 15:29.18 | Greenlight | Wow this opens loads of doors for things I've been putting off |
| 15:29.28 | Greenlight | Might as well use svn as well then |
| 15:29.32 | leifmadsen | if you use git (and github) it can get a little easier and harder at the samet ime |
| 15:29.48 | leifmadsen | I think someone maintains a git mirror somewhere |
| 15:29.55 | file | tzafrir does |
| 15:30.00 | file | ong gitorious |
| 15:30.02 | file | erm on |
| 15:30.27 | leifmadsen | and now, I lunch |
| 15:30.44 | Greenlight | Hmm, but I could just continue to grab the latest version as a tar.gz, extract, and then apply my .diff files using svn |
| 15:30.53 | leifmadsen | yes |
| 15:30.56 | leifmadsen | not with svn |
| 15:30.58 | Greenlight | Cool |
| 15:31.02 | leifmadsen | you could apply your patches, yes |
| 15:31.06 | leifmadsen | because 'patch' is separate of svn |
| 15:31.14 | leifmadsen | you could also use: diff -u |
| 15:31.22 | leifmadsen | which is basically all svn is runnign when you do "svn diff" |
| 15:31.28 | leifmadsen | except the diff is against whatever you checked out |
| 15:31.34 | Greenlight | Ah right so "svn diff > /tmp/patch.diff" would need svn tied into the master svn? |
| 15:31.37 | leifmadsen | you'd have to supply the original and changed files with 'diff' |
| 15:31.44 | Greenlight | Ahhh |
| 15:31.52 | leifmadsen | svn co http://svn.asterisk.org/svn/asterisk/tags/10.3.0 |
| 15:31.57 | leifmadsen | make change there, run svn diff |
| 15:32.06 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 15:32.16 | leifmadsen | otherwise, if you're using tarballs, you need the original and the changed files, and you can diff -u against the original and changed versions |
| 15:32.28 | leifmadsen | tar zxvf asterisk-10.3.0.tar.g.z |
| 15:32.28 | Greenlight | Wow - it really does seem simple, don't know why I've been avoiding it all this time |
| 15:32.44 | leifmadsen | cp -r 10.3.0 10.3.0-altered |
| 15:32.44 | Greenlight | Many thanks leifmadsen :) And enjoy your lunch :) |
| 15:33.17 | leifmadsen | diff -u 10.3.0/channels/chan_sip.c 10.3.0-altered/channels/chan_sip.c > /tmp/my.patch |
| 15:33.28 | leifmadsen | something like that anyways |
| 15:33.30 | leifmadsen | runs away |
| 15:33.31 | Greenlight | And avoid svn completely that way |
| 15:33.34 | Greenlight | :) |
| 15:33.42 | Greenlight | Thanks again, much appriciated :) |
| 15:36.20 | *** join/#asterisk Zakabog (~banshee@its.kobaz.net) |
| 15:39.08 | Kobaz | Zakabog: it's you |
| 15:39.26 | Zakabog | It's me! |
| 15:39.49 | Zakabog | Should the use count show something other than 0 when I do module show? |
| 15:39.56 | Kobaz | depends |
| 15:40.09 | Kobaz | if there's nothing using it then it'll show 0 |
| 15:40.21 | Kobaz | so like if you have music on hold loaded, and someone is on hold, listening to music |
| 15:40.25 | Kobaz | then the count will be > 0 |
| 15:40.30 | Zakabog | Ah |
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| 15:46.27 | *** join/#asterisk Dale_Noll (~dale@kamino.mcts.org) |
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| 15:50.14 | Dale_Noll | I am having a problem with 'talk off' on a TE210P card with relaxdtmf turned off. Does anyone have any suggestions on what to look at/adjust? |
| 15:51.24 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 15:58.32 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:59.44 | WIMPy | 'talk off'? |
| 16:06.58 | Dale_Noll | 'Talk Off' is, as I understand it, when the system detects what it thinks is DTMF when it is actually hearing parts of speech. What we are experiencing is just that, the local user hears random DTMF during a conversation. The log shows dahdi detecting and regenerating digits that cannot be generated on the key pad such as 'D'. |
| 16:10.22 | Zakabog | Having a weird issue with pbx_spool.so, I have a script that creates a .call file but immediately after the .call file is created I get a call back |
| 16:10.34 | Zakabog | I've got some info here http://pastebin.com/vzhwJ64n |
| 16:17.21 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 16:20.39 | WIMPy | Dale_Noll: IIRC there is a module parameter to turn on/off hardware DTMF detection. |
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| 16:28.17 | *** join/#asterisk qakhan (~qakhan@180.178.160.11) |
| 16:28.20 | qakhan | hi all |
| 16:28.31 | qakhan | i have 8 agents in queue |
| 16:29.51 | qakhan | when calls comes in agent take them but after 1 call to each agent. calls dont goes to agents and i see in asterisk cli the following messgae |
| 16:30.58 | qakhan | GOT SIP Response 603 Decline back from 192.168.5.3 |
| 16:36.30 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 16:39.04 | Dale_Noll | I don't really want it turned off, just a little less sensitive. relaxdtmf=yes makes it more sensitive, but I have it off and I still get intermittent DTMF. There is a module param for dtmfthreshold, but I have not determined what it is a threshold of, volume, duration? |
| 16:40.10 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 16:40.10 | Ice_Strike | Hi |
| 16:40.43 | Dale_Noll | Zakabog: are you creating the call file in some other directory and then 'mv'ing it into the outgoing directory? Or are you copying it with the cp command? |
| 16:41.20 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:41.54 | Ice_Strike | After sending DTMF (eg: wwwww2wwwwww2) I want to start recording Monitor() for 6 seconds and then send another DTMF and record again for 5 seconds |
| 16:41.54 | Zakabog | It's being created in the outgoing directory |
| 16:41.57 | Ice_Strike | Is that possible? |
| 16:42.04 | Ice_Strike | It must be on same recording file |
| 16:42.32 | [TK]D-Fender | zaNEVER do that |
| 16:42.33 | Zakabog | Oh actually that might be it since the file is created then the modification time is altered, I'll see if changing that does something |
| 16:43.41 | Dale_Noll | Zakabog: try creating it in /tmp, sent the mtime and then mv it into the outgoing directory. |
| 16:43.47 | Zakabog | Yeah |
| 16:43.51 | [TK]D-Fender | Zakabog, You have to create it outside of the spool folder, and MOVE it there after with a MV from within the same FS |
| 16:48.19 | *** join/#asterisk qakhan (~qakhan@180.178.167.249) |
| 16:48.54 | qakhan | i have 8 agents in queue |
| 16:48.56 | qakhan | when calls comes in agent take them but after 1 call to each agent. calls dont goes to agents and i see in asterisk cli the following messgae |
| 16:49.03 | qakhan | GOT SIP Response 603 Decline back from 192.168.5.3 |
| 16:49.06 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:49.37 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:52.51 | Dale_Noll | WIMPy: Are you referring to the dtmfthreshold parameter or are you thinking of something else? |
| 16:55.23 | *** join/#asterisk scgm11 (~Sebastian@r186-50-100-94.dialup.adsl.anteldata.net.uy) |
| 16:55.58 | scgm11 | hi |
| 16:56.03 | scgm11 | I´m see in this chan_dahdi.c:11807 do_monitor: Whoa.... I'm owned but found (48) in read... |
| 16:56.07 | scgm11 | on 1.8.12 |
| 16:56.20 | scgm11 | any idea? this is through FXO channel |
| 16:56.43 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:58.01 | *** join/#asterisk navaismo (~navaismo@201.123.60.170) |
| 16:58.13 | Zakabog | Okay now I have the script create the file in /tmp, change the modified time, play a prompt (so I have some time to see the file was created) and then does a rename() (it's a php script) to move the file to the outgoing directory |
| 16:58.52 | Zakabog | But as soon as it does the rename, the call is made |
| 17:00.53 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:01.10 | Ice_Strike | Does Monitor() delete if audio file exist or record at the end of audio message? |
| 17:01.50 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:03.42 | Dale_Noll | Zakabog: I do not know if |
| 17:04.22 | Dale_Noll | PHP does a cp and unlink or a linux mv. Try generating a mv and calling System() |
| 17:05.05 | Ice_Strike | I am looking a way to stop Monitor() and then start Monitor() again but Append existing file? |
| 17:05.26 | Dale_Noll | Zakabog: Such as System( mv /tmp/callfile /var/spool/asterisk/outgoing/) |
| 17:06.18 | Dale_Noll | Ice_Strike: I do not believe it is possible with straight Asterisk. You may be able to create multiple files and stich them together with the sox tool. |
| 17:06.22 | Qwell | Dale_Noll: /tmp is likely a different FS, so mv/rename are equivalent anyways. |
| 17:06.36 | scgm11 | |
| 17:06.37 | scgm11 | hi |
| 17:06.38 | scgm11 | I´m see in this chan_dahdi.c:11807 do_monitor: Whoa.... I'm owned but found (48) in read... |
| 17:06.39 | scgm11 | on 1.8.12 |
| 17:06.39 | scgm11 | any idea? this is through FXO channel |
| 17:08.11 | Ice_Strike | @Qwell Do you know the answer to my question? |
| 17:08.14 | *** join/#asterisk KNERD (~KNERD@adsl-99-65-2-30.dsl.hrlntx.sbcglobal.net) |
| 17:08.37 | scgm11 | I´m see in this chan_dahdi.c:11807 do_monitor: Whoa.... I'm owned but found (48) in read... |
| 17:08.37 | scgm11 | on 1.8.12 |
| 17:08.37 | scgm11 | any idea? this is through FXO channel |
| 17:08.42 | Qwell | Ice_Strike: Doesn't Monitor() have an a option to append? |
| 17:08.47 | Qwell | 'a' option |
| 17:10.12 | qakhan | i am getting this message GOT SIP Response 603 Decline back from 192.168.5.3 |
| 17:10.16 | Dale_Noll | Qwell: I have used mv between FS and it maintains the mtime. |
| 17:10.32 | Qwell | Dale_Noll: That seems like a bug. |
| 17:11.02 | Zakabog | The system mv worked, thanks Dale_Noll |
| 17:11.23 | Dale_Noll | Zakabog: Glad to help. |
| 17:11.29 | Ice_Strike | @Qwell Ahh there is no 'a' option for Monitor() but MixMonitor() does have it. |
| 17:11.37 | Ice_Strike | Thanks! |
| 17:13.57 | Dale_Noll | Qwell: It actually makes sense. mv'ing a file is not modifying the file, it is modifying the directories involved. |
| 17:14.18 | scgm11 | did someone saw this error? |
| 17:14.19 | scgm11 | [May 28 15:46:19] ERROR[28099]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 705 (audio_audiohook_write_list): Error releasing mutex: Operation not permitted |
| 17:14.19 | scgm11 | [May 28 15:46:19] ERROR[28099]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 688 (audio_audiohook_write_list): Error obtaining mutex: State not recoverable |
| 17:14.20 | scgm11 | [May 28 15:46:19] ERROR[28099]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 705 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked! |
| 17:14.33 | Qwell | Dale_Noll: but it is, because it's creating the file. I don't think ctime should ever be > mtime |
| 17:18.25 | Dale_Noll | Qwell: I can see your point however, when moving a file between ext3 to ext3 FS, the atime and mtime are not altered, the ctime is updated. I cannot speak for any other FS type. |
| 17:19.01 | Qwell | shrugs |
| 17:19.19 | Dale_Noll | I call it a feature |
| 17:21.29 | Ice_Strike | Is PauseMonitor() work woth MixMonitor() or Monitor()? |
| 17:21.35 | Ice_Strike | with* |
| 17:23.15 | *** join/#asterisk boch (~boch@190.174.9.252) |
| 17:24.37 | anonymouz666 | scgm11: what asterisk version? |
| 17:27.06 | scgm11 | 1.8.12 |
| 17:34.05 | anonymouz666 | not good msgs :/ |
| 17:34.10 | anonymouz666 | it seems a deadlock |
| 17:34.32 | scgm11 | which one of the two that I post here? the one of the audio hook? |
| 17:34.40 | scgm11 | or chan_dahdi.c:11807 do_monitor: Whoa.... I'm owned but found (48) in read... |
| 17:34.54 | anonymouz666 | the audiohook |
| 17:35.22 | anonymouz666 | compile asterisk in debug mode, try to reproduce this and issue a bug report. |
| 17:35.27 | anonymouz666 | help asterisk get better |
| 17:35.47 | scgm11 | I have a lots of them |
| 17:35.59 | scgm11 | let me give you the link |
| 17:36.03 | scgm11 | to the problems |
| 17:36.40 | scgm11 | for example |
| 17:36.40 | scgm11 | https://issues.asterisk.org/jira/browse/ASTERISK-19851 |
| 17:37.21 | scgm11 | there you have backtraces, valgrind, other info from the system like netstat showing udp queueing without been processed (deadlock on chan sip) |
| 17:37.27 | scgm11 | you have sip debug |
| 17:37.31 | scgm11 | core show locks |
| 17:37.35 | scgm11 | etc |
| 17:37.56 | scgm11 | all problems Im facing changing from 1.6 to 1.8 |
| 17:38.11 | scgm11 | 1.6 had the sip deadlock issue but happened not too often |
| 17:38.26 | scgm11 | 1.8 is becoming unusable in medium to large installation |
| 17:38.48 | scgm11 | crashes and deadlocks more than once per day |
| 17:42.48 | *** join/#asterisk shadebob (~shadebob@41.141.93.18) |
| 17:50.51 | *** join/#asterisk dwayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 17:55.06 | *** join/#asterisk Deeewayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 17:55.06 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:00.40 | boch | can i still dial like this: SIP/user:pass@provider/number ? |
| 18:06.21 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 18:06.23 | omani | how can I do a logical expression in gotoif? |
| 18:06.26 | omani | for example: gotoif ($["${extension}" == "11|13|15|17"?1:2) |
| 18:06.27 | fubada | hi, is it possible to use PP_EACH_USER to output csv? |
| 18:06.33 | fubada | im having trouble having it print anew line |
| 18:06.46 | fubada | ${PP_EACH_USER(%{DISPLAY_NAME}\,%{CALLERID},${MAC})} |
| 18:10.30 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 18:12.15 | fubada | anyone/ |
| 18:12.16 | fubada | ? |
| 18:18.10 | *** join/#asterisk HyperNerdV2 (~HyperNerd@108-235-145-192.lightspeed.irvnca.sbcglobal.net) |
| 18:19.39 | [TK]D-Fender | omani, gotoif ($[ "${extension}" == "11" | "${extension}" == "13" | "${extension}" == "15" | "${extension}" == "17"]?1:2) |
| 18:19.46 | fubada | TK :) |
| 18:19.54 | fubada | help me insert a newline there |
| 18:19.59 | fubada | please |
| 18:20.31 | anonymouz666 | scgm11: I use 1.8 in a large installation |
| 18:20.54 | scgm11 | sip endpoints? what version exactly? heave manager use? |
| 18:20.59 | anonymouz666 | yes |
| 18:21.01 | omani | [TK]D-Fender: omfg, I have more than 10 extens to make an OR condition to |
| 18:21.08 | anonymouz666 | 1.8.11.0 exactly |
| 18:21.14 | omani | isn't there a simpler solution for this? |
| 18:21.22 | leifmadsen | fubada: you need to do multiple checks in the goto |
| 18:21.41 | fubada | leifmadsen: i think you mean omani |
| 18:21.43 | leifmadsen | or create multiple Goto()'s |
| 18:21.44 | [TK]D-Fender | And those ar sing =, not ==. |
| 18:21.48 | [TK]D-Fender | are single* |
| 18:21.52 | leifmadsen | I did |
| 18:21.54 | leifmadsen | omani: ^^^ |
| 18:22.05 | fubada | what about me :) a newline in PP_EACH_USER |
| 18:22.07 | fubada | possible? |
| 18:22.11 | leifmadsen | fubada: your % should be $ for channel variables |
| 18:22.25 | leifmadsen | no, you must have a single line for each line of dialplan |
| 18:22.32 | fubada | not according to the docs. |
| 18:22.37 | fubada | the % is correct |
| 18:22.48 | scgm11 | anonymouz666: manager use?? |
| 18:22.50 | leifmadsen | I don't know that function then |
| 18:22.56 | leifmadsen | regardless, single line per dialplan line |
| 18:23.01 | leifmadsen | can't break them up over multiple lines |
| 18:23.13 | fubada | thank you, this is for phoneprov |
| 18:23.16 | omani | [TK]D-Fender: single =? what do you mean with that? I want to make a comparison. so I have to take == |
| 18:23.19 | omani | double = |
| 18:23.27 | [TK]D-Fender | omani, no "++" |
| 18:23.30 | [TK]D-Fender | "==" |
| 18:23.31 | [TK]D-Fender | oops |
| 18:23.39 | [TK]D-Fender | * uses a single = |
| 18:23.46 | *** join/#asterisk HyperNerdV2 (~HyperNerd@108-219-209-80.lightspeed.irvnca.sbcglobal.net) |
| 18:23.59 | omani | so its not == now? |
| 18:24.12 | omani | IF ext = 10 THEN do somesh* |
| 18:24.17 | omani | and not IF ext == 10 |
| 18:24.18 | omani | ? |
| 18:24.52 | leifmadsen | omani: it never has been |
| 18:24.54 | omani | but why does it work with "==" then |
| 18:24.59 | leifmadsen | in dialplan, it's just '=' |
| 18:25.09 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:25.24 | leifmadsen | comparisons are single equal, not double in dialplan |
| 18:25.30 | omani | ok thank you |
| 18:25.36 | omani | I will change that then |
| 18:25.47 | anonymouz666 | scgm11: yes sir. |
| 18:25.47 | omani | but I wonder how this could be working |
| 18:25.50 | omani | hmm |
| 18:26.04 | leifmadsen | omani: don't bother wondering -- it's probably some syntax error that is evaluating to true or false |
| 18:26.10 | leifmadsen | it's not "working" |
| 18:26.16 | anonymouz666 | scgm11: a very large installation of callcenter using res_corosync module to do a distributed device state |
| 18:26.21 | omani | ok =) |
| 18:26.24 | scgm11 | anonymouz666: Im wonder then I have so many crashes and deadlocks…. |
| 18:26.26 | [TK]D-Fender | It is "failing conveniently" ;) |
| 18:26.29 | anonymouz666 | inbound |
| 18:26.47 | leifmadsen | for multiple checks, do: GotoIf($["foo" = "bar" | "bear" = "alive"]?true_label:false_label) |
| 18:26.56 | leifmadsen | or you can use multiple GotoIf() |
| 18:26.57 | scgm11 | using DAHDI? or just SIP? |
| 18:27.10 | leifmadsen | GotoIf($["foo" = "bar"]?true_label) |
| 18:27.17 | leifmadsen | GotoIf($[ .... ]?true_label) |
| 18:27.22 | leifmadsen | Goto(false_label) |
| 18:27.37 | anonymouz666 | scgm11: dahdi AND sip |
| 18:27.54 | scgm11 | and you have no crashes? |
| 18:28.06 | anonymouz666 | scgm11: I wouldn't say that, but it is very RARE |
| 18:28.16 | scgm11 | I will check the memory of the server then |
| 18:28.32 | scgm11 | mine os crashing every day sometimes more than ones |
| 18:28.37 | scgm11 | and deadlock ocurring too |
| 18:28.39 | omani | is there a dialplan config checker for asterisk? |
| 18:28.40 | scgm11 | on sip channel |
| 18:29.05 | anonymouz666 | scgm11: did you provide de core show locks output to the issue? |
| 18:29.13 | scgm11 | yes |
| 18:29.35 | anonymouz666 | that makes things easier for the devs |
| 18:29.38 | scgm11 | there are a lot of full BT not optimized, sip debugs, locks |
| 18:29.46 | scgm11 | and even a valgrind attach |
| 18:30.20 | anonymouz666 | You have to wait or if it is that important, pay someone else to fix for you and to the community. |
| 18:30.30 | scgm11 | the strange is I didn't have this problems of crashing with 1.6. |
| 18:30.50 | anonymouz666 | I already did that for many modules |
| 18:31.25 | scgm11 | I have no problems to pay, should I offer a bounty?? or how do you do? |
| 18:31.39 | *** join/#asterisk shadebob (~shadebob@41.250.199.37) |
| 18:31.41 | anonymouz666 | there's a list called asterisk-biz |
| 18:31.55 | scgm11 | the locking problem is very difficult to fix if you are not someone that work a lot with it |
| 18:32.47 | anonymouz666 | locking problems is something that is happening since day-1 of asterisk. lots of devs are pretty familiar with locking stuff |
| 18:33.57 | *** join/#asterisk jsmith (~jsmith@fedora/jsmith) |
| 18:33.57 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 18:34.09 | jsmith | makes the leap into #asterisk for a while |
| 18:34.15 | *** join/#asterisk kam187 (~kam187@87-194-204-58.bethere.co.uk) |
| 18:34.39 | anonymouz666 | leap i don't remember what means this word, let me look at the dictionary |
| 18:35.02 | kam187 | jsmith: so ever seen something like that? |
| 18:35.26 | jsmith | kam187: Yeah, but only where there was a signaling issue -- can you get a full SIP call trace? |
| 18:35.36 | jsmith | kam187: (using Wireshark or tcpdump) |
| 18:35.42 | jsmith | kam187: along with the Asterisk CLI output? |
| 18:35.55 | kam187 | jsmith: yeah probably.. actually its to my channel driver |
| 18:36.07 | jsmith | Oh, well then... |
| 18:36.08 | kam187 | i'm sending call proceeding, then progress, and even ringing, in that order |
| 18:36.20 | jsmith | That makes a difference |
| 18:36.30 | jsmith | Does the other side answer the channel? |
| 18:36.30 | kam187 | so it doesnt make much sense why its doing that |
| 18:36.57 | kam187 | well.. sjphone -> asterisk -> my channel driver |
| 18:37.05 | jsmith | Right... |
| 18:37.10 | kam187 | i hear ringing until the sjphone counter hits 20 seconds exactly |
| 18:37.23 | kam187 | the far end continues to ring, and my driver still shows audio (ringing) in my debug |
| 18:37.25 | jsmith | But it sounds like sjphone isn't getting back a "200 OK" from Asterisk |
| 18:37.46 | jsmith | Without looking more closely at the SIP trace between sjphone and Asterisk, I'm just guessing |
| 18:37.51 | kam187 | yeah, that makes sense (i know more about h323 than sip) |
| 18:37.56 | *** join/#asterisk angryuser_laptop (~angryuser@40-192-132-95.pool.ukrtel.net) |
| 18:38.11 | kam187 | when the far end picks up (CONNECT) audio kicks in and all is fine |
| 18:38.27 | kam187 | so it definitly looks like some dial/progress timer |
| 18:41.18 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 18:41.33 | kam187 | jsmith: do u happen to know which message (progress, proceeding, ringing) should trigger a 200 OK ? |
| 18:41.55 | jsmith | kam187: None of those do. |
| 18:42.11 | kam187 | oh |
| 18:42.15 | kam187 | what is the 200 OK? |
| 18:42.16 | jsmith | kam187: Those are all provisional messages. None of them says "the call was answered" |
| 18:42.35 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 18:42.44 | kam187 | ahh, but this is all in the alerting stage |
| 18:42.47 | kam187 | on connect, its fine |
| 18:42.54 | jsmith | In the ISDN world, progress and proceeding and alerting would be followed by a "CONNECT" |
| 18:43.08 | jsmith | In the SIP world, a 100 Trying and a 183 Progress would be followed by a 200 OK |
| 18:43.17 | kam187 | yeah, call proceeding, progress, possible more progress messages to keep the alerting timer going, then connect |
| 18:44.25 | jsmith | Yeah, that's the basic concept |
| 18:44.56 | kam187 | right, so what i'm saying... |
| 18:45.12 | kam187 | i dial on sjphone - it says "trying" |
| 18:46.01 | kam187 | i send call proceeding - it says "proceeding" |
| 18:46.01 | jsmith | Right.. .that's the Asterisk sending back a 100 Trying message, basically indicating that it received the INVITE |
| 18:46.17 | jsmith | then Asterisk goes and dials your channel driver |
| 18:46.24 | kam187 | i send progress - it says "session progress" |
| 18:46.34 | jsmith | when your channel driver sends back proceeding, Asterisk sends sjphone a 183 message, and you hear ringback |
| 18:46.46 | kam187 | i can hear the ringing from my channel driver (inband) |
| 18:46.58 | kam187 | but after 20 seconds, sjphone goes to 'trying' again, no audio |
| 18:47.09 | kam187 | if i pickup the far end, it goes to connected and all is fine |
| 18:47.10 | jsmith | blinks |
| 18:47.31 | kam187 | wierd huh |
| 18:48.12 | jsmith | Yeah. |
| 18:48.12 | kam187 | i even tried to send a second progress after 15 seconds, but it still does the same |
| 18:48.23 | jsmith | Why would you send a second one |
| 18:48.26 | jsmith | ? |
| 18:48.30 | kuku | if I have chanspy - and the whistle delay is like 10 seconds.... whats going on ? |
| 18:48.57 | kam187 | jsmith: normally i wouldn't, but i just wanted to see if it would reset its timer |
| 18:49.22 | *** part/#asterisk scgm11 (~Sebastian@r186-50-100-94.dialup.adsl.anteldata.net.uy) |
| 18:49.25 | fubada | http://pastie.org/3990437 folks i have this small conf menu, and everything works as expected but the first time I enter the room number "2500" asterisk says 'that pin is invalid' |
| 18:49.30 | fubada | some sort of a timing issue,can somene help? |
| 18:49.48 | jsmith | kam187: In short, I'm still just guessing until you can get traces |
| 18:50.15 | jsmith | kam187: but in general, Asterisk won't send the 200 OK to sjphone until your channel driver tells Asterisk that it has connected the call. |
| 18:50.27 | kam187 | ok |
| 18:50.36 | kam187 | so 200OK is the connect in the sip world |
| 18:50.43 | kam187 | thats not the issue |
| 18:51.10 | kam187 | i guess its sjphone's progress timer fixed at 20 seconds, maybe its just broken |
| 18:51.48 | jsmith | Try another SIP client, perhaps? |
| 18:52.06 | kam187 | yeah thats a sensible idea :) |
| 18:52.32 | kam187 | i just find it such a obvious "bug" that if it was really a bug lots of people must have reported it |
| 18:52.50 | kam187 | but yeah, instead of stabbing in the dark i'll try another client then take logs |
| 18:52.55 | kam187 | no short cuts :P |
| 18:53.38 | jsmith | There never is, especially if you're trying to build your own channel driver |
| 18:53.50 | jsmith | You either learn to troubleshoot, or your flounder for a long time |
| 18:54.10 | kam187 | yeah |
| 18:54.58 | *** join/#asterisk mazpe (~mazpe@74.212.157.66) |
| 18:55.18 | kam187 | jsmith: know any decent sip clients for windows? |
| 18:55.27 | kam187 | free and configurable :P |
| 18:55.56 | jsmith | kam187: CounterPath has X-Lite, which is pretty decent... and I *think* there might be a windows version of Blink. |
| 18:56.08 | kam187 | cool i'll check them out |
| 18:56.14 | jsmith | kam187: I'm not the right guy to talk to, as I haven't run Windows on my machine in about ten years now |
| 18:56.47 | kam187 | i mostly use osx, but i have a dual boot win7 :) |
| 18:57.00 | fubada | can Background take max number of digits? |
| 18:57.39 | *** join/#asterisk din3sh (din3sh@41.136.83.45) |
| 18:57.53 | din3sh | hey all |
| 18:58.15 | din3sh | is it possible to restrict call transfer in a specific context? |
| 19:01.42 | *** join/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 19:04.30 | wtfitsme | Hello, I have a silly question: I have in meetme.conf: conf => 86261,7632,56261 and in extensions.conf I have: exten => _800.,1,MeetMe(86261,MToAa,7632) What I am trying to do is log in as a regular user with pin 7632 and as an admin with pin 56261. MeetMe only excepts the admin pin. What am I doing wrong? |
| 19:08.45 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:08.50 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 19:12.10 | kuku | you have the conference once in the dialplan probably |
| 19:13.16 | kuku | b/c with teh "a" flag, its admin mode |
| 19:19.47 | wtfitsme | hmm, so how do i allow a non admin user to login with the pin 7632? |
| 19:25.21 | Qwell | wtfitsme: separate extension |
| 19:26.11 | kuku | Qwell: setting up an 80 call center... 1.8 or v10 of asterisk ? |
| 19:26.19 | kuku | 80 user |
| 19:26.30 | Qwell | Do you need features that are in Asterisk 10? |
| 19:26.36 | kuku | no |
| 19:26.59 | Qwell | ~asterisk versions |
| 19:26.59 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 19:28.37 | kuku | So looks like 1.8 is the way to go at this point. |
| 19:28.44 | wtfitsme | kuku: really I need a separate inbound for the the admin of a meet me? |
| 19:28.49 | kuku | thanks. |
| 19:29.19 | kuku | wtfitsme: don't know - try it without the a flag |
| 19:29.35 | wtfitsme | will do…thanks again. |
| 19:30.10 | p3nguin | You can use a different extension or even the same extension. How you do it is up to you. |
| 19:31.25 | *** join/#asterisk zopsi (1818afbd@gateway/web/freenode/ip.24.24.175.189) |
| 19:34.12 | zopsi | Would it be possible to put an asterisk box in a drop in mode with an existing hybrid pbx system using PTSN lines? All I want the asterisk box to do for the existing hybrid pbx (KX-TDA 200) is to provide the ability to dial extensions that are on the existing asterisk system. As of now we have both PTSN and VoIP with asterisk. I would still like the KX-TDA 200 to function as a trunk for the analog phones. Is this possible? |
| 19:42.03 | [TK]D-Fender | What analog phones? Your description of what is conencted to what and how is very vague |
| 19:42.09 | [TK]D-Fender | please rephrase all of that. |
| 19:43.03 | WIMPy | I didn't read anything about analog. |
| 19:44.25 | [TK]D-Fender | " I would still like the KX-TDA 200 to function as a trunk for the analog phones" |
| 19:49.39 | zopsi | Ok sorry about that |
| 19:51.20 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:53.03 | zopsi | We have about 12 KX-T7636 connected to a KX-TDA200 (analog system). Then we have a completely separate asterisk solution with no analog phones connected to it. We have an intercom in our house with the KX-TDA200 and the 12 analog phones attached to it. We would also like to be able to intercom the asterisk digium phones from the analog phones and vice versa. However, I would not like to route the analog calls through SIP trunk provid |
| 19:54.00 | zopsi | Meaning I don't want the analog phones utilizing the VoIP SIP trunk. It would be nice to maintain all the features of the KX-TDA 200 while still having the intercom system between the two systems. |
| 19:54.57 | [TK]D-Fender | I still don't get exactly how * is connected to your other PBX |
| 19:55.12 | [TK]D-Fender | what signalling exactly? What hardware? |
| 19:58.20 | zopsi | It isn't that is what I am asking for help with. They are completely separate currently. |
| 19:58.36 | zopsi | I am using a rack mount server for asterisk. |
| 20:04.34 | zopsi | Currently I have it set up so that you can intercom the phones/specific extensions of asterisk from the analog phones by dialing a DID and entering a pin, but that is more inconvenient and the asterisk phones cannot do the same when they want to intercom the analog phones. |
| 20:04.43 | Naikrovek | can the analog system speak SIP |
| 20:09.36 | zopsi | I don't think so and even if it did configuring the phones isn't really an option. |
| 20:11.33 | zopsi | We are in the process of replacing the KX-TDA200 system and all its phones due to some issues we have had with it, but we would like to have intercom abilities in the interim. |
| 20:13.45 | zopsi | Theres no way to do this with some fxs/fxo card? |
| 20:14.55 | [TK]D-Fender | Ok the description is still in pieces... |
| 20:20.20 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:22.39 | zopsi | Really? Ok well here it goes again. We have 12 KX-7636 phones connected to a KX-TDA200 hybdrid pbx through RJ11 I believe, which is then in turn connected to the ptsn network (AT&T Phone). Then we have a completely separate 6 digium d50 phones and 2 aastra phones connected to the asterisk installation of asterisk 1.8 with freepbx through rj45 and cat5 which then in turn uses Flowroute's SIP Trunking. |
| 20:23.26 | zopsi | We use a peplink 30 loadbalancing router with 2 cable connections and 1 adsl connection. |
| 20:25.23 | zopsi | What I want to do is be able to intercom between all the devices without modifying the KX-TDA200 or the 12 phone attached to it in any way other than possibly routing it through asterisk. For example, I would place asterisk in between the pbx and the phones or inbetween the pbx and the ptsn network. |
| 20:26.01 | zopsi | I do not have the ability to modify the KX-TDA200 dial plan or software or anything inside of it |
| 20:26.11 | WIMPy | That's definitely possible. |
| 20:26.17 | [TK]D-Fender | And how do you expect to connect them? |
| 20:26.36 | zopsi | Through one of the digium cards. |
| 20:26.41 | zopsi | or similar |
| 20:28.00 | zopsi | How would you do it WIMPy |
| 20:28.21 | *** part/#asterisk Dale_Noll (~dale@kamino.mcts.org) |
| 20:29.20 | WIMPy | By connecting your lines to Asterisk and the PBX to Asterisk. |
| 20:30.58 | *** join/#asterisk shadebob (~shadebob@41.250.199.37) |
| 20:32.24 | zopsi | WIMPy, That is what I thought. Using some sort of fxo,fxs card correct? |
| 20:33.01 | *** join/#asterisk shadebob (~shadebob@41.250.199.37) |
| 20:33.11 | WIMPy | Yes |
| 20:35.26 | zopsi | We have 4 ptsn lines. So that would be 4 fxo/ 4 fxs? a little confused on the conventions |
| 20:35.44 | WIMPy | Yes. |
| 20:36.16 | WIMPy | But maybe you shoult compare the price of that card to the price of 12 new phones. |
| 20:37.55 | zopsi | True. What if I buy one of those cheap chinese ones in the interim while I move everything over to asterisk |
| 20:39.00 | WIMPy | You might have finished the transition befor you get the interim working. |
| 20:39.08 | WIMPy | But that could happen anyway. |
| 20:39.49 | zopsi | nope. We need to rewire the whole house. Even so we still have some devices like specialty doorphones and locks in our doors triggered by the phone system that would need a card to access asterisk |
| 20:43.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:44.09 | fubada | can someone clue me in as to what is going on here |
| 20:44.10 | fubada | http://pastie.org/3991087 |
| 20:44.20 | fubada | i see this when doing 'core reload' |
| 20:44.33 | fubada | why the priority lines for every entry in users.conf? |
| 20:44.59 | WIMPy | priority -1 is hint. |
| 20:45.20 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:50.17 | *** join/#asterisk blizzow1 (~jburns@c-67-176-63-14.hsd1.co.comcast.net) |
| 20:54.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:06.03 | fubada | WIMPy: tell me!!! |
| 21:06.25 | WIMPy | What? |
| 21:06.33 | fubada | the deal with that priority 01 |
| 21:06.34 | fubada | -1 |
| 21:06.56 | WIMPy | I told you. |
| 21:07.23 | WIMPy | It's the "hint" priority. |
| 21:07.40 | fubada | i see, not sure what that means |
| 21:08.23 | WIMPy | core show hints |
| 21:08.36 | fubada | ah |
| 21:08.40 | WIMPy | They're mainly used for BLF |
| 21:08.46 | fubada | is that how users.conf 'hassip' works? |
| 21:09.11 | WIMPy | Probably. |
| 21:09.17 | fubada | thank you |
| 21:09.25 | WIMPy | users.con shouldn;t really be used. |
| 21:09.31 | fubada | thank you |
| 21:09.44 | fubada | yea i understand, but the phoneprov...requires it |
| 21:15.19 | *** join/#asterisk shadebob (~shadebob@41.250.199.37) |
| 21:20.10 | *** join/#asterisk blizzow2 (~jburns@c-67-176-63-14.hsd1.co.comcast.net) |
| 21:20.17 | blizzow2 | I got a complaint about a call on my system where the callee was talking normally and in the middle of the call their voice just completely garbled, wasn't fading in and out, just jumbled audio. I don't see anything in the logs that would indicate problems and load on the server seems okay. Anything |
| 21:20.21 | blizzow2 | I can check. |
| 21:20.23 | blizzow2 | ? |
| 21:20.55 | blizzow2 | No other complaints happened around that time either. |
| 21:24.36 | pabelanger | blizzow2: sounds like a bandwidth issue |
| 21:25.13 | WIMPy | Tell him to stopp his torrents while talking :-) |
| 21:28.15 | blizzow2 | Funny haha :). The call seems like my agent's audio is okay, and since the call recorded on the server side has a portion with garbled audio, I'm worried that his bandwidth usage didn't impact the call quality or cause the issue. |
| 21:32.58 | *** join/#asterisk acedia_ (~garland@unaffiliated/ffs) |
| 21:37.01 | resist0r | nah, fragmentation |
| 21:37.02 | resist0r | were they stream'n 5 or 7 youtube vids while play'n worl of warcraft? |
| 21:37.27 | resist0r | what was the codec ? |
| 21:37.58 | *** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au) |
| 21:38.02 | resist0r | in use between the 'garbled' person and the asterisk machine? |
| 21:44.02 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 21:46.01 | shadebob | blizzow2 : he use a softphone ? |
| 21:46.17 | blizzow2 | yes. |
| 21:46.47 | shadebob | have you tested with a SIP phone on the same plug ? |
| 21:47.12 | shadebob | maybe the FLAME virus :p |
| 21:47.21 | blizzow2 | No, I haven't tested with a SIP hard phone on the same plug. |
| 21:48.03 | shadebob | if I were you I will test with another PC |
| 21:48.54 | resist0r | heh, flame virus |
| 21:52.47 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 21:53.40 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:58.10 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.4.1 (2012/05/29), 1.8.12.1 (2012/05/29), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:58.46 | *** join/#asterisk s[x] (~sx]@ppp59-167-154-113.static.internode.on.net) |
| 22:08.56 | *** join/#asterisk jplank (~G_Bove@cpe-076-182-116-060.nc.res.rr.com) |
| 22:09.18 | jplank | does dahdi 2.3 work with 1.4.26.1? |
| 22:10.27 | sruffell | generally, yes. But why stay with DAHDI 2.3? |
| 22:13.03 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-0-219.bois.qwest.net) |
| 22:15.24 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 22:18.11 | Qwell | Why 1.4 is the bigger question? |
| 22:18.38 | *** join/#asterisk shadebob (~shadebob@41.250.199.37) |
| 22:20.04 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 22:26.36 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 22:26.43 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 22:28.15 | nny | I have nat=yes under [general] and localnet=10.1.1.0/255.255.255.0 as well. I have externip=X.X.X.X, 2 network interfaces (10.1.1.250 and the xtern) and my phone is 10.1.1.75, shouldn't asterisk properly communicate with the local phone as well as a phone connecting to the externip? |
| 22:29.55 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 22:31.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-oyawefovavcsvrau) |
| 22:38.49 | nny | anyone around? |
| 22:39.23 | jsmith | nny: There are lots of people around. |
| 22:39.36 | jsmith | nny: Ask a question! Be bold! If someone knows the answer, they'll help. |
| 22:39.42 | nny | hmm i did, i'll try again |
| 22:39.44 | jsmith | nny: If they're busy or don't know, they'll stay quiet :-) |
| 22:39.48 | WIMPy | Some are asquare, tho. |
| 22:39.49 | nny | haha figured |
| 22:39.50 | nny | I have nat=yes under [general] and localnet=10.1.1.0/255.255.255.0 as well. I have externip=X.X.X.X, 2 network interfaces (10.1.1.250 and the xtern) and my phone is 10.1.1.75, shouldn't asterisk properly communicate with the local phone as well as a phone connecting to the externip? |
| 22:39.56 | nny | i'm square fo sho |
| 22:40.28 | jsmith | Asterisk typically doesn't do so well at handling two separate network interfaces |
| 22:40.35 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:40.40 | jsmith | at least in my experience |
| 22:40.59 | nny | jsmith: that's rather odd considering there's no proper way to handle internal and external connections without both :S |
| 22:41.24 | jsmith | NAT firewall, SIP proxy... there are two ways to handle it |
| 22:41.54 | nny | don't follow. External connections include VoIP providers |
| 22:42.44 | nny | you're saying you should register with a voip provider or use a SIP proxy? that's a can of worms either way |
| 22:42.49 | jsmith | Sure... and let's say you've got a NAT firewall (say, a regular DSL modem or cable modem typical for a residential setup) |
| 22:42.54 | nny | nope |
| 22:42.56 | nny | business |
| 22:43.08 | jsmith | OK, then you've got a firewall of some sort, right? |
| 22:43.26 | jsmith | Or are you really wanting to put your Asterisk box directly on a public IP? |
| 22:43.30 | nny | sonicwall on their local net, asterisk isn't using it for very specific reason, one of them being sonicwall is crap |
| 22:43.32 | nny | always |
| 22:43.53 | jsmith | Yeah, sonicwall and Asterisk = bad |
| 22:43.56 | nny | Asterisk is on a public IP as it should be |
| 22:44.01 | [TK]D-Fender | nny: make sure you DISABLE the SIP ALG on it |
| 22:44.15 | nny | [TK]D-Fender: SIP ALG? I'll look it up, thanks |
| 22:44.27 | [TK]D-Fender | nny: And no reason for your * to have any issues for have a private & public IP on it. |
| 22:44.42 | [TK]D-Fender | only real problem is when you have multiple public |
| 22:44.45 | nny | [TK]D-Fender: don't use the sonic wall (although that's handy advice) |
| 22:44.50 | nny | for the future that is |
| 22:45.00 | nny | for now, I have nat=yes under [general] and localnet=10.1.1.0/255.255.255.0 as well. I have externip=X.X.X.X, 2 network interfaces (10.1.1.250 and the xtern) and my phone is 10.1.1.75, shouldn't asterisk properly communicate with the local phone as well as a phone connecting to the externip? is my query |
| 22:45.05 | [TK]D-Fender | PB it all up |
| 22:45.12 | nny | [TK]D-Fender: will do thanks |
| 22:45.21 | jsmith | brb... have to solve a family problem :-( |
| 22:47.40 | nny | [TK]D-Fender: http://pastebin.com/6K4ccQc9 is the sip.conf relevant config |
| 22:48.15 | nny | [TK]D-Fender: my testing friday seems to have not been 100% conclusive, i took nat=XXX off the peer and moved it to the general area, at which point the phone registered |
| 22:48.23 | [TK]D-Fender | nny: what * ver? |
| 22:48.30 | nny | [TK]D-Fender: today it's not registering again, from what I can tell in dumps the phone is replying to the wrong IP |
| 22:48.44 | nny | or rather asterisk is sending info back to the wrong peer, i'll debug it |
| 22:48.57 | nny | [TK]D-Fender: 1.8.11.1 |
| 22:49.06 | nny | er i can't debug it :\ |
| 22:49.11 | [TK]D-Fender | nny: canreinvite is replaced by directmedia |
| 22:49.27 | [TK]D-Fender | set to NO in general and in all your peer, and mark your phone entry as nat=yes as well |
| 22:49.39 | nny | [TK]D-Fender: yeah username too, i'll change those. |
| 22:50.02 | [TK]D-Fender | IIRC externip was replaced by externaddr |
| 22:50.04 | [TK]D-Fender | ^^ |
| 22:50.21 | nny | [TK]D-Fender: i'll try those first |
| 22:50.41 | nny | [TK]D-Fender: do you absolutely have to define NAT=yes/no per peer for switching between local and external networks? |
| 22:51.26 | [TK]D-Fender | it shouldn't hurt either |
| 22:51.44 | [TK]D-Fender | only thing it tends to hurt are ITSP entries |
| 22:51.50 | [TK]D-Fender | that keep separate media servers |
| 22:52.17 | nny | [TK]D-Fender: i understand, the hope here was to allow them to travel in and out of the network with (say soft clients or even desk phones) without changing nat=XXX each time |
| 22:52.27 | *** join/#asterisk agisamentum (~vastina@70.99.103.122) |
| 22:53.12 | nny | [TK]D-Fender: let me tcpdump the interface while the phone tries to register, I need to see what's failing specifically |
| 22:53.56 | [TK]D-Fender | * SIP debug... forget external |
| 22:54.14 | nny | [TK]D-Fender: asterisk won't do it without registration |
| 22:54.24 | nny | [TK]D-Fender: it's not registering with that nat setting right now either |
| 22:55.24 | [TK]D-Fender | show me the new configs and the SIP debug... |
| 22:55.49 | nny | [TK]D-Fender: full SIP debug on? I can have logger dump it to it's own file at least |
| 22:56.03 | nny | it's/its* |
| 22:56.35 | [TK]D-Fender | no, * sip debug |
| 22:56.46 | nny | [TK]D-Fender: how would i enable sip debug if the peer is not registering? |
| 22:57.08 | nny | [TK]D-Fender: i may be chasing the wrong end here, but nat=yes/no seems to be the only thing that fixes the register |
| 22:57.08 | [TK]D-Fender | "sip set debug on" |
| 22:57.12 | [TK]D-Fender | Doesn't care about anything |
| 22:57.16 | nny | [TK]D-Fender: yeah ok, that's what I thought, gotcha |
| 22:58.43 | nny | [TK]D-Fender: what level of logging does sip debug fall under, verbose, debug or ? |
| 22:59.03 | [TK]D-Fender | "none of the above" |
| 22:59.19 | SeRi | who makes the digium phones? |
| 22:59.22 | nny | so you can't isolate sip dialog in logs via logger.conf? |
| 22:59.26 | [TK]D-Fender | just go to CLI |
| 22:59.28 | nny | SeRi: Polycom is my guess |
| 22:59.37 | nny | [TK]D-Fender: yeah that's a quick flood :( |
| 23:00.00 | SeRi | nny: Thanks... for some reason they look like grand stream.... I am glad they are not. |
| 23:00.01 | nny | [TK]D-Fender: let me try and manually catch one |
| 23:00.08 | SeRi | They are pretty inexpensive.... |
| 23:00.10 | nny | SeRi: my guess is based on the HD Voice emblam |
| 23:00.13 | nny | emblem* |
| 23:00.25 | SeRi | ah. good call. |
| 23:00.36 | [TK]D-Fender | means little |
| 23:00.53 | nny | [TK]D-Fender: wasn't sure if that was Polycom's thing. The logo is identical |
| 23:00.55 | [TK]D-Fender | Now if the phone licensed just the SIREN CODEC, I'd bet that'd earn the logo |
| 23:01.16 | [TK]D-Fender | but implies nothing about source origin or hardware manufacturer |
| 23:01.23 | SeRi | I see. |
| 23:01.40 | [TK]D-Fender | and the D50 feels quite different from any polycom. |
| 23:01.44 | nny | [TK]D-Fender: odd, i am seeing a 401 for that phone, but the registration credentials are proper |
| 23:01.57 | nny | [TK]D-Fender: http://pastebin.com/F1Qb3nRB |
| 23:02.04 | SeRi | [TK]D-Fender: You worked with them? |
| 23:02.43 | [TK]D-Fender | I have a D50 on my desk right now |
| 23:02.59 | SeRi | nice! |
| 23:03.05 | SeRi | what are your thoughts? |
| 23:03.49 | [TK]D-Fender | Like everything else there are pluses and minuses. I haven't done enough advanced work with them to build a final opinion yet. |
| 23:04.32 | nny | [TK]D-Fender: asterisk isn't reporting that credentials are incorrect (or other) as per usual in console. What's the meaning of that 401`? |
| 23:04.37 | SeRi | [TK]D-Fender: I see. cool. |
| 23:05.13 | nny | [TK]D-Fender: here's two parts of the dialogue http://pastebin.com/5f1iBa4b |
| 23:05.33 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:05.33 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:05.45 | nny | hides sheriff's here! |
| 23:06.39 | nny | [TK]D-Fender: per your request here is the new sip.conf relevant section http://pastebin.com/a8VhT8n3 |
| 23:11.43 | [TK]D-Fender | srvlookup=yes ; Enable DNS SRV lookups on outbound calls <- I;d disable |
| 23:12.39 | nny | [TK]D-Fender: done |
| 23:13.01 | [TK]D-Fender | alwaysauthreject=yes <- no |
| 23:13.33 | nny | [TK]D-Fender: oh yeah, that explains the 401. We were doing that as a security precaution to avoid revealing known extensions. gotcha |
| 23:14.13 | nny | [TK]D-Fender: er nm still 401 on 6759 |
| 23:15.08 | [TK]D-Fender | 1 last thing, type=peer for your device |
| 23:16.27 | nny | [TK]D-Fender: ok added it, still seeing 401s though. So odd, here's the problem. i could just set nat=no for 6759 and it would register. I then could remove it and it would still register, as it did friday, then two laters later it'd be busted again |
| 23:16.41 | nny | two days* |
| 23:17.14 | [TK]D-Fender | :/ |
| 23:18.09 | nny | [TK]D-Fender: iat tihs point just gonna define nat in the peers. I hate to do it considering this kind of setup would be invaluable in this scenario, but it |
| 23:18.21 | nny | it's at the point where i think there's something at work here I can't figure out |
| 23:22.44 | nny | [TK]D-Fender: From: <sip:6759@phone.smithcarson.com> To: <sip:6759@phone.smithcarson.com> |
| 23:23.49 | nny | [TK]D-Fender: "phone.smithcarson.com" is set to resolve to the pbx IP from the local dns server and the PBX public IP on external dns. Why would the From and To be the same when the peer is 6759@10.1.1.75 ? |
| 23:24.05 | nny | or are those fields not what they seem? |
| 23:26.51 | [TK]D-Fender | perhaps you should not be referring to a nmed host and instad tell the phone the IP.... |
| 23:27.18 | nny | [TK]D-Fender: that's part of the need to move in and out of the network without changing sip.conf or the phone's registration address |
| 23:28.12 | nny | [TK]D-Fender: i am just going to define each peer as nat=yes or no. This seems really wonky considering how many setups would have softphones and laptops moving in and out of the network |
| 23:30.31 | nny | [TK]D-Fender: annnd setting nat=no on the peer fixed it. I give up |
| 23:31.36 | *** join/#asterisk thecardsmith_ (~doug@pdpc/supporter/student/thecardsmith) |
| 23:33.08 | [TK]D-Fender | I don't get it... |
| 23:36.29 | nny | [TK]D-Fender: nor do I, you want me to capture the registration with nat=no? at this point I just want to know for the sake of knowing lol |
| 23:41.41 | nny | http://pastebin.com/gCU2EQ4r vs http://pastebin.com/5f1iBa4b (failure) |
| 23:42.06 | nny | [TK]D-Fender: i can tell you the nat=no has to be set it seems, the failed attempt has <--- Transmitting (NAT) to 10.1.1.75:50353 ---> |
| 23:42.21 | nny | [TK]D-Fender: and the working one has <--- Transmitting (no NAT) to 10.1.1.75:5060 ---> .. |
| 23:43.51 | [TK]D-Fender | can you show BOTH sides complete... |
| 23:45.12 | nny | [TK]D-Fender: i think so, give me asec |