| 00:33.03 | floren | :D |
| 00:33.50 | WIMPy | Good morning |
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| 00:52.17 | Katty | evening |
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| 01:07.14 | floren | hi :) |
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| 02:32.32 | floren | any wise asterisk person who is willing to troubleshoot an unauthorized sip trace I get when I place a call? |
| 02:33.24 | floren | is probably one of my settings I have into sip.conf |
| 02:38.02 | ChannelZ | pastebin |
| 02:38.28 | floren | whoa ChannelZ :) |
| 02:38.39 | floren | i got the inbould calls working :) |
| 02:38.56 | ChannelZ | your phone finally wake up and start playing nice? |
| 02:38.58 | floren | the outboud is spitting a 401 |
| 02:39.11 | floren | heh ya :) thanks to you and [TK]D-Fender |
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| 02:39.41 | ChannelZ | well not me, I don't know jack about that phone or why it's so braindead |
| 02:39.57 | floren | let me paste the 2 config files and the sip trace :) |
| 02:45.17 | igcewieling1 | the first sip packet will always generate a 401 response which includes a "nonce", which the client uses to encrypt the password and make the original request again, but with auth. |
| 02:47.51 | floren | the configs: http://codepad.org/9molT2S3 |
| 02:48.02 | floren | let me start a fresh trace |
| 02:49.42 | floren | the trace: http://codepad.org/sC4WLsng |
| 02:50.46 | ChannelZ | No matching peer for '101' from '192.168.1.9:49231' |
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| 02:51.04 | ChannelZ | you've renamed your peer 'cisco' but not told your phone that |
| 02:51.19 | ChannelZ | and/or the IP doesn't match |
| 02:51.29 | floren | i see |
| 02:51.39 | floren | the phone ip is 1.9 |
| 02:52.16 | floren | *CLI> sip show peers |
| 02:52.16 | floren | Name/username Host Dyn Forcerport ACL Port Status Description |
| 02:52.19 | floren | 101/101 192.168.1.9 D 5060 Unmonitored |
| 02:52.20 | ChannelZ | your incoming dialplan won't work either, as you're dialing SIP/101 which doesn't exist either |
| 02:52.22 | floren | 60201 67.205.74.184 a 5060 Unmonitored |
| 02:52.25 | floren | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 02:52.26 | floren | looks good on this side |
| 02:52.38 | floren | let me chnage the name |
| 02:52.46 | ChannelZ | then your system isn't running on the configs you pasted. You have no peer 101 in the sip.conf you provided |
| 02:53.04 | ChannelZ | have you reloaded since modifying? |
| 02:53.09 | floren | ya |
| 02:53.14 | floren | one sec let me do it again |
| 02:53.24 | ChannelZ | then you either pasted the wrong configs or you've somehow created magic |
| 02:53.32 | floren | :D |
| 02:54.53 | floren | is [101] not [cisco] |
| 02:54.55 | floren | my bad |
| 02:55.01 | floren | jest checked the config |
| 02:55.13 | floren | actually i used a template :) |
| 02:55.15 | ChannelZ | and 60201 not voipms apparently |
| 02:55.32 | floren | but i was affraid you youd kick my ass :) |
| 02:55.41 | [TK]D-Fender | [22:55]florenactually i used a template <- I've told you for DAYS to get rid of that because it would only lead to confusion |
| 02:55.43 | floren | i'll post the real file contents :) |
| 02:55.56 | floren | i know [TK]D-Fender :) |
| 02:56.08 | [TK]D-Fender | [22:55]floreni'll post the real file contents <- do you give your doctor false health reports too? |
| 02:56.10 | floren | i'll chnage them back now |
| 02:56.47 | floren | for me the templates help me keep things organized |
| 02:56.55 | floren | plus soon i will add 2 extra did lines |
| 02:57.12 | floren | so it will avoid duplicating code |
| 02:57.23 | floren | anyways, let me past what i have now |
| 02:58.55 | [TK]D-Fender | You can't get your names straight and that's when you had just TWO peers. |
| 02:59.13 | ChannelZ | and no beers |
| 03:01.17 | floren | [TK]D-Fender: of course i can, is very logical |
| 03:01.40 | [TK]D-Fender | [22:54]florenis [101] not [cisco] [22:54]florenmy bad <- You've ALREADY failed |
| 03:01.47 | floren | http://codepad.org/LextsYlV |
| 03:01.57 | floren | that is because i wanted to bring back the structure you like |
| 03:02.04 | floren | i like the templates and they are very useful imo |
| 03:02.11 | floren | and what i have there is clear to me |
| 03:02.42 | ChannelZ | how can you expect us to debug your problems when you're forging the configs you show? |
| 03:02.57 | floren | you did not wasted yesterday 2 hours for nothing :) |
| 03:03.27 | floren | well, i know [TK]D-Fender did not liked my context names and the templates |
| 03:03.29 | [TK]D-Fender | And I see the dialplan is back to the 101 nonsense |
| 03:03.37 | [TK]D-Fender | Yes, it's all back |
| 03:03.59 | ChannelZ | show what you've got, not what you think we want to see. |
| 03:04.07 | floren | ok :) |
| 03:04.11 | [TK]D-Fender | Yeah, I'm done wasting my time on this. |
| 03:05.09 | floren | sorry you feel this way [TK]D-Fender |
| 03:06.49 | [TK]D-Fender | I know. It's not "sorry I keep screwing with the evidence", it's "sorry you feel that my doing so isn't worth the trouble anymore". |
| 03:07.51 | floren | to me this format look organized and clear. the same way to you it looks backwards and confusing. i really don't think the usage of templates name convention should be a road blocker |
| 03:08.13 | [TK]D-Fender | It simply isn't worth the trouble of trying to teach you to do it the right way and watch all of it goo out the window and you're showing us doctored results aand configs. You want to paste of false junk because it's what you think we want to so? I've got better things to do that to debug a lie |
| 03:08.15 | floren | but that is just me |
| 03:08.44 | [TK]D-Fender | whichEspecially one that shows that my previous time was wasted |
| 03:08.54 | [TK]D-Fender | moves on |
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| 03:09.24 | floren | i dont think it was wasted, i understtod very well what you highlighted and apply it in a different way. i dont see nothing wrong in that |
| 03:09.56 | floren | have a pleasant night [TK]D-Fender |
| 03:10.13 | [TK]D-Fender | [23:01]florenthat is because i wanted to bring back the structure you like <- you're showing us FALSE information |
| 03:11.28 | floren | look at the extra energy we wasted just for a context. i dont think is worth IMO, the improtant thing is that I understood what you explained to as you can see from the new config files |
| 03:12.01 | floren | i was simply trying to avoid this discussion, but it looks like we had to have it |
| 03:12.58 | floren | anyways, i apreciate your time spend and i'm sorry that you are dissapointed |
| 03:14.50 | floren | ChannelZ: i was trying to access your site but the sip page is asking me to login |
| 03:15.01 | floren | just letting you know |
| 03:15.40 | ChannelZ | yeah its not public yet |
| 03:15.51 | floren | i see, yesterday i was able to see it |
| 03:16.57 | floren | installing netbackup on few redhat machines, while i talk to you guys |
| 03:21.07 | [TK]D-Fender | leifmadsen: You about? |
| 03:24.46 | carrar | ChannelZ has a porn site? |
| 03:25.04 | carrar | err pr0n |
| 03:28.31 | ChannelZ | no, just visit others'. |
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| 06:04.59 | bulkorok | hi |
| 06:06.36 | ChannelZ | oh hai |
| 06:09.13 | floren | ChannelZ: still awake? :) |
| 06:09.15 | bulkorok | ChannelZ is on the road :-) |
| 06:09.22 | bulkorok | what the?! |
| 06:09.27 | bulkorok | :-D |
| 06:09.34 | floren | almost done with the servers, i have a headache |
| 06:09.53 | bulkorok | 08 am in the morning here :) |
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| 06:51.25 | random_cat | asterisk 11.4.0 with freepbx 2.11.0.4 on debian 7 is telling me "Failed to execute '/var/lib/asterisk/agi-bin/dialparties.ago': File does not exist." but /etc/asterisk/asterisk.conf has "astagidir => /usr/share/asterisk/agi-bin" in "[directories]" |
| 06:52.08 | random_cat | Why does asterisk do this, and how do I better understand asterisk's path searching? |
| 06:52.51 | random_cat | (asterisk & freepbx built from repository) |
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| 08:55.28 | Rumbles | hi guys, if I initiate a call from the CLI, is there a way to set my callerID as well? |
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| 10:18.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 13:15.16 | catphish | just embarking on a 1.6 -> 11 upgrade |
| 13:16.30 | Greenlight | Good luck, and remember to read the upgrade notes files :) |
| 13:16.57 | catphish | it's extra fun because i was using a modified 1.6 and realtime |
| 13:17.07 | catphish | i'll be sure to check the notes, thanks |
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| 13:20.01 | leifmadsen | ugh... 1.6.x to 11? wow |
| 13:20.03 | leifmadsen | that's going to be a big upgrade |
| 13:20.15 | leifmadsen | hope you're migrating between systems and not upgrading the production system |
| 13:20.21 | leifmadsen | because that's going to end in all sorts of fail |
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| 13:31.23 | catphish | leifmadsen: i'll see what happens :) |
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| 13:44.25 | zrzerenato | what is the possible couses for CONNECTION_LOST ..? |
| 13:46.17 | Greenlight | Cleaner unplugged server |
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| 13:47.55 | zrzerenato | can this CONNECTION_LOST be a data base time_out??? |
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| 14:06.51 | catphish | well upgrading 1.6->11 worked as far as starting up and allowing calls to be made with no changes which is nice, now to see what IS broken :) |
| 14:10.29 | Katty | GOOD MORNING CUPCAKES. |
| 14:11.07 | catphish | sadly it didn't resolve the behaviour i had hoped it would, whereby transferring a call ends the CDR session for the original call |
| 14:22.51 | Greenlight | catphish: If you're looking for CDR's to show the whole duration of calls to the PSTN, I find the safest solution is to stick another box in between your actual asterisk box and your PSTN, and record the CDR's on that |
| 14:22.58 | Greenlight | Transferrring tends to do funny things |
| 14:23.49 | catphish | transferring ends the original call as far as the CDR is concerned :( |
| 14:24.05 | Greenlight | Yea |
| 14:24.24 | catphish | an extra lossing host is certainly a reliable option, but quite an overhead |
| 14:24.25 | Greenlight | Which may, or may not, be the behaviour you want/expect |
| 14:25.01 | catphish | there's an epic thread on the subject here: https://issues.asterisk.org/jira/browse/ASTERISK-11309 |
| 14:25.02 | LieutPants | [ASTERISK-11309] [Status: Closed] Missing CDR's for Transfers - https://issues.asterisk.org/jira/browse/ASTERISK-11309 |
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| 14:25.11 | Greenlight | Depends on your call volume, and infrastructure. If you just want to capture CDR's then a VM would suffice and easily handle 100+ simultenous calls with low resources |
| 14:25.15 | catphish | but i'm still unsure if there's a real fix |
| 14:25.52 | catphish | injecting an extra asterisk into an existing setup is quite a complication sadly |
| 14:26.15 | Katty | idly wonders who LieutPants is |
| 14:26.15 | catphish | but at the same time i understand that the fix probably isn't trivial |
| 14:26.39 | Katty | LieutPants: oh. you. |
| 14:27.18 | Greenlight | I'm not even sure that i'd consider it "broken" |
| 14:27.21 | *** join/#asterisk transfinite (~aleph_0@unaffiliated/transfinite) |
| 14:27.27 | Greenlight | Technically it's a separate call |
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| 14:28.02 | Greenlight | You could potentially do something clever when you query the database to show the CDR records as a single entry |
| 14:28.16 | Greenlight | Or use CEL |
| 14:28.17 | catphish | Greenlight: it's a good question |
| 14:28.30 | catphish | most people would consider it to be 2 siultaneous calls |
| 14:28.51 | catphish | since there are 2 outbound sip calls active |
| 14:28.54 | Greenlight | Have you looked at CEL, and checked if that gives you the info you need? |
| 14:29.02 | catphish | but asterisk of course only considers that to be one bridged call |
| 14:29.31 | catphish | i haven't, that thread suggested it may be of assistance |
| 14:29.58 | Greenlight | It's generally more detailed, and easier to "join" different legs together |
| 14:30.12 | Greenlight | From what I recall |
| 14:31.44 | catphish | parsing CEL looks like a special kind of hell |
| 14:31.48 | catphish | but maybe necessary |
| 14:32.13 | Greenlight | You can dump it straight to a db |
| 14:32.23 | catphish | sure, but it's extremely verbose |
| 14:32.43 | Greenlight | Yup |
| 14:33.04 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:35.52 | Katty | coppice: ohai |
| 14:35.56 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 14:35.57 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:36.01 | Katty | YOU |
| 14:36.14 | Katty | GET OUT |
| 14:36.20 | Katty | from behind your desk so i can give you a hug. |
| 14:36.25 | Katty | hugs Qwell |
| 14:36.34 | Qwell | I'm in front of my desk though. |
| 14:37.37 | Katty | facepalms |
| 14:40.41 | file | Qwell *is* a desk |
| 14:41.00 | [TK]D-Fender | Deskception (tm) |
| 14:41.03 | Qwell | Little known fact. file asked me which desk he should buy. |
| 14:41.12 | Qwell | file: That just happened. |
| 14:41.14 | file | Qwell, chair. |
| 14:41.17 | Qwell | desk |
| 14:41.28 | Katty | why do i feel like i'm witnessing a couple arguing |
| 14:41.29 | file | taco |
| 14:41.34 | Qwell | file: mmm, tacos |
| 14:41.40 | coppice | you buy the desk which best hides your oversized gut |
| 14:41.48 | Qwell | coppice: exactly |
| 14:41.52 | Katty | no. not at all. |
| 14:41.57 | Katty | Qwell is a fine looking sort |
| 14:42.10 | Katty | he doesn't need a desk to hide behind. |
| 14:42.15 | Qwell | does now |
| 14:42.41 | coppice | a few years on IRC will do that to you |
| 14:42.52 | Qwell | coppice: So will food. |
| 14:42.58 | Katty | omnomnom |
| 14:43.05 | Katty | Qwell: you have any tasty recipes to share? |
| 14:43.11 | Katty | Qwell: i'm always looking for new ones |
| 14:43.11 | *** part/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 14:43.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 14:43.20 | [TK]D-Fender | gah |
| 14:43.21 | coppice | yeah, its the food IRC combo that realyl gets you |
| 14:43.35 | Qwell | Katty: I don't measure things. |
| 14:43.36 | Katty | coppice: that's what the cardio + sarah conner chronciles combo is for |
| 14:43.39 | Qwell | so, that's hard |
| 14:43.48 | Katty | Qwell: ah, right. |
| 14:43.54 | Katty | Qwell: i'd have to watch you then. |
| 14:44.48 | Katty | Qwell: video blog! |
| 14:47.07 | *** join/#asterisk k1ng440 (~k1ng@unaffiliated/k1ng) |
| 14:48.03 | *** join/#asterisk chazzam (~cmoye@nat/digium/x-jlkvuuvvysoyfbvd) |
| 14:48.23 | *** join/#asterisk pcAngel (yoink@S0106602ad072bdf0.vc.shawcable.net) |
| 14:48.26 | zrzerenato | does any one knows why my agents Log-Out with status "CONNECTION_LOST"???? |
| 14:50.25 | pcAngel | Is there any way to view the media path on a live call, other than watching sip debug? I'm trying to verify that my sip termination trunk doesn't stay in the media path |
| 14:50.57 | pcAngel | they're all busy servers, making sip debug not optimial |
| 14:53.23 | ChannelZ | I think 'sip show channel xxxx' might |
| 14:54.59 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 14:58.21 | *** join/#asterisk serafie (~erin@nat/digium/x-elcecokhndgrfvqt) |
| 14:58.48 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:06.18 | igcewieling1 | pcAngel: tcpdump / wireshark and you want to look at RTP too |
| 15:07.05 | igcewieling1 | I think when it shows "Local bridge" audio is not reinviting, if the CLI shows "remote bridge" then the audio re-invited. |
| 15:10.14 | *** join/#asterisk mic_ (~mic@0305ds4-vby.2.fullrate.dk) |
| 15:10.35 | mic_ | does anyone have a d210e openvox card? |
| 15:11.29 | [TK]D-Fender | ~poll |
| 15:11.29 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 15:11.31 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 15:11.32 | [TK]D-Fender | ~polls |
| 15:11.33 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 15:12.46 | mic_ | [TK]D-Fender: idea was to verify some behavior |
| 15:14.15 | *** join/#asterisk russum (~russum@m83-184-32-58.cust.tele2.hr) |
| 15:15.05 | mic_ | I had the system freeze once the isdn cable has been connected |
| 15:15.19 | mic_ | upgraded to latest DAHDI, and suddenly there are no more "NMI received" messages |
| 15:15.38 | mic_ | (but right now I have no access to an ISDN to verify it's going to freeze again or not) |
| 15:17.27 | [TK]D-Fender | I'd tell you to ask Openvox support ... but that'd be cruel... |
| 15:17.27 | Greenlight | Do you have the latest firmware on the card? Are you sure it's compatible with the latest DAHDI version ? |
| 15:21.53 | zFleshMissile | Does anyone know what causes this error "Channel 0/1, span 1 got hangup request, cause 41" ? Getting a busy tone when trying to dial out. Supposedly it's a temporary error but it happens every time. |
| 15:22.19 | *** join/#asterisk hrolf (b6b47dc4@gateway/web/freenode/ip.182.180.125.196) |
| 15:22.23 | Greenlight | Maybe you have a problem on the line? |
| 15:22.45 | hrolf | Hi #asterisk, I'm doing SIP REFER (Transfer) from my Asterisk machine to Avaya IP Office, but it is not working. |
| 15:23.21 | zFleshMissile | Could be a potential issue with the ISDN? |
| 15:23.23 | hrolf | Any ideas? |
| 15:23.37 | zFleshMissile | sorry, that was directed at Greenlight |
| 15:23.46 | Greenlight | zFleshMissile: Perhaps |
| 15:24.33 | hrolf | http://pastie.org/8124812 |
| 15:24.41 | hrolf | ^ above is the SIP debug log at my end |
| 15:24.42 | Greenlight | hrolf: A little more information... *when* does it fail? Can you call from Asterisk to the Avaya normally (eg not when transferring?) |
| 15:24.46 | hrolf | i.e. asterisk |
| 15:24.49 | zFleshMissile | Greenlight: Okay, i'll speak to the provider, do you think there's anything else i should check into? |
| 15:25.05 | Greenlight | zFleshMissile: Check the lights on the NTE |
| 15:25.24 | hrolf | Greenlight: Okay. It is like, we are receiving calls from the Avaya IP Office through a SIP trunk (by setting host=, username= etc.)\ |
| 15:25.40 | zFleshMissile | Greenlight: When I remove the asterisk server and connect the ISDN directly to phone system the calls work fine |
| 15:25.52 | hrolf | Greenlight: Then afterwards, we RUN FastAGI, and use Transfer inside it, to transfer back to an extension on the Avaya IP Office. |
| 15:26.16 | Greenlight | zFleshMissile: Then the line is likely okay. It's either a misconfiguration or faulty hardware. |
| 15:26.22 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:26.25 | hrolf | Greenlight: Looking at the SIP debug log, I can see that after I send the REFER message, I get this response from the PBX: "SIP/2.0 481 Dialog/Transaction Does Not Exist" |
| 15:26.35 | hrolf | Greenlight: The complete log is http://pastie.org/8124812 |
| 15:26.55 | pcAngel | Igcewieling1: thanks re: remote/local bridge |
| 15:27.51 | Greenlight | Can you call from Asterisk to an extension on the avaya normally ? |
| 15:28.56 | hrolf | Greenlight: Yes. |
| 15:29.32 | hrolf | Greenlight: Previously, we were doing Dial() from inside the AGI for transferring, but now we are using Transfer(). Dial() worked perfectly fine. |
| 15:29.35 | zFleshMissile | Greenlight: Okay thanks, i'll check the hardware |
| 15:30.04 | hrolf | Greenlight: http://pastie.org/8124829 are the logs from the Avaya IP Office. |
| 15:30.29 | Greenlight | If Dial worked, why change it ? |
| 15:30.44 | mic_ | Greenlight: firmware v1.3 |
| 15:30.54 | hrolf | Greenlight: Actually, if I remain with Dial(), then my channel will remain connected. |
| 15:31.17 | Greenlight | hrolf: And that's bad because ... |
| 15:31.33 | hrolf | Greenlight: Because you will have two channels connected. |
| 15:31.46 | Greenlight | hrolf: Sounds like a call to me |
| 15:31.53 | hrolf | Greenlight: ? |
| 15:32.59 | Greenlight | Asterisk is a B2BUA; I'd expect a call to have two connected channels. |
| 15:34.03 | hrolf | Greenlight: We'll I have 6 licenses on the SIP trunk I have. |
| 15:34.12 | hrolf | s/We'll/well |
| 15:34.18 | igcewieling1 | Sounds like the Avaya is not telling Asterisk when the call ends. |
| 15:34.20 | *** join/#asterisk davlefouAMD (~david@41.225.182.70) |
| 15:34.32 | hrolf | Greenlight: And I'm supposed to receive 6 simultaneous calls. |
| 15:34.56 | hrolf | Greenlight: So |
| 15:35.13 | hrolf | Greenlight: I receive the call on one channel. |
| 15:35.15 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 15:35.24 | hrolf | Greenlight: And if I Dial, another channel will be used. |
| 15:35.44 | hrolf | Greenlight: Means, 2 channel for one call. When I can just free my channels by REFERiing |
| 15:36.10 | hrolf | Greenlight: The agent is on the Avaya IP PBX. So there is no need for the call to remain on Asterisk after transfer. |
| 15:36.20 | igcewieling1 | hrolf: That seldom works as expected. Asteirsk is pretty bad at "transferring" calls. |
| 15:36.28 | Greenlight | Except the Avaya isn't understanding the REFER |
| 15:36.44 | Greenlight | Or Asterisk isn't sending it correctly |
| 15:36.50 | igcewieling1 | hrolf: SIP or PRI handoff between Asterisk and the Avaya? |
| 15:37.08 | hrolf | Greenlight: I tried doing it with two Asterisk servers. It worked properly. Now it is not working with Avaya. |
| 15:37.23 | Greenlight | So, the issue is with the Avaya |
| 15:37.45 | hrolf | Greenlight: Yes. But can it be determined by looking at the logs, what exactly is the issue? |
| 15:38.39 | Greenlight | Remote host can't match request REFER to call 'f3b1f16c15af65392a46ff7ce19d7ee1@192.168.20.1'. Giving up |
| 15:39.14 | igcewieling1 | j |
| 15:40.12 | Greenlight | 1.40 is your asterisk box, and 20.1 is your AvayA? |
| 15:40.23 | hrolf | Greenlight: 1.6 is Asterisk |
| 15:40.35 | Greenlight | Sorry, IP's |
| 15:40.45 | hrolf | Greenlight: Yes. |
| 15:40.47 | Greenlight | 192.168.X.X |
| 15:40.56 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:41.03 | hrolf | Greenlight: Yes. 1.40 is Asterisk and 20.1 is Avaya. |
| 15:41.06 | Greenlight | And, 1.6 is quite old with many flaws. You hould upgrade |
| 15:42.05 | *** join/#asterisk jmls (~julian@77.107.171.82) |
| 15:42.26 | jmls | struggling with nat and outbound proxies |
| 15:42.29 | jmls | ok. |
| 15:42.49 | igcewieling1 | jmls: drop the outbound proxy config and things will get easier |
| 15:42.53 | jmls | heh |
| 15:42.58 | jmls | can't |
| 15:42.59 | jmls | :( |
| 15:43.09 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:43.09 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:43.10 | igcewieling1 | I'm not joking. dial the proxy with Dial() and don't configure a proxy in sip.conf |
| 15:43.31 | Greenlight | It seems the REFER is sending it to "avaya/252@192.168.1.40" |
| 15:43.58 | igcewieling1 | Wait, we have a chan_avaya??? 8-) |
| 15:44.06 | jmls | I have a problem with sangoma lyra set up |
| 15:44.07 | jmls | http://wiki.sangoma.com/NCA-and-NAT#main-issue-with-firewall--nca---sdp |
| 15:44.07 | Greenlight | No.... |
| 15:44.10 | Greenlight | We dont |
| 15:44.14 | Greenlight | Hence it's not working |
| 15:44.52 | jmls | so I have dialer->nca->firewall, which apparently is not supported |
| 15:45.16 | [TK]D-Fender | jmls: Correct we support precisely none of those components :)] |
| 15:45.38 | jmls | weeeellllll the dialler is asterisk ;) |
| 15:46.00 | Greenlight | hrolf: See -> Refer-To: <sip:avaya/252@192.168.1.40> |
| 15:46.02 | hrolf | Greenlight: Yes it is supposed to send it to 252 which is the SIP Softphone. |
| 15:46.09 | igcewieling1 | jmls: not using Asteria's software are you? |
| 15:46.14 | hrolf | Greenlight: i.e. in Avaya IP Office. |
| 15:46.15 | jmls | and the solution is dialler->nca->asterisk->firewall |
| 15:46.24 | [TK]D-Fender | jmls: So what problem does Asterisk have there? :) |
| 15:46.26 | jmls | igcewieling1: no .. |
| 15:46.27 | Greenlight | But this is a call *from* the Avaya |
| 15:46.32 | igcewieling1 | jmls: I'm still not seeing the need for a proxy |
| 15:46.33 | hrolf | Greenlight: Yes. |
| 15:46.37 | Greenlight | Isn't 1.40 Asterisk ? |
| 15:46.45 | jmls | that's the point. I tried to narrow down the problem |
| 15:46.45 | hrolf | Greenlight: 1.40 is Asterisk. |
| 15:46.58 | jmls | so now I have asterisk->asterisk->firewall |
| 15:46.59 | Greenlight | So, why are you referring the call to that desitation, if you want it to goto the avaya |
| 15:47.24 | Greenlight | Also, I doubt the avaya will understand your asterisk trunk names such as "avaya" |
| 15:47.39 | Greenlight | You really want to refer to "252@192.168.20.1" I would imagine |
| 15:47.47 | Greenlight | And *hope* the Avaya then understands |
| 15:48.02 | igcewieling1 | jmls: great! 1) disable SIP ALG and SPI on the firewall 2) set your externip=, set your localnet=, and nat=yes |
| 15:48.08 | igcewieling1 | 3) profit! |
| 15:48.23 | jmls | 2) already done |
| 15:48.23 | hrolf | Greenlight: Avaya sends the call to our asterisk server. It then entertains the call on FastAGI application. |
| 15:48.33 | igcewieling1 | you should have localnet= options for any networks you talk to which are not outside the firewall |
| 15:48.36 | jmls | 1) will look into that. Using watchguard x510 |
| 15:48.38 | hrolf | Greenlight: Now after the IVR runs, we are supposed it to send it back to an agent on Avaya IP softphone. |
| 15:48.49 | hrolf | Greenlight: So we send it back to the Avaya IP office agent. |
| 15:48.50 | Greenlight | mmhmm |
| 15:49.19 | igcewieling1 | jmls: something like http://www.watchguard.com/help/docs/webui/11-xtm/en-US/index_CSH.html#en-US/proxies/sip/sip_proxy_about_c.html |
| 15:50.22 | mic_ | Greenlight: installing OpenVox made DAHDI |
| 15:50.27 | mic_ | Greenlight: seems to have fixed the issue. |
| 15:50.46 | igcewieling1 | where is the pastebin of hrolf's CLI output of a failed call? |
| 15:51.02 | Greenlight | Some versions of dahdi are not compatible with all the cards, and sometimes manufactueres like Openvox have a custom build |
| 15:51.15 | Greenlight | http://pastie.org/8124812 <-- |
| 15:51.35 | igcewieling1 | Digium tests DAHDI cards. They REALLY don't care of your OpenVox card works or not. |
| 15:51.37 | Greenlight | Call *from* Avaya to Asterisk |
| 15:51.38 | jmls | igcewieling1: thanks for the links and info. Will digest and try to implement. |
| 15:51.42 | jmls | ta |
| 15:51.53 | igcewieling1 | Greenlight: I'm missing the Dial |
| 15:52.11 | Greenlight | He's not using Dial |
| 15:52.37 | igcewieling1 | Greenlight: then I'm not helping. 8-) |
| 15:52.45 | Greenlight | I did tell him this |
| 15:53.05 | Greenlight | He's trying to avoid getting extra avaya licenses |
| 15:53.18 | igcewieling1 | Greenlight: or even contacting Avaya tech support? |
| 15:53.34 | Greenlight | £10 says they tell him to get more licenses :) |
| 15:53.37 | igcewieling1 | Anyway, this is what happens when you use commercial PBXs. I don't have a whole lot of sympathy. |
| 15:53.52 | Greenlight | Yup tis very true |
| 15:54.13 | Greenlight | If I was him, I'd be at the stage of ripping the Avaya out completely |
| 15:54.17 | drmessano | "Hello, Avaya Tech Support, How can I help you?" "Yes, I am trying to send calls between my Avaya system and the Asterisk open sou..." *CLICK* |
| 15:54.24 | drmessano | GFLWT |
| 15:54.37 | Katty | hi danny |
| 15:54.50 | drmessano | Hiya |
| 15:55.23 | igcewieling1 | drmessano: you missed the part about the credit card info |
| 15:55.31 | hrolf | igcewieling1: Asterisk SIP debug log http://pastie.org/8124812 |
| 15:55.44 | igcewieling1 | hrolf: (11:52:37 AM) igcewieling1: Greenlight: then I'm not helping. 8-) |
| 15:56.16 | drmessano | igcewieling1, yep.. and that's easily handled too. I just use the LifeLock guys credit card info |
| 16:02.27 | hrolf | Okay a question. Transfer() is unsupervised transfer, right? |
| 16:02.51 | Greenlight | Yes |
| 16:03.27 | [TK]D-Fender | Call is thrown right off your server, never to return. No extra CDR, no "passthrough, nada |
| 16:04.08 | hrolf | Can we do supervised transfer in Asterisk? |
| 16:04.41 | Greenlight | Yes |
| 16:04.42 | *** join/#asterisk ibercom (4de568b5@gateway/web/freenode/ip.77.229.104.181) |
| 16:05.04 | *** join/#asterisk DBordello (~DBordello@unaffiliated/dbordello) |
| 16:05.07 | Greenlight | But you're not going to save your licneses |
| 16:05.10 | hrolf | Greenlight: How? |
| 16:05.35 | hrolf | Greenlight: How can we do a supervised transfer? |
| 16:05.46 | Greenlight | https://www.google.co.uk/search?q=asterisk+attended+transfer+howto |
| 16:06.11 | Greenlight | You'll still use TWO channels... |
| 16:06.27 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:06.37 | igcewieling1 | Has anyone ever gotten Transfer to actually work? |
| 16:07.12 | Greenlight | I've never had a need... Dial() works perfectly for all my needs when I need to send a call elsewhere |
| 16:07.55 | igcewieling1 | Greenlight: the one time I tried, it failed. I don't recall seeing anyone on this channel who managed to get Transfer to work. |
| 16:09.08 | [TK]D-Fender | Greenlight: Three is no attended transfer OFF of your server |
| 16:09.15 | Greenlight | I didn't say there was |
| 16:09.16 | [TK]D-Fender | igcewieling1: I have |
| 16:09.48 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:11.21 | hrolf | [TK]D-Fender: You mean this is not possible? I receive call from a user, I dial another user, he picks up and both of them are connected (this is supervised transfer right?) Is this not possible? |
| 16:11.45 | Greenlight | Of coures that is possible, but it'll still be going via the Asterisk box |
| 16:11.49 | igcewieling1 | [TK]D-Fender: Wow! |
| 16:12.17 | igcewieling1 | hrolf: the differences between attended/supervised and blind transfers are well documented |
| 16:12.42 | Greenlight | hrolf: Unless your goals have changed, what you are trying to do is pass the call off so the two endpoints speak to eachother SIP AND RTP. This canot be done with attened transfer. |
| 16:12.46 | hrolf | igcewieling1: So the issue in my case is what? Asterisk Transfer doesn't work? Or Avaya is not configured properly? |
| 16:13.29 | Greenlight | hrolf: I already pointed out that you're trying to Transfer to the wrong destination. |
| 16:13.32 | igcewieling1 | Greenlight: I thought Greenlight already determined the Avaya is sending an incorrect SIP packet? |
| 16:13.58 | Greenlight | It's asterisk that's sending the wrong Refer-To, from what I can see |
| 16:14.13 | Greenlight | I'm not sure what arguments you're passing to Transfer though |
| 16:14.19 | igcewieling1 | Until that is fixed all discussion is simply technical masturbation. |
| 16:14.28 | hrolf | Greenlight: If I replace the Avaya with Asterisk, then it works. Then why is it the wrong destination? |
| 16:15.11 | hrolf | Greenlight: I mean two Asterisk servers. And Refer-To from one to another. |
| 16:15.27 | Greenlight | hrolf: I don't know - you've not shown that SIP trace or the configs |
| 16:15.41 | hrolf | Greenlight: Which one, two Asterisk servers? |
| 16:15.54 | Greenlight | All I've said is you're trying to pass the call to an avaya extension but you're REFERing it to the IP of the asterisk box. |
| 16:16.10 | Greenlight | hrolf: Yes |
| 16:16.39 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:16.58 | Greenlight | wonders how expensive Avaya SIP licenses have got these days to warrent so much time spent on it |
| 16:17.40 | igcewieling1 | Greenlight: doesn't matter, look at how much people try to avoid paying $10/channel for g729 licenses. |
| 16:18.08 | Greenlight | Heh, true |
| 16:18.15 | hrolf | Greenlight: Actually I'm under the assumption that sending it to sip:avaya/252@asteriskIP would send it to Avaya |
| 16:18.23 | igcewieling1 | We finally gave up on adding more and more g729 licenses and bought transcoder cards with 120 channels of transcoding awsomeness. |
| 16:18.31 | hrolf | Greenlight: Because sip/avaya's host=Avaya |
| 16:18.40 | Greenlight | [05:18pm] <hrolf> Greenlight: Actually I'm under the assumption that sending it to sip:avaya/252@asteriskIP would send it to Avaya <-- Why |
| 16:18.48 | igcewieling1 | hrolf: but Asterisk is not seeing that as the host. |
| 16:18.51 | Greenlight | "@asteriskIP" |
| 16:18.59 | Greenlight | ... |
| 16:19.06 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:19.19 | igcewieling1 | if it was seeing it as a HOST then it would be sip:252@avayaip, wouldn't it? |
| 16:19.21 | hrolf | Greenlight: Because let's say I want to dial 222 on Avaya IP, I would do SIP/avaya/222 |
| 16:19.35 | igcewieling1 | hrolf: correct, but that is not what you are doing.\ |
| 16:19.42 | Greenlight | Maybe it's the heat, but this just seems nuts. |
| 16:19.44 | igcewieling1 | you are doing some other Dial statement. |
| 16:19.56 | igcewieling1 | hrolf: SHOW US the CLI OUTPUT of the DIAL LINE. |
| 16:20.18 | Greenlight | He's not using DIal .. |
| 16:20.23 | Greenlight | He's still on SIP REFER by using Transfer |
| 16:20.24 | igcewieling1 | Greenlight: (12:19:21 PM) hrolf: Greenlight: Because let's say I want to dial 222 on Avaya IP, I would do SIP/avaya/222 |
| 16:20.32 | igcewieling1 | Well then show us the Transfer! |
| 16:20.32 | Greenlight | Oh |
| 16:20.50 | Greenlight | SIP/avaya/222 is Asterisk syntax. |
| 16:22.05 | igcewieling1 | you can also do SIP/222@avaya Maybe Transfer wants a different format than Dial |
| 16:25.17 | hrolf | igcewieling1: SIP/222@avaya also fails |
| 16:25.22 | hrolf | igcewieling1: -- Executing [888@access:1] Transfer("SIP/avaya-0000005c", "SIP/252@avaya") in new stack -- Auto fallthrough, channel 'SIP/avaya-0000005c' status is 'ANSWER' [Jul 9 21:21:28] WARNING[27072]: chan_sip.c:18722 handle_response_refer: Remote host can't match REFER request to call '6ba214308e469b09d41723a0659d76d2@192.168.20.1'. Giving up. |
| 16:26.38 | igcewieling1 | hrolf: even more stuff pointing at a problem with your Avaya |
| 16:26.52 | *** join/#asterisk hrolf (b6b47dc4@gateway/web/freenode/ip.182.180.125.196) |
| 16:27.03 | hrolf | igcewieling1: Sip/222@avaya doesn't works. |
| 16:27.06 | igcewieling1 | hrolf: even more stuff pointing at a problem with your Avaya |
| 16:27.06 | hrolf | igcewieling1: -- Executing [888@access:1] Transfer("SIP/avaya-0000005c", "SIP/252@avaya") in new stack -- Auto fallthrough, channel 'SIP/avaya-0000005c' status is 'ANSWER' [Jul 9 21:21:28] WARNING[27072]: chan_sip.c:18722 handle_response_refer: Remote host can't match REFER request to call '6ba214308e469b09d41723a0659d76d2@192.168.20.1'. Giving up. |
| 16:27.27 | hrolf | igcewieling1: Actually I just want to be sure that it is Avaya problem.\ |
| 16:27.39 | hrolf | igcewieling1: The Avaya guys are not accepting. |
| 16:27.59 | *** part/#asterisk igcewieling1 (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:31.50 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:33.15 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:33.15 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:42.55 | *** join/#asterisk Ryushin (~Ryushin@gateway/tor-sasl/ryushin) |
| 16:45.07 | Ryushin | Is there a way to set up a redundant outbound sip proxy manually using the configuration files and not using something like FreePBX? https://support.flowroute.com/entries/20056998-Redundant-Proxy-Setup |
| 16:50.50 | [TK]D-Fender | Ryushin: That does not require freepbx.... |
| 16:53.45 | Ryushin | [TK]D-Fender: I would not think it would require it. I've been googling a bit and I cannot find examples on how to set it up manually. Still googling right now though. |
| 16:54.17 | Ryushin | Does this really matter since Flowroute uses SRV records? |
| 16:54.23 | [TK]D-Fender | Ryushin: It's RIGHT THERE |
| 16:54.30 | [TK]D-Fender | Ryushin: that IS the config |
| 16:56.09 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:56.43 | Ryushin | I see that and how it's doing it. I did not see how to specify in the extensions.conf file how to set one up as primary and the other as secondary. |
| 16:56.59 | Ryushin | I just don't want a round robin situation to take place. |
| 16:57.26 | *** join/#asterisk aruntomar (~Thunderbi@49.248.159.160) |
| 16:57.28 | [TK]D-Fender | Ryushin: You HAVE to do that in the dialplan. That's the entire point |
| 16:57.41 | [TK]D-Fender | Ryushin: You should really read the instructions on that page. |
| 16:57.58 | [TK]D-Fender | Ryushin: This is the alternative to using DNS-SRV which * already supports. |
| 16:58.33 | [TK]D-Fender | Ryushin: And as long as you don't chace as it says... it should work. You would simply use the hostname they offer that does the rotation. |
| 16:58.35 | random_cat | asterisk 11.4.0 with freepbx 2.11.0.4 on debian 7 is telling me "Failed to execute '/var/lib/asterisk/agi-bin/dialparties.ago': File does not exist." How to persuade asterisk to look in /usr/share/asterisk/agi-bin where it will find the appropriate file? |
| 16:59.02 | random_cat | s/ago/agi/ |
| 16:59.42 | Ryushin | [TK]D-Fender: Okay, thanks. I did read it, but I like doing things manually a bit. But I guess, that is the whole purpose of SRV records and if they have it set up correctly on their end, * will do the rest. |
| 16:59.43 | [TK]D-Fender | ~freepbx |
| 16:59.43 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:59.47 | [TK]D-Fender | random_cat: ^ |
| 17:00.24 | random_cat | thanks, [TK]D-Fender |
| 17:11.55 | *** join/#asterisk jzaw (~jzaw@2001:8b0:7:0:5054:ff:fe8e:3b24) |
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| 17:42.36 | *** join/#asterisk derjanni (~derjanni@ip-178-202-27-28.unitymediagroup.de) |
| 17:42.54 | derjanni | Good afternoon everybody! SMS in *11 does have some own variables, right? |
| 17:43.00 | *** join/#asterisk igcewieling1 (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:43.31 | derjanni | I want to run a script with SHELL that send a PSTN SMS to an e-mail address. |
| 17:43.55 | derjanni | I could read the mrtx folder, but I wanted to check whether there is an easier way |
| 17:45.03 | igcewieling1 | derjanni: once you have the SMS message in Asterisk, using System to e-mail out the message is a good and easy way |
| 17:46.54 | derjanni | Im using a script that delivers it to the DECT phone, but also sends emails to two people |
| 17:47.35 | derjanni | i saw that there was a variable ${INCOMING_SMS} in a documentation and was wondering whether it really exists |
| 18:10.04 | igcewieling1 | *grumble* "Transfer-Encoding: chunked" is going to drive me insane. |
| 18:34.01 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
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| 19:22.00 | Ryushin | Will Asterisk compile without Dahdi? I'm running a pure sip solution now and I don't need to support PSTN or Analog fax any longer. |
| 19:23.18 | [TK]D-Fender | Obviously |
| 19:24.31 | brad_mssw | I would think you'd still want dahdi_dummy for proper timing support (meetme, etc) |
| 19:24.54 | Ryushin | [TK]D-Fender: Just curious. Are you still a fan of Sangoma over Digium for hardware. |
| 19:25.08 | [TK]D-Fender | yes |
| 19:25.44 | Ryushin | brad_mssw: Yea, I use meetme. So I guess I'm compiling Dahdi then. |
| 19:26.16 | brad_mssw | I just google'd it ... apparently new asterisk versions don't necessarily need it ... gah, been a long time since I looked down pure-sip ... we ditched that years ago and brought in a PRI via a Sangoma card |
| 19:26.32 | brad_mssw | https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces |
| 19:27.46 | Ryushin | brad_mssw: Gee, and I just ditched the PRI and the Songoma cards we have to move to Flowroute and pure SIP. |
| 19:32.04 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 19:34.43 | *** join/#asterisk BThomson (55b74b83@gateway/web/freenode/ip.85.183.75.131) |
| 19:34.51 | BThomson | Hi all |
| 19:36.34 | BThomson | Is anyone in the mood to help me? I have trouble recording voicemails in asterisk. It's only creating empty files. Google didn't help. Any ideas? |
| 19:37.09 | drmessano | brad_mssw, you don't need DAHDI anymore except for MeetMe. MeetMe uses DAHDI as a MUX. That's pretty much it. |
| 19:37.35 | igcewieling1 | brad_mssw: ConfBridge does not need a mixing service (dahdi), Meet me requires something to mix the audio (dahdi). Dahdi is no longer require for timing, only audio mixing for MeetMe |
| 19:38.18 | igcewieling1 | Ryushin: you likely made a mistake if you care about reliability |
| 19:41.11 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 19:41.59 | Ryushin | igcewieling1: I gave the client that currently has the system all the options and pros and cons. Cost was a major factor for them. The own has been using magicjack for their home phone for the last 18 months and has been happy with it. They will be seeing about a $400 a month in savings with using Flowroute over a PRI. |
| 19:43.13 | drmessano | We've been pure SIP for 2 years and it's been as reliable as copper |
| 19:43.14 | igcewieling1 | Yes, running your phones over the internet is cheaper. |
| 19:43.44 | Ryushin | I understand that a PRI is a absolute proven and rock solid technology. It's why they have been using it for the last decade. But they want to at least try this route. It will take three months of using Flowroute to reach the break even point of porting to them. |
| 19:43.45 | igcewieling1 | I was only talking about how reliable, but if they were using MagicJack they are crazy anyway. |
| 19:44.00 | drmessano | I wouldn't run this over a $25 cable modem from Comcast though.. Get a solid connection |
| 19:44.10 | jmetro | ew, grandstream |
| 19:44.19 | igcewieling1 | hey, jmetro! |
| 19:44.31 | igcewieling1 | were you the one interested in a web transcoder? |
| 19:44.31 | Ryushin | It's business cable through Comcast. |
| 19:44.44 | drmessano | I ditched PRI because AT&T should be the first against the wall when the revolution comes |
| 19:44.58 | jmetro | igcewieling1: i cant remember. maybe |
| 19:44.59 | Ryushin | It was not me. |
| 19:45.02 | jmetro | Oh, i was. |
| 19:45.22 | igcewieling1 | jmetro: http://140.239.234.172/transcode/ |
| 19:45.37 | Ryushin | I was hoping the FED would be the first against the wall. |
| 19:45.43 | jmetro | Nice |
| 19:46.08 | igcewieling1 | including g729 7-) |
| 19:46.34 | igcewieling1 | drmessano: there is nothing wrong with using Evil companies to attain non-evil goals. |
| 19:46.35 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 19:46.57 | jmetro | My name is Jmetro and i approve of this transcoder |
| 19:47.01 | drmessano | Ryushin, I have had mixed experiences with Comcast Business. If they do it right, it will be reliable as hell. If you're in a spot where they're stringing twine and slapping duct tape on everything, it will be horrible. The better hardware seems to make a huge difference, and I imagine there's some prioritization going on |
| 19:47.13 | Ryushin | Anyone trying the open g.729 for asterisk? http://www.opentelecoms.org/codecs-open-source-g729-g723.1-asterisk |
| 19:47.14 | jmetro | My name is Jmetro and this is my favorite transcoder on the citadel. |
| 19:47.30 | drmessano | Ryushin, I moved from Comcast Residental to Comcast Business, and back, and I can tell there's a big difference.. even on the same copper. |
| 19:48.03 | [TK]D-Fender | Ryushin: That does not license you you to run any actual code |
| 19:48.11 | igcewieling1 | Ryushin: Talking about that software or posting links is one of the very few things which will get you banned from the channel |
| 19:48.18 | Ryushin | They have been on Comcast Business for the last 2 years with only one issue in that time. |
| 19:48.20 | [TK]D-Fender | Ryushin: And is something you'd be srtongly advised to to carry on about in here |
| 19:48.33 | [TK]D-Fender | strongly* |
| 19:49.34 | jmetro | hm, if its not illegal, theres nothing wrong with an open g729? unless i've got the idea wrong |
| 19:49.40 | Ryushin | Oh, I thought it was a pure open source cold room implementation that was not needed to have a patent. |
| 19:49.52 | igcewieling1 | jmetro: does it matter if Digium will ban you from the channel for talking about it? |
| 19:50.08 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:50.18 | igcewieling1 | jmetro: does it matter since Digium will ban you from the channel for talking about it? |
| 19:50.28 | jmetro | igcewieling1: i think that means i have the wrong idea |
| 19:50.35 | igcewieling1 | (I support this BTW) |
| 19:51.00 | igcewieling1 | jmetro: it still isn't licensed. |
| 19:51.01 | [TK]D-Fender | Ryushin: That is incorrect |
| 19:54.11 | drmessano | The "open source G729" is still an implementation of a licensed codec... Just the same as generic medication are implementations of patented formulations |
| 19:54.33 | BThomson | Has anyone experience with asterisk creating empty voicemail recording files? |
| 19:54.43 | jmetro | drmessano: thats a better explanation that i understand |
| 19:54.57 | jmetro | drmessano: i dont deal with legal things, i just make stuff works :3 |
| 19:56.48 | drmessano | I think the misunderstanding is the actual legal bit... It's not an issue of pirated software or writing software that replicates the function of paid software, it's a matter of the actual function of that software _across_the_wire_ is what is patented. |
| 19:56.59 | [TK]D-Fender | BThomson: If its empty it's because no audio made it, not because of a voicemail issue |
| 19:59.00 | BThomson | [TK]D-Fender: Alright, i can see that. So, what is the reason? |
| 19:59.51 | [TK]D-Fender | BThomson: Well... you've told us nothing about your setup.... |
| 20:00.00 | BThomson | [TK]D-Fender: Furthermore, when I say empty, I mean that the files 0kb in size |
| 20:00.11 | [TK]D-Fender | BThomson: That leaves us with "a whole w=hack of possibilities" |
| 20:01.53 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:02.42 | BThomson | [TK]D-Fender: True... I'm running Debian 2.6.32 with asterisk 1.8 |
| 20:03.21 | [TK]D-Fender | BThomson: I would think the details directly pertaining to this call would be where you'd aim first... |
| 20:03.23 | BThomson | [TK]D-Fender: Calls come in through SIP |
| 20:03.38 | [TK]D-Fender | BThomson: From where? Past what? |
| 20:03.48 | BThomson | [TK]D-Fender: I've got several Cisco 509g working |
| 20:04.13 | BThomson | [TK]D-Fender: Uhm |
| 20:04.14 | [TK]D-Fender | BThomson: And the part that isn't working? |
| 20:04.29 | BThomson | [TK]D-Fender: Well, from "outside" to one phone |
| 20:04.51 | BThomson | [TK]D-Fender: After 5 rings the call is redirected to the VM |
| 20:05.10 | BThomson | [TK]D-Fender: You can hear the message, then the beep, etc. |
| 20:05.15 | [TK]D-Fender | I'd look at the complete call attempt with SIP debug enabled.... and also fully describe the networking between the two ends... |
| 20:05.28 | BThomson | [TK]D-Fender: ok... |
| 20:05.36 | BThomson | [TK]D-Fender: Just a second |
| 20:06.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.57) |
| 20:06.43 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 20:06.43 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:06.43 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:06.43 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 20:06.43 | *** mode/#asterisk [+o sruffell] by orwell.freenode.net |
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| 20:06.54 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 20:11.48 | BThomson | [TK]D-Fender: Alright |
| 20:12.18 | BThomson | [TK]D-Fender: So, the debug output does not mention problems with the file creation etc. |
| 20:12.52 | BThomson | [TK]D-Fender: Only when trying to play it back, it says: WARNING[30293]: file.c:483 ast_filehelper: File /var/spool/asterisk/voicemail/company/101/tmp/xbui42.ulaw detected to have zero size |
| 20:14.13 | BThomson | [TK]D-Fender: The created sound file (ulaw) has 0 bytes, the txt file is bigger (301 byte) |
| 20:14.28 | jmls | igcewieling1: got it working ! |
| 20:14.45 | BThomson | [TK]D-Fender: I tried every available sound format, it's alwasy empty |
| 20:14.49 | jmls | thanks. I changed the firewall rules for SIP-ALG and I now don't need the proxy |
| 20:14.57 | jmls | many thanks for your pointers |
| 20:15.18 | igcewieling1 | jmls: got what working? |
| 20:15.33 | jmls | the ability to remove the outboundproxy |
| 20:15.39 | jmls | you helped me earlier |
| 20:15.41 | igcewieling1 | BThomson: sounds like the partition is full |
| 20:15.54 | igcewieling1 | jmls: ah, yes. nat confusion. |
| 20:16.17 | igcewieling1 | SIP ALG breaks Asterisk's NAT |
| 20:16.47 | igcewieling1 | like adding two negative numbers does not give you a positive numbers, having two devices doing NAT translation does give you working audio |
| 20:16.52 | BThomson | igcewieling1: Nope, 99% empty... |
| 20:16.59 | igcewieling1 | does NOT give you... |
| 20:17.20 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 20:17.30 | igcewieling1 | BThomson: an EMPTY wav file is still something like 68 bytes (for the header) |
| 20:17.51 | zrzerenato | how can i export my Agents to compare if exist duplicate users... |
| 20:18.39 | BThomson | igcewieling1: That's my thinking. So i'm guessing, it's not that it won't record anything, it |
| 20:18.54 | BThomson | igcewieling1: is just missing the right codec |
| 20:19.07 | BThomson | igcewieling1: or something similar |
| 20:19.07 | [TK]D-Fender | It isn't the codec. |
| 20:19.11 | [TK]D-Fender | Three is no DATA |
| 20:19.13 | zrzerenato | can any one help me please? |
| 20:19.23 | *** join/#asterisk tapout_ (~tapout@unaffiliated/tapout) |
| 20:19.38 | igcewieling1 | zrzerenato: I doubt it, but stick around just in case. |
| 20:20.03 | BThomson | [TK]D-Fender: ok... so what would cause that? |
| 20:20.08 | igcewieling1 | BThomson: do you have SELinux disasabled? |
| 20:20.29 | igcewieling1 | if not, you should. |
| 20:21.08 | [TK]D-Fender | BThomson: packets not arriving <- |
| 20:21.28 | *** join/#asterisk cmendes0101| (~cmendes01@72.1.46.254) |
| 20:22.56 | igcewieling1 | [TK]D-Fender: shouldn't there be a wav header, even if there is no audio? |
| 20:22.57 | BThomson | igcewieling1: SELinux is not running |
| 20:23.09 | igcewieling1 | BThomson: that was my only idea. 8-( |
| 20:23.23 | [TK]D-Fender | igcewieling1: Dunno... but I asked for details and here we are with nothing. |
| 20:23.56 | BThomson | [TK]D-Fender: Why wouldn't they arrive? Calls are working fine with the phones |
| 20:24.36 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 20:24.36 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:24.36 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:24.36 | *** mode/#asterisk [+o sruffell] by orwell.freenode.net |
| 20:26.36 | BThomson | [TK]D-Fender: I can't give you much to work with. The files have no header: -rw-r--r-- 1 root root 0 9. Jul 22:11 msg0004.ulaw |
| 20:26.48 | [TK]D-Fender | BThomson: I asked for NETWORKING DETAILS |
| 20:26.59 | [TK]D-Fender | For which.. I am out of time for now |
| 20:27.06 | [TK]D-Fender | BBL |
| 20:27.25 | *** join/#asterisk russum (~russum@83-131-73-118.adsl.net.t-com.hr) |
| 20:27.28 | *** part/#asterisk russum (~russum@83-131-73-118.adsl.net.t-com.hr) |
| 20:27.41 | BThomson | igcewieling1: A little high strung, Fender? |
| 20:27.47 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:29.55 | jmetro | is htere a var_dump option for asterisk dialplan? |
| 20:30.41 | igcewieling1 | jmetro: Dumpchan() |
| 20:30.47 | igcewieling1 | closest thing asterisk has |
| 20:31.09 | igcewieling1 | BThomson: He has little patience for people who ask for his help not providing him the information, but he is VERY good with Asterisk. |
| 20:31.36 | igcewieling1 | jmetro: We put a dumpchan just before the dialplan ends for calls as standard practice |
| 20:31.42 | BThomson | igcewieling1: :( |
| 20:32.28 | jmetro | igcewieling1: that seems like log-file overkill. |
| 20:32.34 | BThomson | igcewieling1: Alright, thanks anyway |
| 20:33.31 | igcewieling1 | jmetro: you'll learn there is no such thing |
| 20:38.02 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:38.58 | Cuzner | Woot, finally got my VVX 1500 working properly at my desk :) |
| 20:39.18 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 20:40.59 | jmls | ok, time for another stupid question |
| 20:41.15 | jmls | does asterisk 11 strip any sip headers at all ? |
| 20:41.33 | jmls | I have a message sent to my * box that contains some headers, |
| 20:41.46 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 20:41.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:41.46 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:41.46 | *** mode/#asterisk [+o sruffell] by orwell.freenode.net |
| 20:41.49 | jmls | but a sip debug in asterisk shows that these headers are not there |
| 20:42.08 | jmls | I know that I sound like a typical user, but .. |
| 20:42.30 | jmls | I'm kinda at a loss to know how / where to start looking |
| 20:42.34 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 20:42.51 | igcewieling1 | jmls: Asterisk does not generally remove headers, however proxies often remove headers. Asterisk will NOT automatically pass the headers from the incoming leg of the call to the outbound leg of the call. |
| 20:43.25 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:44.54 | jmls | I have asterisk <-> lyra proxy, and the lyra is sending a message to asterisk with the header "CPD-Result: Voice" |
| 20:45.22 | jmls | however, on asterisk, sip debug does not show this header at all |
| 20:46.46 | igcewieling1 | jmls: you can confirm the proxy is sending the header by using tcpdump or wireshark |
| 20:46.55 | jmls | ok, thanks |
| 20:47.48 | carrar | ngrep FTW |
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| 20:50.47 | *** part/#asterisk aelliott22 (~aelliott@198.41.29.45) |
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| 21:05.54 | jmls | igcewieling1: got the following from tcpdump on the asterisk box : SIP/2.0 200 OK [snip] CPD-Result: Voice |
| 21:06.06 | jmls | so it's reaching asterisk |
| 21:06.48 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.166) |
| 21:07.26 | *** join/#asterisk klasstek (~Adium@173-164-56-25-colorado.hfc.comcastbusiness.net) |
| 21:08.39 | klasstek | hello. I have a question regarding registration in sip.conf. |
| 21:09.13 | klasstek | The user is something like CT_10001@CT10001.ficticiousdomain.com |
| 21:09.45 | klasstek | and the registar is somethign like siptrunk.providerdomain.com |
| 21:10.02 | klasstek | I assume from the docs that something like this should work |
| 21:10.55 | klasstek | register => CT_10001@CT1001.ficticousdomain.com:supersecretpass@siptrunk.providerdomain.com |
| 21:11.22 | klasstek | but all I see on the registrar is CT10001@siptrunk.providerdomain.com |
| 21:11.48 | klasstek | when I should see mailto:CT_10001@CT10001.ficticiousdomain.com |
| 21:12.03 | klasstek | scratch the mailto part of that |
| 21:12.36 | klasstek | Asterisk is 1.6.0.26 |
| 21:13.01 | klasstek | Can anyone point me in the right direction? |
| 21:14.29 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 21:16.18 | jmls | wtf |
| 21:17.03 | jmls | it's just appeared :( |
| 21:23.47 | [TK]D-Fender | klasstek: 1.6.x is no longer supported and there were fixes to formatting like this since then |
| 21:24.33 | klasstek | I saw a bug that referenced a documentation change, but not a functionality change. |
| 21:24.51 | jmetro | klasstek: what happens when you update asterisk to ast11? |
| 21:25.20 | *** join/#asterisk russum (~russum@83-131-73-118.adsl.net.t-com.hr) |
| 21:25.23 | [TK]D-Fender | 1.6.0 branch is dead |
| 21:25.27 | [TK]D-Fender | 1.6.1 branch is dead |
| 21:25.31 | [TK]D-Fender | 1.6.2 branch is dead |
| 21:25.39 | [TK]D-Fender | Asterisk 10 Branch is dead |
| 21:25.48 | [TK]D-Fender | that's 4 branches since then gone |
| 21:25.52 | klasstek | TK. point made |
| 21:26.03 | [TK]D-Fender | 11 = LTS and effectively dates 1.8 |
| 21:26.13 | [TK]D-Fender | And 12 is about to enter feature-freeze |
| 21:26.43 | jmetro | [TK]D-Fender: That was quick |
| 21:26.56 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 21:28.02 | klasstek | The problem is that I have customers still using 1.6.0 and I'm testing against SIP trunks on our SBC. |
| 21:28.12 | klasstek | So… if it works I can say here's how. |
| 21:28.42 | klasstek | I have 49 customers where version is not defied in the UA |
| 21:28.50 | Qwell | ~upgrade asterisk |
| 21:28.50 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 21:29.36 | klasstek | 50 or so that are 1.6.x |
| 21:29.53 | klasstek | 2 that are 1.8.x |
| 21:29.57 | klasstek | and one that is 10.3.1 |
| 21:29.58 | jmetro | cant you logn, core show version? |
| 21:30.51 | klasstek | so I'm stuck with customers that don't upgrade. |
| 21:31.12 | klasstek | I don't manage the customers systems. I just provide the SIP trunks |
| 21:31.45 | klasstek | and was trying to test with what is currently most widely deployed by my user base. |
| 21:32.43 | klasstek | without having to upgrade. or spin cycles on that, do the current versions support a user realm that does not match the sip server domain? |
| 21:33.09 | klasstek | or domain that doesn't match host if using terms from the register line doc |
| 21:34.58 | *** join/#asterisk moy (~textual@UNVLON55-1176057127.sdsl.bell.ca) |
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| 21:37.53 | jmetro | any way to shorten the delay that asterisk has built in when its waiting for additional digits for possible multiple matches? |
| 21:38.06 | jmetro | like matching "exten => 100x" and "exten => 100" |
| 21:38.30 | *** join/#asterisk russum (~russum@83-131-73-118.adsl.net.t-com.hr) |
| 21:38.35 | *** part/#asterisk russum (~russum@83-131-73-118.adsl.net.t-com.hr) |
| 21:39.51 | klasstek | http://www.voip-info.org/wiki/view/Asterisk+func+timeout seems to be what you want jmetro |
| 21:40.22 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 21:46.34 | floren | good afternoon |
| 21:50.31 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
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| 21:56.19 | jmetro | "asterisk cleanly ending" always scares the bajeebus out of me |
| 22:02.31 | ChannelZ | better than dirtily ending |
| 22:02.48 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:03.20 | jmetro | id rather it says "Closing remote connection and not stopping your asterisk process. calm down" |
| 22:07.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 22:17.35 | floren | i have a quick question related to wav49 codec |
| 22:18.27 | ChannelZ | it's really gsm |
| 22:18.43 | ChannelZ | if I remember right.. but kind of jammed in a wav wrapper |
| 22:18.52 | floren | running asterisk 11.3 on centos 6, for now to fix the ignoring unknown format 'wav49/gsm' warning, i disabled them into voicemail.conf |
| 22:19.04 | floren | ChannelZ: i see |
| 22:19.04 | *** part/#asterisk klasstek (~Adium@173-164-56-25-colorado.hfc.comcastbusiness.net) |
| 22:19.06 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 22:19.16 | floren | is this a codec dep on a centos package? |
| 22:19.42 | floren | i did a bit of searching and the solution everyone proposes is to disable them |
| 22:19.50 | ChannelZ | hmm. I think gsm is self-contained |
| 22:20.01 | floren | that will make attachments big, IMO. i would rather use wav49 |
| 22:20.30 | ChannelZ | I'm not sure how widely supported it is; Windows Media Player I believe will play it but I've no idea anywhere else |
| 22:20.58 | ChannelZ | phones probably |
| 22:22.04 | jmetro | we had issues with phones playing wav |
| 22:22.05 | jmetro | they prefer mp3 |
| 22:26.30 | ChannelZ | yeah I think my Android phone couldn't ever play wav49 but my old Palm Pre could |
| 22:26.36 | ChannelZ | It's a frankenformat |
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| 22:53.13 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:22.45 | jmetro | Anyone remember the webURL to look at a Cisco's config? |
| 23:26.20 | Xetrov` | do polycoms typically have an issue being provisioned by freepbx? |
| 23:26.59 | igcewieling1 | points Xetrov` to #FreePBX, 2 doors down and on the left. |
| 23:29.25 | Xetrov` | appreciated. |
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