| 00:02.38 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 00:13.22 | SpaceInvaders | I have a question on the documentation. The docs say runuser= rungroup=asteriskpbx but the .conf files say runuser= rungroup=asterisk. Is it arbitrary as long as I've created the ID? |
| 00:13.48 | SpaceInvaders | I just installed on Fedora 20 so I'm referring to the default .conf files I see with that installation just having completed via yum. |
| 00:15.50 | [TK]D-Fender | It should match... |
| 00:16.34 | SpaceInvaders | So there *should* be no problem running with the defaults (the ID being asterisk rather than asteriskpbx) |
| 00:17.00 | SpaceInvaders | and if everything dies I'll reformat and try again :D |
| 00:17.07 | SpaceInvaders | it's a fresh, sandbox install |
| 00:19.05 | SpaceInvaders | Thank you, TKD |
| 00:19.52 | [TK]D-Fender | Just make sure the owner fo the files matches |
| 00:20.01 | [TK]D-Fender | That aside you can call it FRED if you feel like.... |
| 00:20.15 | SpaceInvaders | It doesn't check that on 1st-run? |
| 00:20.29 | SpaceInvaders | I'll check it |
| 00:22.04 | *** join/#asterisk theron (~theron@173-29-74-236.client.mchsi.com) |
| 00:23.47 | [TK]D-Fender | No, it just does what you tell it and then the smoke starts pouring out.... |
| 00:24.02 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 00:34.24 | SpaceInvaders | Sometimes I love that. Other times... |
| 00:35.29 | SpaceInvaders | OK, the yum install on Fedora 20 seems to have taken care of that--at least everything I've checked has owner:group as asterisk:asterisk |
| 01:04.47 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 01:08.44 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 01:20.45 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:40.39 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:49.25 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.77.103) |
| 01:53.51 | ruben23 | hi guys |
| 01:57.25 | *** join/#asterisk theron (~theron@173-29-74-236.client.mchsi.com) |
| 01:59.40 | TazzNZ | hi |
| 02:01.33 | ruben23 | guys what does this means..? ---> -- Called DAHDI/g0/0433305715 <---------dialing using VoIP..? trunkline or any idea..? |
| 02:01.33 | ruben23 | <PROTECTED> |
| 02:01.59 | [TK]D-Fender | Obviously a call over DAHDI... |
| 02:02.06 | TazzNZ | that means that a call was sent to DAHDI, group 0 and the number dialed was 04333..... |
| 02:02.40 | [TK]D-Fender | That sure doesn't say SIP/IAX/PJSIP/H323/ooH323/MGCP or SCCP |
| 02:03.07 | [TK]D-Fender | So no.. not "VoIP" |
| 02:03.20 | ruben23 | so what type of connection are being used..? |
| 02:03.44 | ruben23 | maybe analog lines..? or digitial lines..? |
| 02:03.50 | TazzNZ | hard to say |
| 02:03.55 | TazzNZ | but yes - either of those |
| 02:03.57 | TazzNZ | not IP |
| 02:04.54 | ruben23 | dahdi are being used only for analog line or digital lines with hardware being plug into asterisk server to interface with those media right..? |
| 02:05.07 | TazzNZ | yip |
| 02:05.31 | ruben23 | what device si use for digitial lines like PRI and BRI..? |
| 02:05.45 | TazzNZ | also dahdi |
| 02:05.53 | TazzNZ | unless you want to know what *hardware* ? |
| 02:07.40 | ruben23 | yes what hardware types used by dahdi to interface with digitial lines |
| 02:08.17 | TazzNZ | there is quite a few - the most common (imho) is Sangoma and Digium |
| 02:08.28 | TazzNZ | (if I understand the question) |
| 02:09.00 | ruben23 | yes you got it, is it plug and play..? |
| 02:09.13 | [TK]D-Fender | No |
| 02:09.22 | [TK]D-Fender | those are PCI(?) cards |
| 02:09.35 | [TK]D-Fender | ther are some USB DAHDI devices, but they are relatively rare |
| 02:09.56 | TazzNZ | I would love to get my hands on a USB device [TK]D-Fender |
| 02:10.25 | ruben23 | ok when i have them - i plug on PCi cards..what would be the next step..? |
| 02:10.46 | [TK]D-Fender | plug your LINE into it. |
| 02:11.05 | [TK]D-Fender | Configure it. Use it. Fight. WIN. |
| 02:11.51 | TazzNZ | ruben23: I am confused.....you are placing calls via DAHDI, do you need to extend what you doing ? |
| 02:13.16 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 02:18.20 | ruben23 | imtrying to learn from an existing asterisk with Dahdi in used already adn plan to replicate the setup on another box. |
| 02:30.18 | TazzNZ | ok - I see |
| 02:31.03 | TazzNZ | http://www.digium.com/en/products/telephony-cards/analog <-- start there |
| 02:31.21 | TazzNZ | http://www.digium.com/en/products/telephony-cards <-- sorry - there :) |
| 02:33.39 | ruben23 | ok thanks let me check on this |
| 02:34.27 | ruben23 | but how about the configuration side..? |
| 02:36.01 | TazzNZ | the config is stored in /etc/dahdi/ |
| 02:36.46 | TazzNZ | you can run dahdi_genconf to generate a sample config |
| 02:36.49 | TazzNZ | from your hardware |
| 02:39.17 | *** join/#asterisk viro (~viro@108-253-70-172.lightspeed.cntmoh.sbcglobal.net) |
| 02:40.47 | [[thufir]] | offtopic I know, but what is a "skypebot"? http://sevabot-skype-bot.readthedocs.org/en/latest/ |
| 02:47.14 | stevePearPear | hi inside sip.conf, we could do register => sip1222:pw:sip_provider_domain to recieve call from sip provider |
| 02:47.29 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 02:47.31 | stevePearPear | is there any chance I could load this like dynamically through Asterisk realtime? |
| 02:47.47 | TazzNZ | stevePearPear: look into Asterisk Realtime |
| 02:47.54 | TazzNZ | it will pull that info from a database |
| 02:48.09 | TazzNZ | but it will need a "core reload" when a register is added |
| 02:48.16 | stevePearPear | yeah cuz I was reading this mailing list http://lists.digium.com/pipermail/asterisk-users/2009-November/241425.html |
| 02:48.39 | stevePearPear | it must do a sip reload? It says this would lose all current registration :( does calls get hanged up? |
| 02:48.59 | TazzNZ | current calls will carry on without interruptions |
| 02:49.51 | stevePearPear | any idea, if I have 500 register, would it take very long? |
| 02:50.12 | TazzNZ | you are registering to 500 ITSP's ? |
| 02:50.25 | TazzNZ | or you have 500 devices that register to *you* |
| 02:51.01 | stevePearPear | i have 500 devices that registered to me, however they have their own sip provider account which would thus be 500 register too |
| 02:51.22 | TazzNZ | devices registered to you will not be removed |
| 02:51.44 | TazzNZ | if that was the case, my phones (550+-) would be lost on every reload |
| 02:51.49 | TazzNZ | and we do several a day |
| 02:53.04 | stevePearPear | but your phone would initiate a re-register right? Im concern if sip reload would take a long time if there’s 500 register => :( |
| 02:53.32 | TazzNZ | they don't |
| 02:53.41 | TazzNZ | the phone doesn't know that asterisk was reloaded |
| 02:53.50 | TazzNZ | so it will try to register again at the timeout |
| 02:54.22 | stevePearPear | ic, thanks :) I will try and see how Asterisk Realtime could do for register=>xx:xx@yy.com |
| 02:56.25 | TazzNZ | how often are you changing register's that you need to put that in Realtime ? |
| 02:57.06 | TazzNZ | imho, these should not change that often....maybe 2-3 times a year |
| 03:07.40 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 03:15.38 | ruben23 | Jun 9 03:11:34] NOTICE[2614]: chan_sip.c:14983 sip_reg_timeout: -- Registration for 'yyyyxxccc@xxxx.xxx.xx.xxx' timed out, trying again (Attempt #2 <---any comment on this guys |
| 03:17.09 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 03:17.59 | ruben23 | i see this evey 5 minutes |
| 03:20.23 | stevePearPear | my register’s could change a few time per day as I’ve no controls over the account. i know this sound weird but my users have their own sip trunk accounts with the providers. I am somehow a middleman in order to support webrtc |
| 03:24.38 | TazzNZ | stevePearPear: I see :) |
| 03:24.50 | TazzNZ | ruben23: can you ping that IP ? |
| 03:25.07 | TazzNZ | from your asterisk box |
| 03:28.07 | ruben23 | TazzNZ: yes, its registerd, i can do mtr and ping it..but this message still appears- the source carrier IP is reachable |
| 03:28.36 | TazzNZ | to me, that shows that you are not registered |
| 03:28.40 | TazzNZ | and it is trying |
| 03:28.51 | TazzNZ | and it can't do it, so it's retrying |
| 03:29.08 | TazzNZ | what does "sip show register" show ? |
| 03:32.53 | ruben23 | it show the IP is registered.. |
| 03:33.54 | ruben23 | but the latency changings in a period of time |
| 03:33.55 | TazzNZ | that doesn't make sense |
| 03:34.10 | TazzNZ | by how much does the ping change ? |
| 03:34.17 | ruben23 | ms 35, ms24,ms43,ms26 |
| 03:34.30 | ruben23 | when i do ship show peers |
| 03:34.37 | TazzNZ | that is ok - I wouldn't want rtp to run over than, but hey |
| 03:35.26 | TazzNZ | ruben23: you are saying that it *is* registered, but the error you pasted shows that it timed out - that is *very* weird |
| 03:35.56 | TazzNZ | are you sure you are looking at the same IP's ? |
| 03:36.01 | ruben23 | its appearing every 5 minutes..always atetmpt #2...but i check sip show peers - its registerde with latency |
| 03:36.13 | TazzNZ | oh right |
| 03:36.21 | TazzNZ | so after 5 mins, it shows this |
| 03:36.29 | TazzNZ | and then after the 2nd try, it registered ? |
| 03:36.35 | TazzNZ | it is* |
| 03:37.22 | ruben23 | its always registered- it does not disconnect - but this stil appears like 5-6 minutes interval.. |
| 03:37.49 | TazzNZ | I would talk to the provider and find out what your register time out should be |
| 03:38.06 | TazzNZ | also find out if they have a limit on the number of sip messages that you can send to them per second |
| 03:38.21 | TazzNZ | (i had a provider black list us for exceeding a limit like this) |
| 03:56.52 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 04:02.24 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
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| 04:44.48 | [TK]D-Fender | [23:15]ruben23Jun 9 03:11:34] NOTICE[2614]: chan_sip.c:14983 sip_reg_timeout: -- Registration for 'yyyyxxccc@xxxx.xxx.xx.xxx' timed out, trying again (Attempt #2 <---any comment on this guys <--- No, we'd need to see the actual full attempt to have something to comment on |
| 04:45.08 | [TK]D-Fender | [23:36]ruben23its appearing every 5 minutes..always atetmpt #2...but i check sip show peers - its registerde with latency <- peer has nothing to do with registration attempt |
| 05:00.00 | *** join/#asterisk zopsi (~zopsi@zopsi.com) |
| 05:32.01 | zopsi | Can anyone take a look at the attached log, pjsip.conf, and extensions.lua and see if you can help me out with what seems like a basic pjsip configuration problem? http://pastebin.com/QSSSeE3c |
| 05:32.19 | zopsi | I get the no endpoint when I have incoming calls as well. |
| 05:33.33 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 05:34.02 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 05:35.25 | zopsi | Also just adding that I'm running Asterisk12 from SVN with chan_pjsip and no chan_sip. |
| 05:38.04 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 06:43.18 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 06:50.37 | stevePearPear | hi im having a problem with audio in asterisk :( The call went, my asterisk in inside AWS. and I used rtp set debug on I noticed this |
| 06:50.39 | stevePearPear | Sent RTP packet to 192.168.0.136:64408 (type 00, seq 007779, ts 1794783912, len 000160) |
| 06:50.53 | stevePearPear | where 192.168.0.136 is my PC internal IP |
| 06:51.08 | stevePearPear | its not sending to my public ip :( |
| 06:52.02 | Chainsaw | Then you need to declare what your public IP is. |
| 06:58.17 | stevePearPear | sorry, but do u know where can I declare it ? |
| 06:58.24 | stevePearPear | i dun see the config from x-lite :( |
| 06:58.27 | stevePearPear | nor SIPML5 |
| 06:59.14 | Chainsaw | If you cannot predict your external IP, then you need NAT handling like STUN (or ICE to automatically set up STUN). |
| 06:59.26 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:59.45 | Chainsaw | ICE you would set up on the Asterisk side, if you cannot or will not do that, then you will need STUN manually configured at the client end. |
| 06:59.51 | stevePearPear | yup I just gave it a try after I’ve asked the question, SIPML5 does use ice to automatically set the IP |
| 07:00.03 | stevePearPear | Sent RTP packet to 202.166.x.xxx:45169 (via ICE) (type 00, seq 010567, ts 3217404536, len -000012) |
| 07:00.12 | Chainsaw | That'll help yeah. |
| 07:00.23 | stevePearPear | but just that there’s isn’t voice, could there be any chance the router or somehow is blockign it? |
| 07:00.39 | Chainsaw | Hard to say, you're not giving me much to go on. |
| 07:01.22 | stevePearPear | this is my environment: Asterisk at AWS with public IP. My house with SIPML5, my office with SIPML5 and a SIP provider end point configured at Asterisk. |
| 07:01.38 | stevePearPear | My house could use SIPML5 to call Asterisk to call out through the SIP provider (both way audio is fine) |
| 07:02.03 | stevePearPear | however my office’s SIPML5 is not working both way. office can hear the audio, but the other part couldn’t |
| 07:02.44 | stevePearPear | both have the message of sent RTP packet to 202.xxx.xxx.xxx ( VIA ICE) |
| 07:04.04 | stevePearPear | I could only think of my office network being the problem :( |
| 07:04.28 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 07:04.54 | Chainsaw | stevePearPear: If it's a "clever" router that thinks it understands SIP, turn that functionality off. |
| 07:05.44 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 07:06.25 | stevePearPear | what if I do not have access to my router? can I try to debug using wireshark on my end? |
| 07:08.44 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 07:09.24 | Chainsaw | stevePearPear: Yes, you can see whether addresses appear to get rewritten. |
| 07:09.42 | Chainsaw | stevePearPear: Particularly if you capture on your AWS instance and on your machine past the office router. |
| 07:10.02 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 07:10.03 | Chainsaw | would expect to see "sudden" changes in SIP invites |
| 07:13.44 | stevePearPear | what you are saying is that in the SIP invite message, i would see a change in the address cause of the router ‘cleverness’ which resulted in no audio ? |
| 07:14.14 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 07:22.35 | Chainsaw | stevePearPear: That is normally what it does, yes. |
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| 08:09.40 | Varazir | Hello, can I use a e-mail adress as username when I regitrate to a sip server? |
| 08:12.20 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.77.103) |
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| 08:35.45 | *** join/#asterisk Geek-Linux (~mubbashir@101.50.78.240) |
| 08:36.59 | Geek-Linux | Hi to All of you. can i use Dail application and only hangup the call to the called party and the play files to the calling party. ? Any help would be appriciated.. |
| 08:37.51 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:39.34 | TazzNZ | Geek-Linux: not sure I understand 100% |
| 08:39.44 | TazzNZ | you want to dial a number, and play files ? |
| 08:40.39 | TazzNZ | Varazir: yes |
| 08:43.15 | Varazir | TazzNZ: Okay, I tried on my Phone apps towards my ISPs SIp server but they translate the @ as %20 (or somthing like that) |
| 08:44.14 | TazzNZ | that will be an app issue |
| 08:44.18 | TazzNZ | not an asterisk issue |
| 08:45.08 | Varazir | TazzNZ: my quetsion was more general SIP question, but I'll try to conf a SIP trunk with my e-mail adress as username |
| 08:47.03 | Varazir | TazzNZ: data I have recived is: gatewayname = my e-mail adress, a proxy server adress and a password |
| 08:49.45 | Varazir | I'm just confused how the conf it all |
| 08:50.30 | Varazir | well I just need to keep reading wiki |
| 08:50.45 | Varazir | the wiki |
| 08:54.46 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:55.33 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 08:56.38 | jaflong | Hi, if I have such a string "Local/66@stream-00000013;2 Local/66@stream-00000013;1" How can "Local/66@stream-00000013" be extracted? |
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| 09:02.09 | Varazir | TazzNZ: So registration string would be mau@mailadress.com@password@proxysipadress.com ? |
| 09:02.40 | Varazir | TazzNZ: err my@mailadress.com:password@proxysipadress.com ? |
| 09:08.36 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbe89f.pool.mediaWays.net) |
| 09:09.13 | *** join/#asterisk stevePearPear (~stevePear@202.166.82.164) |
| 09:26.53 | Varazir | can someone just please point me to the documentation I need to to read to set up my SIP connection from my asterisk to my ISPs IP-phone system |
| 09:30.38 | TazzNZ | Varazir: yes - that looks correct |
| 09:31.18 | TazzNZ | Varazir: you need a register in sip.conf under the general section |
| 09:31.49 | TazzNZ | then you need a "peer" section in sip.conf, I normally make it the same as the username |
| 09:32.09 | TazzNZ | this will set stuff like the username and secret, along with what codec to use |
| 09:33.09 | Varazir | TazzNZ: Okay thanks, then I'm on the correct way |
| 09:33.23 | Geek-Linux | TazzNz: No i want to dial a number and set its duration to 60 secs after that the dialed number disconnects and the diler listen to the file. |
| 09:34.26 | TazzNZ | Geek-Linux: if you disconnect the call it will "die" and hangup on the user |
| 09:34.47 | TazzNZ | unless, what you want is |
| 09:35.14 | TazzNZ | I dial a number, via your asterisk box, I am connected for 60 seconds, then the remote side is hung up and you play me a file ? |
| 09:35.29 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 09:35.34 | TazzNZ | (this sounds like a card calling implementation ? ) |
| 09:40.14 | *** join/#asterisk iulhk (iulhk@182.189.50.126) |
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| 10:14.16 | Varazir | I get |
| 10:14.18 | Varazir | 11:32 < TazzNZ> this will set stuff like the username and secret, along with what codec to use |
| 10:14.21 | Varazir | ops |
| 10:14.54 | Varazir | TazzNZ: I get this [2014-06-09 11:05:56] NOTICE[7411] chan_sip.c: -- Registration for 'my@email.com' timed out, trying again (Attempt #2) |
| 10:30.29 | *** join/#asterisk TechAdam (c27dc475@gateway/web/freenode/ip.194.125.196.117) |
| 10:32.50 | TechAdam | hey guys, does anyone have a good website that teaches troubleshooting skills(specifically for conferencing but in general too!) Or is Asterisks wiki and voip-info the best there is? :L |
| 10:33.50 | *** join/#asterisk danjenkins (~dan@nat/digium/x-zecxkeimdwoagtrp) |
| 10:35.18 | wdoekes | ~voip-info |
| 10:35.19 | infobot | i guess voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 10:35.24 | wdoekes | hm |
| 10:35.34 | wdoekes | it should state something about it being very old and wrong at times |
| 10:35.50 | wdoekes | TechAdam: did you check the book? |
| 10:35.53 | wdoekes | ~newbook |
| 10:35.53 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 10:35.57 | wdoekes | ~thebook |
| 10:35.57 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:38.17 | *** join/#asterisk bhavikpatel6842 (~root@122.169.31.139) |
| 10:38.22 | bhavikpatel6842 | Hi Guys |
| 10:38.41 | bhavikpatel6842 | Can any one have latest Asterisk manager php script. |
| 10:39.01 | *** join/#asterisk fisted__ (~fisted@unaffiliated/fisted) |
| 10:39.04 | bhavikpatel6842 | I want to connect asterisk manager using latest php class file. |
| 10:39.24 | bhavikpatel6842 | Can any one give me suggestion how can I find in. ? |
| 10:39.46 | TechAdam | @wdoekes: I downloads Asterisk: Future of Telephony but was hoping for a broken down guide on a website |
| 10:40.08 | Geek-Linux | TazzNZ: yes exactly i want the same: |
| 10:48.03 | Varazir | I can't get the SIP trunk towards my IP telephone SIp server to Work :( |
| 10:59.46 | *** join/#asterisk Torenn (~Valinor@mimas.lightwitch.org) |
| 11:07.35 | Geek-Linux | TazzNZ: yes exactly i want the same: |
| 11:11.58 | Torenn | Hello I'm having a great deal of issues, using the XMPP module in component mode in Asterisk 12. Did experience issues on this regard? |
| 11:14.42 | *** join/#asterisk stevePearPear (~stevePear@202.166.82.164) |
| 11:16.57 | Torenn | ...[Jun 4 21:02:59] ERROR[27596]: res_xmpp.c:3832 xmpp_client_config_post_apply: Jabber identity 'asterisk.domain.tld' could not be created for client 'asterisk' - client not active ===> this is mainly the error the if statement above checks that the user member in the jid struct set by iks_id_new() isn't empty but that will be the case for components. And even patching it so the check won't be done if the XMPP_COMPONENT flag is set will have |
| 11:16.57 | Torenn | the component connect but not perform authentication after it opens the stream to the server. |
| 11:17.09 | *** part/#asterisk bhavikpatel6842 (~root@122.169.31.139) |
| 11:28.47 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:33.48 | *** join/#asterisk Ice_Strike (~Ice_Strik@cpc1-oldh7-0-0-cust772.10-1.cable.virginm.net) |
| 11:34.23 | Ice_Strike | How do I find out what the total active calls and also currenly rining calls on a sip trunk? |
| 11:35.02 | phpboy | Ice_Strike: start with core show channels |
| 11:35.15 | Ice_Strike | Yea I did core show channels |
| 11:35.31 | WIMPy | sip show inuse |
| 11:36.06 | Ice_Strike | Hmmm |
| 11:36.08 | Ice_Strike | I just see * User name In use Limit |
| 11:37.14 | WIMPy | You have no sip users? |
| 11:37.43 | WIMPy | I'm pretty sure you need call counters enabled. No idea if other users aren't shown. |
| 11:38.59 | phpboy | seks |
| 11:40.47 | Ice_Strike | I do have sip users |
| 11:40.55 | Ice_Strike | Will check call counters. |
| 11:43.04 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-sswdraiqdrxavjkh) |
| 11:43.04 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 11:51.30 | *** part/#asterisk joesmoe (~joesmoe@c-73-181-72-237.hsd1.co.comcast.net) |
| 11:51.48 | *** join/#asterisk beanie (~beanie@bgareth.plus.com) |
| 11:53.20 | TechAdam | is there anyway to view the sip call set up/tear down of a call thats already happened? i know it shows in realtime in asterisk cli but not sure how to find it from a few mins ago |
| 11:53.42 | beanie | hello - I have a problem whereby when I call a particular IVR internally (it has a short code associated with it for transferring calls) it cuts off half way through but it doesn't do it when the call is coming in internally, if the problem is obvious that's great but I suspect that you are going to want the debug information to move forward, i'm a little rusty on making sure I get the right parts though... |
| 11:54.01 | beanie | doesn't do it when the call is coming in externally** |
| 11:54.33 | beanie | TechAdam I think there is a more/less command that I'm not very good at... |
| 11:55.03 | beanie | I also believe that there is a file that keeps a running log although I may be wrong on this point |
| 11:55.48 | TechAdam | i was thinking a log file stored somewhere, im looking for the ack, bye 200 etc for a call that dropped |
| 11:57.45 | WIMPy | beanie: Is that IVR answering the call? |
| 11:58.07 | WIMPy | SIP debug is not written to any file. If you want that use the usual network debugging tools. |
| 11:58.21 | *** join/#asterisk gusto (~gusto@dial-5-178-50-224.orange.sk) |
| 11:59.32 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 11:59.35 | beanie | ah Tech Adam - it may be a case of capturing the output as soon as possible - are you actually copying from the terminal itself or, like me using Putty on Windows? |
| 11:59.45 | beanie | if so you need to go into the settings and change the amount it captures |
| 12:00.06 | beanie | WIMPy - thanks for picking up on my question, yes the IVR is answering the call when the outside world calls in |
| 12:00.10 | beanie | and it works fine |
| 12:00.29 | beanie | it's an ivr that routes to further ivr levels - each ivr level has its own shortcode for transferring calls |
| 12:01.12 | WIMPy | Well, I'm done guessing then. Give use some debug. |
| 12:03.21 | beanie | ok, what way do you prefer it - I use putty and I struggle to know which bits you need without overloading you and whilst making sure everything thats needed is included :) |
| 12:03.53 | WIMPy | core set verbose 3 |
| 12:04.26 | beanie | sweet :) |
| 12:04.32 | beanie | and what website do you prefer for the output? |
| 12:05.02 | WIMPy | Whichever works. |
| 12:05.19 | beanie | fair play :) brb |
| 12:05.40 | TechAdam | im using putty and freepbx is installed on top of asterisk, im new to voip and my new job is training me in, i want to solve a problem without asking my co workers for help though :L im checking through the full.log file now so many lines :o |
| 12:06.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:07.00 | WIMPy | TechAdam: You know that you're in the wrong channel since installing FreePBX? |
| 12:07.54 | TechAdam | i figured cause i was looking at asterisk logs they would be stored in the same place |
| 12:07.58 | TechAdam | im using cli for this |
| 12:08.19 | TechAdam | the freepbx reports dont deserver to be called reports :/ |
| 12:08.33 | beanie | WIMPy - http://pastebin.com/5mRcwDAZ |
| 12:09.31 | WIMPy | 'd rather like to find out if FreePBX deserves to be called PBX. |
| 12:09.48 | WIMPy | Well, if I ever have too much time.... |
| 12:11.46 | WIMPy | beanie: I see a rtp timeout. |
| 12:12.03 | WIMPy | That means Asterisk killed the call because the remote end seemed to have gone. |
| 12:12.16 | beanie | yeah i noticed this yesterday - it made me hang my head in despair - rtp packets have made my life miserable |
| 12:12.41 | WIMPy | I think that's what they are there for. |
| 12:12.53 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:13.00 | beanie | well, im still connected at this point and hearing it, it's an ivr reading out a recording so it doesn't require my input so i'm a bit lost |
| 12:13.12 | beanie | i just then get cut off |
| 12:13.53 | WIMPy | I'm not sure I get the full call. There was other stuff in there. |
| 12:14.14 | WIMPy | Switch off the sip debug next time. |
| 12:14.22 | beanie | ah i set the debug on before i made the test call for the purposes of the log |
| 12:14.24 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 12:14.31 | beanie | oh I did switch it off - did I switch it off too early? |
| 12:14.48 | WIMPy | So the question remains: What does your dialplan do? Does it Answer() the call or use Playback? |
| 12:15.08 | [TK]D-Fender | We don't see the call come in or any of the processing up to the supposed bridge at all. |
| 12:15.21 | beanie | I don't know how to answer that other than to tell you how I set it up... |
| 12:15.51 | beanie | the IVR is set to divert to other IVR levels further down |
| 12:16.34 | beanie | depending on the option pressed (1-9) on the main ivr it will then route the call through to the corresponding IVR and eventually the IVR routes to respective call queues |
| 12:17.09 | beanie | TK D-Fender - i've set the verbose to 3 |
| 12:17.12 | [TK]D-Fender | beanie: Descriptions aren't going to get you much. You should start with showing a complete call... |
| 12:17.15 | WIMPy | It does not help to know what could happen if you got past your issue. |
| 12:17.33 | WIMPy | Give us the complete call. |
| 12:17.49 | beanie | ok - sure - It worked before though, just not sure why it has stopped working |
| 12:18.02 | beanie | tried restarting everything |
| 12:18.02 | WIMPy | Before what? |
| 12:18.15 | beanie | before I come to do a test just to see if everything was ok |
| 12:18.23 | beanie | i had done a test previously and it was fine |
| 12:18.28 | beanie | ok - let's get a complete call |
| 12:18.39 | beanie | what do I need to do differently to the test before? |
| 12:18.48 | beanie | to get what we are looking for |
| 12:19.49 | [TK]D-Fender | get the complete call. |
| 12:20.22 | beanie | yes, but how - last time I did sip set debug on, then I set core set verbose 3 - ran the call and then turned sip set debug off |
| 12:20.31 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.76.92.218) |
| 12:20.33 | beanie | i can run it without verbose |
| 12:20.59 | Greek-Boy | Would it anyone be so kind to give me a working example of Polycom config files for distinctive ringtones? |
| 12:21.15 | *** join/#asterisk KOPRajs (52638852@gateway/web/freenode/ip.82.99.136.82) |
| 12:22.03 | TechAdam | what is "endmixmoncheck" mean in the log files? |
| 12:22.33 | [TK]D-Fender | [08:20]beaniei can run it without verbose <- You should not be trying to show us LESS.... |
| 12:22.39 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 12:23.07 | [TK]D-Fender | TechAdam: Perhaps you should show us the line before asking what it means. |
| 12:23.13 | KOPRajs | hi, what is the correct syntax for (if first 3 letters from caller id match xyz?true:false) please? |
| 12:24.02 | TechAdam | ahh i thinks its mix monitor |
| 12:24.23 | TechAdam | [Jun 9 12:30:48] VERBOSE[22201] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/222-00000194", "1?endmixmoncheck") in new stack |
| 12:24.35 | TechAdam | [Jun 9 12:30:48] VERBOSE[22201] pbx.c: -- Executing [s@macro-hangupcall:9] NoOp("SIP/222-00000194", "End of MIXMON check") in new stack |
| 12:25.00 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 12:25.03 | TechAdam | sorry im semi-talking aloud, trying to get my head around this ^.^ |
| 12:25.30 | [TK]D-Fender | TechAdam: that is a dialplan label which is specific to freepbx and tells us nothing |
| 12:25.55 | KOPRajs | something like IF(${CALLERID(number):0:3} = "xyz"?true:false) |
| 12:26.23 | TechAdam | im guessing its this guy: http://www.voip-info.org/wiki/view/MixMonitor |
| 12:27.12 | [TK]D-Fender | TechAdam: No... that is some dialplan logic around a part of your call meant to decide about using it. |
| 12:27.19 | beanie | ok people - http://pastebin.com/6xe3a6UR |
| 12:27.50 | [TK]D-Fender | TechAdam: And that is in the cleanup phase of a call anyway |
| 12:28.43 | [TK]D-Fender | beanie: Why have you turned off verbose? |
| 12:30.24 | WIMPy | TechAdam: Ask in #freepbx |
| 12:30.29 | WIMPy | ~freepbx |
| 12:30.29 | 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 |
| 12:30.47 | beanie | TK-Defender - [TK]D-Fender> [08:20] beanie i can run it without verbose <- You should not be trying to show us LESS.... |
| 12:30.54 | TechAdam | yes masters! :) |
| 12:31.23 | [TK]D-Fender | beanie: beaYou ARE now shoing us less. We had verbose before... |
| 12:31.30 | beanie | Do you want me to do it with verbose set to three? |
| 12:31.36 | [TK]D-Fender | beanie: And now we DON'T. |
| 12:31.53 | [TK]D-Fender | beanie: 10 <- |
| 12:31.58 | *** join/#asterisk u0m3_ (~u0m3@92.80.73.182) |
| 12:32.22 | beanie | ok so will core set verbose 10 suffice |
| 12:34.59 | beanie | so are you saying set it to 10 or less |
| 12:35.07 | beanie | not sure about the <- bit |
| 12:35.31 | WIMPy | Setting it to 3 or 300 probably makes no difference. |
| 12:35.55 | beanie | ok - just trying to get this right by D-Fender's request :) |
| 12:38.30 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 12:40.34 | beanie | ok - this is better I now realise - http://pastebin.com/7bRdmAD3 |
| 12:40.55 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 12:41.38 | [TK]D-Fender | <--- SIP read from UDP:188.165.221.151:5102 ---> REGISTER sip:192.168.1.104:5060 SIP/2.0 To: "600" <sip:600@80.229.154.247:5060> |
| 12:41.55 | [TK]D-Fender | I'm seeing local IP reads for things targeting a WAN IP all over the place |
| 12:42.07 | [TK]D-Fender | This reeks of hairpin-NAT issues |
| 12:42.40 | [TK]D-Fender | <--- SIP read from UDP:80.229.154.247:61483 ---> INVITE sip:7772@192.168.1.104:5060 SIP/2.0 |
| 12:42.59 | [TK]D-Fender | And this actual call we're looking at looks like it comes from the server's IP itself |
| 12:43.12 | [TK]D-Fender | As though it were looping back in (hence hairpin) |
| 12:43.12 | beanie | yeah thats 80.... is the static ip here |
| 12:43.17 | [TK]D-Fender | this is a recipe for disaster |
| 12:43.27 | WIMPy | That's a big pile of ... stuff. Was that hand written? |
| 12:43.51 | beanie | ah right - my softphone is currently logged in through the external ip even though im currently on the home network - could that be the issue? |
| 12:44.14 | [TK]D-Fender | YES |
| 12:44.27 | beanie | ok you get to slate me now :-p |
| 12:44.35 | beanie | i feel like a prat |
| 12:46.50 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 12:46.56 | *** join/#asterisk theron_ (~theron@66.220.145.150) |
| 12:47.29 | beanie | oh im ever so sorry - issue seemingly sorted - so with hairpin nat - is it always about that or in a nutshell what other issues could there be causing hairpin nat |
| 12:48.09 | [TK]D-Fender | There isn't an issue causing hairpin NAT. |
| 12:48.16 | [TK]D-Fender | Hairpin NAT *IS* the issue |
| 12:48.24 | [TK]D-Fender | You are phoning your cell phone... FROM you cell phone. |
| 12:48.51 | [TK]D-Fender | Don't point to your public IP while behind the router that it lands on. |
| 12:51.08 | beanie | thanks point learned :-D |
| 12:51.18 | beanie | thank you D-Fender and thank you WIMPy :) |
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| 13:07.20 | Katty | hello my asterisk does not work at all how to fix pls thx |
| 13:09.20 | *** join/#asterisk ghoti_ (~paul@scratch.it.ca) |
| 13:10.13 | wdoekes | Katty: ask getafisk to make him some potion |
| 13:12.43 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 13:16.04 | *** join/#asterisk ghoti_ (~paul@scratch.it.ca) |
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| 13:23.15 | leifmadsen | Katty: step 1) get really really drank |
| 13:23.23 | leifmadsen | Katty: step 3) woooo party! |
| 13:24.55 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 13:27.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hxnmjlvyzetcoqsa) |
| 13:27.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:34.10 | Katty | woo, party. |
| 13:36.47 | [TK]D-Fender | Step 5) find out what happened during step 2. |
| 13:36.58 | [TK]D-Fender | Step 6) Deny. Deny. Deny. |
| 13:38.21 | jameswf | leifmadsen: she didn't ask the agenda for Astricon |
| 13:40.43 | Katty | i don't /need/ the agenda for astricon ^_^ |
| 13:41.03 | Katty | i bring the party with me! |
| 13:41.50 | jameswf | what happens at *con stays at *con... except herpies |
| 13:42.20 | Katty | rolls eyes |
| 13:43.44 | jameswf | technically not even vegas psssshhhhh |
| 13:44.42 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:52.25 | beanie | I have a problem whereby an extension user behind their own router is able to login and call me but she can hear me and I cannot hear her - the output is here http://pastebin.com/GSx1h4aw |
| 13:54.45 | beanie | ok i have posted way too much from there - the call is from kirsty fowler |
| 13:54.55 | beanie | sorry about that - should be easier to find with her name as they keyword |
| 13:54.58 | beanie | the* |
| 13:56.09 | [TK]D-Fender | <--- SIP read from UDP:2.217.171.128:7220 ---> INVITE sip:048665@192.168.1.104:5060 SIP/2.0 <-- why would we be seeing a local IP on this? |
| 13:57.08 | beanie | that'll be the old stuff - if you "find" "Kirsty Fowler" that's where the fun starts - i've already logged in to the internal ip as i'm behind my router - she's logged in to my static ip from behind her router |
| 13:57.46 | [TK]D-Fender | 874 <----- |
| 13:57.48 | [TK]D-Fender | that is her |
| 13:57.53 | [TK]D-Fender | look at the packet |
| 14:01.32 | beanie | her ip ends 128 |
| 14:01.38 | beanie | i'm not sure where i'm looking |
| 14:01.44 | beanie | for the packet |
| 14:01.58 | beanie | are you hinting at it being the nat settings on the extension |
| 14:02.00 | [TK]D-Fender | [09:57][TK]D-Fender874 <----- |
| 14:02.47 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
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| 14:07.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:13.07 | *** join/#asterisk gusto (~gusto@178.143.108.176) |
| 14:13.18 | gusto | hi |
| 14:13.50 | gusto | does anyone here have an idea if one can adjust the ATAs in a way that they distribute their time to a DECT phone? |
| 14:14.08 | gusto | because everytime i loose my time on the phone i have to manually put it in |
| 14:14.20 | [TK]D-Fender | You should probably read it's manual. |
| 14:14.33 | [TK]D-Fender | And never assume that one model will work exactly like another. |
| 14:14.33 | *** join/#asterisk TechAdam (c27dc475@gateway/web/freenode/ip.194.125.196.117) |
| 14:15.37 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 14:16.33 | gusto | yes, there is nothing in there about time and date setting in the manual |
| 14:16.46 | gusto | they just say that when it does not already have its time, i can put some in |
| 14:18.18 | sgriepentrog | gusto: usually the presence of time in CID is either coded or not coded into the ATA - in my experience some models send time always, others never do. Modifying the code of course is not something you can expect to do, unless you build your own adapter with Asterisk and an FXO port. |
| 14:18.43 | gusto | lol |
| 14:18.44 | gusto | ok |
| 14:19.05 | gusto | so you mean that it sets the time when a call comes in? |
| 14:19.59 | sgriepentrog | The CID signal (sent between the first and second ring) MAY contain the time, and nearly always the phone will store that. |
| 14:20.39 | gusto | ok |
| 14:20.49 | sgriepentrog | Note I'm talking bellcore north american caller id, other country formats may not have time. |
| 14:20.50 | gusto | that is good to know, so i have to make a call to test it |
| 14:20.54 | sgriepentrog | Yup. |
| 14:20.57 | gusto | well, i can try them out |
| 14:22.47 | gusto | heh, it works |
| 14:22.49 | gusto | lol |
| 14:22.58 | gusto | it did set the time ... after the second ring |
| 14:23.08 | gusto | exactly like you have predicted |
| 14:23.14 | sgriepentrog | Yup. That's when it decodes what it received. |
| 14:23.31 | gusto | so from now on i have to call myself first to set the time when my DECT phone goes out of power |
| 14:29.49 | *** join/#asterisk serafie (~erin@nat/digium/x-lmbblfgfztydlbgp) |
| 14:32.42 | beanie | TK Defender - I've been looking through, presumably you were referring to where to read from on the line numbers - I really cannot see the issue being documented - the only thing i can see is "SIP/2.0 401 Unauthorized" |
| 14:33.05 | beanie | and since she's logged in, im not sure why it's coming back with that error |
| 14:38.14 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 14:38.49 | SuperNull | anyone using snmp for channel activity monitoring ? i used a template for Cacti and i think they must have an old mib its showing MGCP calls yet we only use SIP lol |
| 14:45.44 | mjordan | have you tried the MIB definition on the wiki? https://wiki.asterisk.org/wiki/display/AST/Asterisk+MIB+Definitions |
| 14:50.45 | SuperNull | yeahh looks like this thing template was designed to not explicitly get the channel type name .. so if compiled without mgcp the name order is off. |
| 14:59.09 | *** join/#asterisk KOPRajs (52638852@gateway/web/freenode/ip.82.99.136.82) |
| 15:00.08 | KOPRajs | hi, I've got expression like this $[${CALLERID(num)}:0:3=ab-] |
| 15:00.43 | KOPRajs | how do I tell Asterisk that "ab-" is a string and not "ab" and minus operator? |
| 15:01.30 | [TK]D-Fender | put quotes around both sides |
| 15:01.41 | KOPRajs | do I simply use doublequotes? |
| 15:01.52 | [TK]D-Fender | yes |
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| 15:06.45 | *** join/#asterisk toothe (~mongolian@unaffiliated/toothe) |
| 15:06.49 | toothe | thanks all for the hlep last friday |
| 15:06.53 | toothe | i was pretty frustrated |
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| 15:18.06 | beanie | TK D-Fender, I'm thinking of dropping the whitelist solution on the server in favour of Fail2ban - is Fail2ban on its own useless or is it accepted as commonplace security? |
| 15:21.17 | beanie | question opened to anyone :) |
| 15:21.45 | tm1000 | beanie: Its fine but you shouldn’t rely on it 100%. It’s not foolproof |
| 15:22.03 | beanie | what other things should i rely on |
| 15:22.13 | beanie | i cant get access to putty which is doing my head in when i am away from home |
| 15:26.19 | tm1000 | beanie: well you could change the port of ssh for starters |
| 15:26.26 | tm1000 | lock down every port not required by asterisk |
| 15:26.35 | tm1000 | setup permit deny lists in asterisk |
| 15:26.51 | tm1000 | user strong passwords in asterisk |
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| 15:37.34 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:39.14 | cusco | hi folks |
| 15:39.36 | cusco | what is the provider parameter in manager action PresenceState |
| 15:41.31 | cusco | ow its the hint, nvm |
| 15:41.47 | cusco | aw no its not |
| 15:41.49 | cusco | er.. |
| 15:43.04 | *** join/#asterisk danjenkins_ (~dan@nat/digium/x-tkprpuqxhrcvwxkg) |
| 15:43.27 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 15:43.40 | Katty | PIZZA ROLLS. |
| 15:46.13 | stasiu7 | Hi. Is there a way to insert cdr record as soon as the call starts and the update it when it ends? |
| 15:47.29 | Faustov | stasiu7: you can perform db operations from the dialplan if you want to |
| 15:50.20 | stasiu7 | ok. do you know if there's a table with all currently connected calls? I want to check if there's a call already being made to given phone number, without creating lock files |
| 15:50.57 | cusco | I guess I would use ami to show channels |
| 15:52.30 | beanie | @tm1000 - what are permit deny lists? |
| 15:53.55 | Faustov | stasiu7: probably in the return info from "core show channels"? |
| 15:59.31 | Katty | FLUFFY BUNNIES |
| 16:02.17 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:05.04 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
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| 16:14.00 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:16.44 | *** join/#asterisk SpaceInvaders (~SpaceInva@adsl-74-235-60-29.clt.bellsouth.net) |
| 16:19.31 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:19.52 | stevePearPear | hi |
| 16:20.11 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:20.49 | stevePearPear | i only have one way audio, my Asterisk is via AWS. I noticed that the SDP c= IN IP4 xxx.xxx.xxx.xxx |
| 16:21.08 | stevePearPear | where xxx is my private LAN IP instead of externIP, is that probably the reason for 1 way audio? |
| 16:21.21 | stevePearPear | I’ve already set my externIP in sip.conf [general] section too :( |
| 16:21.38 | stevePearPear | externip=yyy.yy.yyy.yyyy |
| 16:21.52 | stevePearPear | any reason why Asterisk is still using its internal ip? |
| 16:22.28 | [TK]D-Fender | Not having specified nat=, not having defined your localnet rages.... |
| 16:23.00 | stevePearPear | my peers have nat=force_rport,comedia |
| 16:23.05 | stevePearPear | hmm localnet is something i dun have |
| 16:25.12 | [TK]D-Fender | You have to tell * what's local and what isn't |
| 16:26.43 | toothe | Do SBC's primarily speak SIP? or their own custom protocol? |
| 16:26.46 | toothe | or multiple ? |
| 16:28.02 | [TK]D-Fender | SBC is a CONCEPT |
| 16:28.16 | [TK]D-Fender | what protocols the one you're looking at use... is up to that product itself |
| 16:28.44 | [TK]D-Fender | SIP SBC's... speak SIP |
| 16:28.45 | stevePearPear | [TK]D-Fender, in this case the value inside localnet should be my local address range? |
| 16:28.51 | [TK]D-Fender | yes |
| 16:29.11 | stevePearPear | thanks, i’ll give it a try |
| 16:33.13 | stevePearPear | oh yes, my c=IN became my public ip now |
| 16:33.15 | stevePearPear | thanks :) |
| 16:34.29 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 16:38.19 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-zqhnsrarozhuyivx) |
| 16:43.22 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
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| 16:58.17 | *** join/#asterisk ctaloi (~textual@66.218.0.5) |
| 17:01.20 | cusco | I was asking the other day, and I'm asking again: a call placed via originate or a call file, I can Set(CDR(custom)=foo) on the channel (its local) but not on the Exten that answers it ... |
| 17:03.50 | cusco | here is my example: |
| 17:03.51 | cusco | http://paste.debian.net/hidden/732d75df/ |
| 17:04.22 | cusco | is there something I can do to set the custom cdr var again when it comes to the second channel leg ? |
| 17:04.57 | *** join/#asterisk Ichabod (~Arty@unaffiliated/ichabod) |
| 17:05.51 | Ichabod | Hello, I'm trying to configure Asterisk to work with an LDAP database, while I can login using a softphone (Telephone on OS X), I cannot make calls to another ldap user, as I get 503 errors from Asterisk |
| 17:06.23 | Ichabod | besides this obvious issue, sip show peers and sip show users returns nothing, even if I have 2 accounts logged in on the asterisk server |
| 17:06.45 | [TK]D-Fender | Show us |
| 17:06.46 | [TK]D-Fender | ~pb |
| 17:06.47 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:06.47 | *** join/#asterisk litn (~blice@alrig.ht) |
| 17:06.48 | [TK]D-Fender | ^^^^ |
| 17:06.58 | litn | is asterisk-biz the right mailing list to e-mail if I need a consultant? |
| 17:07.25 | file | unless you are using caching in chan_sip the users/peers will not show up if they are stored in realtime |
| 17:07.36 | newtonr | litn, yeah |
| 17:07.58 | Ichabod | http://paste.debian.net/104155/ |
| 17:08.21 | Ichabod | this is the entire asterisk debug output after I quit the SIP application and restarted |
| 17:08.46 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:08.55 | Ichabod | http://paste.debian.net/104156/ |
| 17:09.03 | Ichabod | this is the output of sip show peers/users |
| 17:09.25 | Ichabod | file: ah, that's something at least |
| 17:09.45 | file | and also in that it says it can't modify the LDAP entry... |
| 17:10.33 | Ichabod | yea, but that appears to not be a major issue |
| 17:10.39 | Ichabod | (as far as I found resources online) |
| 17:11.10 | *** join/#asterisk ctaloi_ (uid34941@gateway/web/irccloud.com/x-oirknidbpprtnckl) |
| 17:11.11 | file | ...it's probably updating to put the contact information in from the registration, if that fails then you can't call it |
| 17:11.47 | Ichabod | right |
| 17:12.33 | Ichabod | any idea which attribute type that would need to be defined? |
| 17:12.54 | file | not a clue, never touched it |
| 17:16.22 | Katty | OR DID YOU |
| 17:16.31 | Katty | dusts for file's prints |
| 17:18.35 | file | hi |
| 17:19.56 | Ichabod | http://paste.debian.net/104160/ # This is the resulting log after a (failed) call |
| 17:20.28 | *** join/#asterisk troyt (~troyt@2601:7:6200:1362:44dd:acff:fe85:9c8e) |
| 17:21.16 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 17:22.07 | ChkDigit | Is there any way to turn off jb warnings on the console? |
| 17:23.15 | *** join/#asterisk jpoz (~jpoz@207.173.72.195) |
| 17:25.04 | *** join/#asterisk jpoz_ (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 17:25.56 | Ichabod | upped verbosity of the CLI |
| 17:26.00 | Ichabod | now I get: -- Executing [herpderptest@bogon-calls:1] Congestion("SIP/tstrickx-00000004", "") in new stack |
| 17:26.33 | Ichabod | so it's claiming congestion, while nothing else is using it :s |
| 17:29.03 | file | because the call went into "bogon-calls" and that's what the dialplan has it doing |
| 17:29.16 | file | probably because LDAP isn't working right so it's not getting the correct information |
| 17:30.03 | Ichabod | aight, looks like I have to relook my ldap stuff as you said :) |
| 17:47.12 | zopsi | is pjsip fully implemented? meaning I can remove chan_sip and rely purely on chan_pjsip. |
| 17:49.14 | litn | newtonr: do posts to the mailing list require approval before going up? |
| 17:49.35 | newtonr | litn, asterisk-biz ? no, but you must be subscribed |
| 17:50.04 | litn | oh! |
| 17:50.40 | *** join/#asterisk navaismo (~navaismo@189.191.243.137) |
| 17:52.01 | newtonr | litn, http://asterisk.org/community/discuss |
| 17:56.26 | zopsi | just for clarification. How would I make this extensions.lua work with inbound DIDS http://pastebin.com/CsQTSyFx? |
| 17:59.13 | file | zopsi, if PJSIP has all the features you need... yes |
| 18:00.08 | zopsi | file: I can't seem to get my pjsip dialplan working with inbound from Flowroute. Works fine with outbound. I'm fairly confident it is a simple mistake. |
| 18:00.55 | *** join/#asterisk bmurt (~brendan@static-71-242-238-15.phlapa.east.verizon.net) |
| 18:00.57 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 18:04.38 | zopsi | Is PJSIP compatible with DPMA and digium phones? |
| 18:05.12 | Qwell | malcolmd: ^^? I was wondering that myself the other day |
| 18:06.19 | mjordan | zopsi: yes - https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration has the information. You'll need version 2.0.0 from the Digium website, and Asterisk 12.3.0 |
| 18:06.56 | mjordan | zopsi: examples of the PJSIP configuration (pjsip.conf) is here - https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phones+when+used+with+the+DPMA |
| 18:08.17 | ChannelZ | Are you getting an AOR error or what with your incoming? |
| 18:11.39 | zopsi | <PROTECTED> |
| 18:11.51 | zopsi | ChannelZ: I'll post some logs. |
| 18:11.52 | mjordan | daring! |
| 18:14.05 | SpaceInvaders | Hey, if I have apache installed should enable=no in http.conf? |
| 18:14.15 | SpaceInvaders | the asterisk/http.conf |
| 18:14.43 | SpaceInvaders | or does asterisk somehow know not to use the built-in if apache is running? |
| 18:14.56 | SpaceInvaders | FYI I'm on a Fedora 20 system |
| 18:15.11 | mjordan | Asterisk has no knowledge of apache |
| 18:15.16 | [TK]D-Fender | SpaceInvaders: What are you hoping to use http.conf for? |
| 18:15.41 | SpaceInvaders | I was following the instructions I found at http://www.savelono.com/linux/how-to-set-up-the-asterisk-20-gui-with-asterisk-16.html |
| 18:15.42 | toothe | [TK]D-Fender: Thanks again for your help...genuinely. |
| 18:15.49 | SpaceInvaders | I was hoping to see the GUI it mentioned there. |
| 18:16.00 | [TK]D-Fender | SpaceInvaders: No.. it has been dead for over 3 years now |
| 18:16.41 | SpaceInvaders | was it replaced by the asterisk-gui I saw when I yum searched on asterisk? |
| 18:16.59 | [TK]D-Fender | no, that is the only QUI that uses it |
| 18:17.01 | [TK]D-Fender | and it is dead |
| 18:17.04 | [TK]D-Fender | forget about it |
| 18:17.12 | SpaceInvaders | got it. what's the asterisk-gui? |
| 18:17.44 | SpaceInvaders | asterisk-gui.noarch : Graphical interface for Asterisk configuration |
| 18:17.51 | [TK]D-Fender | DEAD |
| 18:18.00 | SpaceInvaders | ok both are dead |
| 18:18.06 | [TK]D-Fender | there is no "other" |
| 18:18.12 | SpaceInvaders | they were both the same thing? |
| 18:18.20 | [TK]D-Fender | the only thing to ever use it was that singular GUI, and it's dead |
| 18:18.36 | zopsi | here's the part of my log http://pastebin.com/JdrVNdxS |
| 18:18.42 | SpaceInvaders | ok thanks! |
| 18:30.38 | ChannelZ | uhm well that looks like a trainwreck. |
| 18:34.16 | zopsi | ChannelZ: yeah. |
| 18:34.42 | zopsi | ChannelZ: Outbound works fine. Inbound goes to the failover route (through SIP TRUNK not asterisk) |
| 18:35.17 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 18:36.54 | *** part/#asterisk bkruse (~Adium@64.89.97.127) |
| 18:38.23 | cusco | [TK]D-Fender: perhaps I could dare you to take a look at my question? |
| 18:40.38 | ChannelZ | I triple-dog-dare you |
| 18:50.53 | mjordan | zopsi: yeah, if you're getting that many config errors, something has gone horribly wrong :-) |
| 18:50.57 | mjordan | zopsi: are you trying to use realtime? |
| 18:51.43 | mjordan | and are you explicitly loading modules via modules.conf? |
| 18:51.51 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:55.33 | cusco | :/ |
| 18:55.46 | ipengineer | In asterisk 12 did the delete option in voicemail.conf get changed to deletevoicemail? That is what I am seeing in my realtime db but it does not appear to be working correctly |
| 18:55.50 | cusco | http://paste.debian.net/hidden/732d75df/ this is my dialplan and test call file |
| 18:56.18 | cusco | cdr custom var doesn't get set to bar |
| 18:59.32 | zopsi | mjordan: I don't want to use realtime, but asterisk does. |
| 19:00.30 | mjordan | ipengineer: voicemail wasn't touched |
| 19:00.44 | mjordan | zopsi: I'm not thinking that's what's happening here... |
| 19:01.03 | mjordan | zopsi: are you explicitly loading modules in modules.conf? What is your pjsip.conf? |
| 19:01.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:01.23 | zopsi | mjordan: no I have autoload on with noloads |
| 19:01.29 | zopsi | I'll paste those |
| 19:01.52 | ipengineer | mjordan: Alright Thanks. I didnt think so.. Not sure where that schema came from. |
| 19:04.21 | zopsi | mjordan: http://pastebin.com/8M0XweMT thats my modules, and here is my pjsip.conf http://pastebin.com/9GbnkdCr |
| 19:04.35 | zopsi | I'm using bare minimum to see if I can get things to work before throwing everything in. |
| 19:06.43 | zopsi | sorry about that pjsip.conf was pasted multiple times. http://pastebin.com/9GbnkdCr |
| 19:08.15 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 19:12.11 | zopsi | any help would be greatly appreciated. I'm still trying to get the hang of pjsip and lua extensions. I do like the lua extensions just don't know if I am doing it 100% correct. |
| 19:13.02 | file | do be aware the lua stuff is extended support and basically nobody uses it |
| 19:13.31 | zopsi | file: yeah I just wanted to give it a whirl. It still doesn't work with standard extensions.conf dialplans. |
| 19:13.38 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 19:17.40 | ChannelZ | I'm not sure an identify record makes any sense with no 'match'? |
| 19:18.30 | zopsi | ChannelZ: I just commented it. It doesn't change anything in terms of functionality for me. |
| 19:19.02 | zopsi | the client_uri I was a little confused about. |
| 19:19.55 | *** join/#asterisk huatou (~huatou@107.170.182.241) |
| 19:22.49 | file | I know what would cause that... but I don't know why... |
| 19:23.57 | zopsi | hm |
| 19:24.05 | ChannelZ | well I'm not sure what the deal is with all those errors, I listerally just pasted your config into mine and it loads |
| 19:24.12 | zopsi | hrm |
| 19:24.20 | *** part/#asterisk Ichabod (~Arty@unaffiliated/ichabod) |
| 19:24.32 | zopsi | I get a lot of NOTICE logs saying No matching endpoint found. |
| 19:25.06 | zopsi | I think it isn't responding to the SIP Trunks pings so it automatically routes to backup. Does that make sense? ChannelZ |
| 19:25.24 | ChannelZ | not to me no |
| 19:25.35 | file | it might be module load order with DPMA involved |
| 19:25.45 | zopsi | well I have it set to failover to my cell phone. |
| 19:25.46 | ChannelZ | yeah it has to be |
| 19:25.48 | file | although I still don't know why it would do this... |
| 19:26.05 | *** join/#asterisk voxter (~voxter@irc.voxter.net) |
| 19:26.07 | file | full console output with high verbose would show the order of things being loaded |
| 19:26.17 | zopsi | I have that if you want it |
| 19:26.31 | file | pastebin it, sure |
| 19:26.47 | ChannelZ | oh cool I just wiped out my own config. FML |
| 19:28.06 | voxter | hey all, I've got an asterisk 11.5 box that is not seeing all DTMF coming in via a SIP Trunk (ulaw/rfc2833) - looking at the pcap using wireshark, and analyzing the call, it sees all the correct digits being received on the interface, yet asterisk is disagreeing. Any tips/ |
| 19:28.45 | WIMPy | Ha, the master of pjsip is here. So let me ask that old question again: Is it possible to the th to domain (for invites) to something different than the hostname in chan_pjsip? |
| 19:28.58 | WIMPy | That has been a PITA with chan_sip so far. |
| 19:29.38 | WIMPy | s/the th/set the/ |
| 19:30.05 | file | WIMPy, I don't know what that means. |
| 19:30.08 | file | domain where |
| 19:30.37 | WIMPy | In the INVITE messages when placing a call to a peer. |
| 19:30.47 | file | there's numerous URIs in the INVITE message |
| 19:31.04 | WIMPy | To: |
| 19:31.33 | WIMPy | Actually the same issue also applies to registering. |
| 19:33.10 | zopsi | heres the full log guys http://pastebin.com/NfssK4nW |
| 19:33.40 | file | WIMPy, the request URI is used as the to URI currently |
| 19:33.44 | file | so, yes |
| 19:33.47 | malcolmd | Qwell: yup, as mjordan said. managed to get dpma for pjsip & asterisk 12 out last week |
| 19:33.53 | file | and you can also specify it for registering |
| 19:34.12 | WIMPy | Sounds good. |
| 19:35.31 | zopsi | sorry ChannelZ. Do you have a backup? |
| 19:35.42 | file | zopsi, well that looks better... |
| 19:36.06 | file | zopsi, although your transport-udp-nat transport can't be started |
| 19:36.14 | zopsi | yeah I saw that. |
| 19:37.23 | zopsi | Address '216.115.69.144:5060' provided on ip endpoint identifier 'flowroute' did not resolve to any address |
| 19:37.36 | file | try removing the port |
| 19:37.59 | zopsi | on just endpoint or identify as well? |
| 19:38.04 | file | identify |
| 19:38.17 | zopsi | sorry meant aor |
| 19:38.36 | zopsi | aor I also have the port |
| 19:38.45 | file | zopsi, ah yeah... you can't have two transports binding to the same thing like you do |
| 19:38.54 | file | port is fine there |
| 19:39.18 | zopsi | so delete transport-udp-nat? |
| 19:39.37 | file | or delete the other one, if you want to actually have it use your NAT settings |
| 19:40.04 | zopsi | file: the client_uri I think is wrong. Mind checking that real quick? |
| 19:40.07 | ChannelZ | zopsi: an old one |
| 19:40.30 | ChannelZ | oh wait.. I think I rsync to a server at work every few days.. |
| 19:41.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:42.04 | TazzNZ | voxter: is your box a VM or physical ? |
| 19:42.49 | voxter | TazzNZ: VM, but also it (seems) to only be happening to people coming from a particular carrier, everyone else is fine. what i mean is its not consistently failing for everyone. |
| 19:43.02 | file | zopsi, just set both server_uri and client_uri to what you have commented out for client_uri and see what that does... |
| 19:43.26 | zopsi | file: with the did included? |
| 19:43.30 | file | yes. |
| 19:43.42 | file | WELL |
| 19:43.47 | file | or whatever flowroute wants you to authenticate as |
| 19:44.58 | file | I don't think they want you to authenticate using your DID, so if you try... won't work |
| 19:46.07 | zopsi | OK now it goes to my software voip client correctly, but I get "Offer contianed no valid media descriptions" when I answer. Is this a client issue or asterisk issue? |
| 19:46.45 | zopsi | <PROTECTED> |
| 19:46.56 | file | pastebin with "pjsip set logger on" enabled |
| 19:47.32 | marceloamorim | guys, anyone install asterisk over debian lately? |
| 19:48.16 | Chainsaw | marceloamorim: You make it sound like some sort of a fancy alcoholic drink. |
| 19:50.15 | zopsi | file: I sent it in a PM as it contains some unsanitized informationl. |
| 19:50.56 | file | resend |
| 19:52.15 | zopsi | sent |
| 19:52.20 | file | what is your transport? there's a private IP in the traffic to flowroute |
| 19:52.39 | zopsi | I'm using transport-udp right now. |
| 19:53.10 | file | then it's likely Flowroute will be broken for calling as the NAT settings aren't there |
| 19:53.30 | zopsi | Okay so I'll switch to transport-udp-nat that has my settings |
| 19:55.23 | zopsi | file: same issue. maybe my NAT transport is wrong. |
| 19:55.33 | marceloamorim | sorry about my english =) |
| 19:55.36 | TazzNZ | voxter: what settings are the other carriers using, rfc as well ? |
| 19:55.37 | file | otherwise the SDP is fine going to your softphone |
| 19:55.55 | voxter | TazzNZ: yup. |
| 19:55.57 | marceloamorim | I said that because there is some problem when you try to run asterisk if you already started the mysql |
| 19:56.37 | TazzNZ | voxter: how busy is the host running the VM ? |
| 19:56.59 | voxter | TazzNZ: not busy. its not a clock or idle steal issue i don't think |
| 19:57.04 | TazzNZ | and how many CPU's did you assign to the VM, and what is the hypervisor |
| 19:57.24 | voxter | TazzNZ: one full cpu assigned, and affinity set so its not shared, and its ivm |
| 19:57.46 | zopsi | file: http://pastebin.com/52mcaBii |
| 19:58.02 | TazzNZ | uhm.....ivm ? |
| 19:58.14 | file | zopsi, this won't solve your softphone fyi... I don't know why it is responding how it is |
| 19:58.26 | voxter | KVM. |
| 19:58.27 | voxter | sorry. |
| 19:58.35 | TazzNZ | lol - thought so - just wanted to confirm |
| 19:58.48 | zopsi | file: I think I'm an idiot. the external_media_address is my public IP or the sip providers IP? |
| 19:58.55 | file | zopsi, your public IP |
| 19:59.05 | file | signaling and media should be your public IP |
| 19:59.49 | TazzNZ | voxter: personally, I would try this via a physical box - I have seen some issue where we had a dedicated host running a single VM |
| 20:00.07 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:00.07 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:00.26 | zopsi | file: theres the issue. I am an idiot. |
| 20:00.31 | zopsi | well one issue anyways |
| 20:00.36 | TazzNZ | imho, hypervisors and realtime stuff, like DTMF, transcoding and the likes do not mix at all |
| 20:00.54 | voxter | TazzNZ: yeah, i run over 100 asterisk boxes this way, i dont think KVM is the issue here |
| 20:01.12 | voxter | TazzNZ: not to mention its rfc2833, so its not actual audio/transcoding, its a SIP Packet |
| 20:01.47 | file | voxter, what version of Asterisk on those 100 boxes? |
| 20:01.53 | voxter | file: 11.5 |
| 20:01.57 | file | nifty |
| 20:02.03 | TazzNZ | oh true - how did I miss that :( |
| 20:03.08 | TazzNZ | voxter: when you did the network capture, was that on the VM or the host ? |
| 20:04.24 | voxter | TazzNZ: Carrier, SBC, PBX, all three. |
| 20:05.30 | *** join/#asterisk af_ (~af@93-43-45-195.ip90.fastwebnet.it) |
| 20:05.34 | TazzNZ | voxter: have you tried enabling debug on asterisk to see if it shows more/any info as to why it might be rejecting the packet/DTMF ? |
| 20:05.36 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 20:05.48 | Cuzner | TazzNZ: yea, i lost count of how many * VMs i run w/o DTMF issues in vsphere :P |
| 20:05.56 | voxter | the pcaps all show in wireshark that the correct dtmf was received at each spot. |
| 20:06.07 | Cuzner | enable inband on the peer and have it done with :P |
| 20:06.15 | Cuzner | haha, don't listen to that advice. |
| 20:06.23 | TazzNZ | Cuzner: yip - KVM "is the best" out there, but even there it sucks :( |
| 20:06.42 | TazzNZ | for inband stuff etc |
| 20:06.55 | mjordan | voxter: check the RTP timestamps and the seqno |
| 20:07.40 | mjordan | voxter: If the seqno isn't sequentially increasing and the timestamp isn't increasing appropriately - and if the DTMF End packets aren't maintaining the same timestamp - you may get interesting behaviour |
| 20:07.41 | TazzNZ | oh Cuzner to note on my comment, "when using a redhat based guest" |
| 20:07.54 | mjordan | voxter: I'd also enable RTP debug on Asterisk. See if the RTP stack at least is happy with the DTMF |
| 20:08.03 | Cuzner | TazzNZ: just ignore me, i'm a nubcake |
| 20:08.18 | mjordan | voxter: it may be that the RTP stack is getting the DTMF and passing it up the chain, only to have the core get unhappy due to DTMF digits arriving too close together (or something like that) |
| 20:08.42 | TazzNZ | mjordan: even with rfc2833 dtmf ? |
| 20:08.44 | mjordan | You may also want to turn on the DTMF logging in logger.conf |
| 20:09.12 | mjordan | yes. There's code in channel.c that attempts to handle DTMF arriving on top of each other. And with out of order packets, it is not impossible for that to happen |
| 20:09.27 | mjordan | I've seen pcaps of digits stomping on top of each other |
| 20:09.44 | mjordan | and people probably don't want a digit sequence of 1 - 2 to come out as 1 - 2 - 1 - 2 |
| 20:09.46 | TazzNZ | cool - so the rfc dtmf is timestamped to match the rtp stream ? |
| 20:09.50 | voxter | mjordan: Ah - thats a good suggestion. I have enabled DTMF debugging, it definitely does feel like a seqno/timestamp issue although on all of my equipment, times are correct... DTMF logging appeared to show all digits but claimed it was ignoring some, but not for any specific reason |
| 20:09.55 | voxter | mjordan: which implies probably sequence or timestamp |
| 20:10.12 | Cuzner | CenturyLink doesn't allow for rfc2833 dtmf in g.711 RTP, which makes absolutely no sense to me... I'm trying to get them to enable it tomorrow so I can stop forcing inband for them. |
| 20:10.57 | voxter | mjordan: Im pretty sure this carrier is sending both RFC2833 + inband at the same time, we're discarding inband by the looks of it, btu it could be contributing to the mess. |
| 20:11.04 | TazzNZ | Cuzner: I have had some weird requests from "ITSP" when it comes to SIP - I hate it when they run SIP like the old telco's |
| 20:11.06 | mjordan | voxter: DEBUG logs at lvl 1 will show more 'reasons' when rtp debug enabled as well |
| 20:11.12 | mjordan | voxter: ew |
| 20:11.19 | voxter | mjordan: hooray pacwest. :| |
| 20:12.11 | Cuzner | TazzNZ: dealing with carriers is the worst part of my job, qwest/clink has now taken ~6months now to deal with this DTMF issue for us. They're finally going to try something tomorrow. |
| 20:13.13 | Cuzner | Level3 is by far the worst to deal with though, 2nd only to VZN :P |
| 20:13.31 | TazzNZ | Cuzner: sounds typical - I have diverted certain calls via ISDN to avoid calling a service on my ITSP because of DTMF issues, I now route from SIP, to ISDN, from there into the ITSP which takes it back to SIP and over to the service |
| 20:14.20 | TazzNZ | and this is only one of the issues, don't get me started on the blacklisting issues :) |
| 20:14.43 | Cuzner | most of our TF taffic likes to take the qwest trunk, where you guessed it, DTMF problems. I just force inband, since it's all I can do until they fix their shit. |
| 20:15.49 | TazzNZ | so mjordan - do you know what the order of preferance is in channel.c of DTMF ? |
| 20:16.08 | TazzNZ | if all the DTMF get's in at the same time that is |
| 20:16.11 | mjordan | channel.c doesn't care |
| 20:16.25 | mjordan | it gets a control frame that indicates the beginning or end of a DTMF |
| 20:16.31 | mjordan | where it came from is a 'meh' |
| 20:16.39 | mjordan | that's up to the individual channel drivers |
| 20:17.05 | TazzNZ | right - I see |
| 20:17.12 | mjordan | channel.c tries to make sure that a subsequent begin or end doesn't stomp on a DTMF that's currently being played out on a channel |
| 20:17.30 | TazzNZ | oh I see |
| 20:17.42 | mjordan | your settings in sip.conf (or pjsip.conf!) tell you what kinds of DTMF you want to have |
| 20:17.46 | TazzNZ | so the first begin "rules them all" |
| 20:18.01 | mjordan | if you don't specify 'inband', for example, there is no DSP set up that will process the audio |
| 20:19.54 | *** part/#asterisk toothe (~mongolian@unaffiliated/toothe) |
| 20:20.08 | TazzNZ | cool - thanks mjordan :) |
| 20:21.32 | TazzNZ | Cuzner: my ITSP doesn't accept inband DTMF (which will most prob work, since we have a dedicated fibre to them) but then, their stuff doesn't work with rfc DTMF, so kinda a rock and a hard place thing |
| 20:23.13 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 20:27.54 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 20:30.57 | voxter | mjordan: so this is interesting. |
| 20:31.06 | voxter | mjordan: i found that the timestamp on a couple of the broken digits is screwed up |
| 20:31.41 | voxter | mjordan: Digit 1 timestamp: 58845880. Digit 2 timestamp: 381. Digit 3 timestamp: 58853440 |
| 20:31.49 | voxter | I wonder wtf is doing that?! |
| 20:31.50 | file | uh |
| 20:32.33 | TazzNZ | that is quite interesting - you learn something new every day hey |
| 20:38.32 | mjordan | heh |
| 20:38.45 | mjordan | since timestamps are supposed to have the same 'wallclock' in RTP ... that's bad. |
| 20:42.23 | voxter | yeah this is going through a freeswitch box in the middle |
| 20:42.42 | voxter | i dont have a clue why it would be screwing up the timestamp so badly... time to investigate! |
| 20:42.52 | voxter | Thanks for the pointers! :) |
| 20:43.15 | mjordan | np, good luck :-) |
| 20:53.00 | *** join/#asterisk beanie (~beanie@bgareth.plus.com) |
| 21:03.58 | Kobaz | so, i have a wiggity wack problem |
| 21:04.15 | Kobaz | i have a polycom vvx 400 that's asking for 3111-46157-002.sip.ld |
| 21:04.25 | Kobaz | which doesn't exist in any of the firmware zip files, as far as I can tell |
| 21:04.48 | Kobaz | i've searched everything from 4.1.4 to 3.2 |
| 21:05.30 | Kobaz | and if i put on a combined sip.ld any version between 3.2 and 4.1.4 it says image not compatible |
| 21:06.53 | malcolmd | that is wack |
| 21:06.59 | malcolmd | is it a development phone? |
| 21:08.41 | Chainsaw | Kobaz: VVX is a combined image these days. You should be using UCS 4. |
| 21:09.04 | Chainsaw | Kobaz: Make sure to select the exact model, the SoundStation/SoundPoint firmware train is not the same. |
| 21:09.31 | zopsi | what is a decent softphone for Kubuntu other than Jitsi? |
| 21:09.41 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:10.54 | Kobaz | malcolmd: i have many other vvx phones set up no problem, i dont think it's a dev phone |
| 21:11.06 | Kobaz | the weird thing is that this was previously set up and working just fine |
| 21:11.19 | Chainsaw | Kobaz: Did you switch from SIP to UCS? |
| 21:11.24 | Kobaz | and one day it was like factory reset, provisioning all gone |
| 21:11.25 | Chainsaw | Kobaz: Missing the upgrader/downgrader? |
| 21:11.42 | Kobaz | i'm using Polycom UC Software 4.0.4 for everything |
| 21:11.54 | Chainsaw | Kobaz: Okay, make sure you have the UCS upgrader for the hardware type. |
| 21:12.03 | Chainsaw | Kobaz: And the last pre-UCS bootblock. |
| 21:12.21 | Kobaz | k |
| 21:12.35 | Chainsaw | Kobaz: Knowing the VVX it'll go for a bootblock first, then the UCS upgrader, then the UCS firmware itself. 3 reboots and it'll be fine. Factory resets are... deep. |
| 21:13.32 | Kobaz | what's weird |
| 21:13.45 | Kobaz | is that the very first file it's asking for after mac.cfg... is 3111-46157-002.sip.l |
| 21:13.46 | Kobaz | d |
| 21:14.43 | Kobaz | which i can't find anywhere |
| 21:16.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:17.57 | Kobaz | and what's weird, is the release notes for 4.1.4 say that it's included: 3111-46157-002.sip.ld SIP application executable for VVX 400 |
| 21:19.00 | Kobaz | oooooh |
| 21:19.03 | Kobaz | there's new 5.x firmware |
| 21:19.26 | Kobaz | but still, that doesn't explain why 3111-46157-002.sip.ld is missing from 4.1.4 |
| 21:19.53 | Chainsaw | The 5.x is Lync though, isn't it? |
| 21:20.02 | Chainsaw | Not sure you want that. That's almost-but-not-quite-SIP. |
| 21:20.15 | Chainsaw | Kobaz: That's an upgrader, not a UCS core file. |
| 21:20.39 | Chainsaw | Kobaz: You want the latest non-UCS bootblock, the latest UCS upgrader and potentially the SIP downgrader, even though you won't use it. |
| 21:21.10 | Kobaz | http://downloads.polycom.com/voice/voip/uc_sw_releases_matrix.html |
| 21:21.13 | Kobaz | that's upgraders? |
| 21:21.44 | Kobaz | 4.62.119.138 - - [09/Jun/2014:17:02:06 -0400] "GET /3111-46157-002.sip.ld HTTP/1.1" 404 300 [FileTransport PolycomVVX-VVX_400-UA/5.1.4.0844 Type/Updater] |
| 21:21.57 | Chainsaw | Kobaz: VVX400 you say? |
| 21:22.00 | Kobaz | yeah |
| 21:22.15 | Chainsaw | is on a train held up by a jumper |
| 21:22.23 | Chainsaw | So I shall have a look for you fortwith. |
| 21:22.53 | Kobaz | i just downloaded 5.0.2 and i'm checking if that firmware file is in there |
| 21:23.04 | Kobaz | yeap |
| 21:23.05 | Kobaz | there it is |
| 21:23.15 | Kobaz | that was a hell of a lot of searching |
| 21:23.42 | Chainsaw | Kobaz: Otherwise 4.1.4 split may have it. |
| 21:23.53 | Kobaz | it doesn't |
| 21:23.55 | Chainsaw | Kobaz: I do apologise, I was applying SoundPoint IP thinking there. There is no SIP firmware for the VVX400/410. |
| 21:24.27 | Chainsaw | Kobaz: Must be fairly new. |
| 21:24.31 | Kobaz | yeah |
| 21:24.37 | Kobaz | so weird |
| 21:24.44 | Kobaz | like, in a group of three vvx 400 phones |
| 21:24.50 | Kobaz | this one just randomly stopped working one day |
| 21:25.00 | Kobaz | and it got power cycled and lost its settings and firmware |
| 21:26.58 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 21:27.18 | sbrath | Anyone have experience with a Sangoma Vega 100 unit? |
| 21:27.41 | Chainsaw | Kobaz: NVRAM checksum failure. It takes that very seriously. |
| 21:27.43 | Qwell | ~ask |
| 21:27.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:28.35 | sbrath | I can't seem to get DTMF to transmit from an inbound call over my Sangoma to Asterisk via SIP. I have tried RFC and INFO modes on the Vega, and the peer is configured for RFC on Asterisk? Any sugestions? |
| 21:28.54 | beanie | Does anybody rate this http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 21:29.31 | Kobaz | Chainsaw: ah |
| 21:29.59 | Chainsaw | Kobaz: Brown-out on the power supply, particularly on 48V PoE. |
| 21:30.06 | Chainsaw | Kobaz: Seen it happen once or twice on our IP670s. |
| 21:31.26 | Kobaz | ah, interesting |
| 21:33.22 | gusto | anyone an idea how to get rid of this one: tcptls.c: SSL_shutdown() failed: 5 ? |
| 21:36.36 | beanie | how do I know whether I have Asterisk Security Framework (Asterisk 10+). |
| 21:39.14 | ChannelZ | I don't think you have a choice |
| 21:39.43 | ChannelZ | It's basically just a new means of logging |
| 21:40.11 | ChannelZ | You just need to turn it on in logger.conf (security) |
| 21:41.44 | beanie | oh right, how can i be sure that i have a version with it contained within the logger.conf |
| 21:42.00 | beanie | i suppose i need to find the version of asterisk im using |
| 21:42.16 | ChannelZ | core show version |
| 21:43.46 | beanie | yep just got it |
| 21:43.48 | beanie | im on 11.6 |
| 21:43.50 | beanie | so that'll do! |
| 21:44.09 | beanie | and is it not turned on by defaul? |
| 21:44.10 | beanie | t |
| 21:45.46 | ChannelZ | I'm not sure if the security log is on by default |
| 21:46.01 | ChannelZ | I don't think so |
| 21:46.23 | ChannelZ | It's commented out in v12 anyway |
| 21:46.43 | beanie | This is what is contained within the file - [general] |
| 21:46.43 | beanie | #include logger_general_additional.conf |
| 21:46.43 | beanie | #include logger_general_custom.conf |
| 21:46.43 | beanie | [logfiles] |
| 21:46.44 | beanie | #include logger_logfiles_additional.conf |
| 21:46.46 | beanie | #include logger_logfiles_custom.conf |
| 21:48.48 | beanie | Channelz - im not sure which one it is? |
| 21:48.54 | ChannelZ | security |
| 21:49.46 | ChannelZ | you could/would maybe put it in logger_logfiles_additional.conf |
| 21:50.01 | ChannelZ | whatever that means to your setup |
| 21:50.47 | beanie | oh is this not something that will already be in one of these files |
| 21:50.53 | beanie | i thought it would just be commented out? |
| 21:51.12 | [TK]D-Fender | is that ... a question? |
| 21:51.14 | ChannelZ | well it's in the default one, but yours obviously is not default because all of those #includes aren't |
| 21:51.32 | ChannelZ | (aren't default that is) |
| 21:51.48 | beanie | well i haven't edited it :) |
| 21:51.55 | beanie | its just what was installed |
| 21:52.00 | ChannelZ | you basically want "security => security" somewhere under [logfiles] |
| 21:52.10 | ChannelZ | So this is probably FreePBX or something then |
| 21:52.51 | beanie | yep freepbx - i thought that this would just be an asterisk thing |
| 21:53.07 | beanie | additional has this in it - logger_logfiles_additional.conf |
| 21:53.26 | beanie | full => debug,error,notice,verbose,warning |
| 21:53.27 | beanie | console => debug,error,notice,verbose,warning |
| 21:53.27 | beanie | rather |
| 21:54.31 | ChannelZ | well I have no idea if it'll write over one or the other or not |
| 21:54.40 | beanie | :) thanks though |
| 21:54.47 | beanie | TK Defender - the question was how do I know whether I have Asterisk Security Framework (Asterisk 10+). |
| 21:55.00 | ChannelZ | by having Asterisk 10+ |
| 21:55.04 | beanie | im trying to follow this - http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 21:55.27 | ChannelZ | I'd just add it to that 'additional' |
| 21:55.54 | ChannelZ | If it stops working, you'll know where to look.. |
| 21:56.43 | beanie | well i thought it was either something that was there or not depending on the version of asterisk |
| 21:56.47 | beanie | :) |
| 21:56.56 | beanie | i didnt think you would need to reference it |
| 21:57.05 | beanie | more likely delete any commenting out |
| 21:57.08 | ChannelZ | both logger_logfiles_additional.conf and logger_logfiles_custom.conf get read so either will work, I just don't know if it's UI overwrites one but not the other. There's some "logical" FreePBX reason why there are 2.. |
| 21:57.30 | beanie | argggghhhh how confusing |
| 21:57.39 | beanie | i just want to bloomin secure my pbx |
| 21:58.03 | beanie | without a bloody "solution" that causes more problems than it seems to solve |
| 21:58.42 | ChannelZ | I'd figure there's a checkbox somewhere in FreePBX to turn on the security log so it does it in its own preferable way, but I don't run FreePBX so I dunno |
| 21:59.03 | beanie | but is there not more than one security log? |
| 22:01.06 | ChannelZ | eh? |
| 22:05.03 | *** join/#asterisk Matt (matt@freenode/staff-emeritus/matt) |
| 22:55.28 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 22:57.49 | *** join/#asterisk gusto (~gusto@178.143.0.10) |
| 22:58.29 | gusto | i have a theory why ppl are bithing about this problem with tcp tls bla bla when the ressource goes offline, it comes around every minute so i think it is the qualification |
| 22:58.55 | gusto | when ppl disable qualification on server (only client needs to keep the connection open) then those messages would disappear |
| 23:18.07 | *** join/#asterisk fisted__ (~fisted@unaffiliated/fisted) |
| 23:27.24 | beanie | im following the guidance after a massive coffufall where i have established that you only edit custom files |
| 23:27.32 | beanie | the instructions im following advise "Likewise, you willl also need to ensure the date format has been changed in logger.conf to "dateformat=%F %T"." |
| 23:27.48 | beanie | do i somehow change these in a custom file to or so I have to change these in the original? |
| 23:35.14 | *** join/#asterisk mrjazzman (~mrjazzman@230.210.233.220.static.exetel.com.au) |
| 23:37.43 | mrjazzman | Hi All, Need some help getting the right terminology to find the right docs. I have an asterisk server which is being used as a DR system for landlines. Basically when a fault on our ISDNs happen I diverto to an extension on this Asterisk service and then use the extensions.conf to divert to a range of mobile phones. My issue is that I am using "DAIL_GROUP" to do this and it doesn't keep track of which connections are currently |
| 23:50.04 | ChannelZ | Do what now? |
| 23:51.23 | TazzNZ | beanie: do you have freepbx installed ? |
| 23:52.44 | beanie | yes TazzNZ :) |
| 23:55.52 | TazzNZ | beanie: ah yes - then it's all _custom |