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| 00:44.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:56.46 | SeRi | Penguin: so dijib still does not know who "hacked" his pbx |
| 01:56.48 | SeRi | amazing |
| 02:15.31 | nobodyathome | like this |
| 02:15.54 | dijib | nobodyshome |
| 02:16.47 | Penguin | seri: Maybe not. He did call me a liar, though. |
| 02:17.03 | SeRi | oh snap |
| 02:17.06 | SeRi | :( |
| 02:17.18 | SeRi | sorry dude |
| 02:17.31 | SeRi | I am trying to help him understand iirc basics |
| 02:17.44 | SeRi | and how somebody was able to get in his pbx |
| 02:21.48 | SeRi | ok |
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| 02:32.01 | Penguin | seri: iirc basics? |
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| 02:52.30 | SeRi | Penguin: IRC* |
| 02:52.44 | Penguin | seri: What do IRC basics have to do with his PBX? |
| 02:53.04 | SeRi | Penguin: he has no clue how people in IRC cand find each others IP |
| 02:53.11 | Penguin | oh |
| 02:53.18 | SeRi | :) |
| 02:53.25 | Penguin | He doesn't know we have eyes. Got it. |
| 02:53.33 | SeRi | ROFL |
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| 02:56.11 | Penguin | seri: More than just IRC basics, he published his FQDN multiple times. |
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| 03:01.49 | ngharo | anyone run into the 'invalid opcode' error on startup while running inside a VM? |
| 03:02.05 | ngharo | looks like my first try would be to disable optimizations during the build but curious if there is a better solution |
| 03:03.41 | ngharo | tries without build_native |
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| 04:01.38 | phix | Penguin, SeRi: Who are you talking about? |
| 04:02.19 | Penguin | phix: dijib |
| 04:02.39 | phix | ah |
| 04:02.47 | phix | pwned |
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| 04:57.42 | tehrabbitt | what is the reccomended version of asterisk nowadays? |
| 05:03.24 | [TK]D-Fender | same as always, the latest LTS release |
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| 06:12.10 | ChannelZ | eleventy point whateva! |
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| 06:43.22 | snadge | which latest LTS release? the certified one? ;) |
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| 06:47.27 | hrolf | Hi #asterisk. |
| 06:47.42 | hrolf | I have issue that I receive calls on SIP, then execute AGI. |
| 06:48.01 | hrolf | Right now, there is one call stuck, it is stuck in the AGI command. |
| 06:48.23 | hrolf | But on the other machine (where FastAGI) is running no connection is received. |
| 06:48.54 | hrolf | When I do core show channels I get this line: |
| 06:49.03 | hrolf | SIP/308-00000456 308@access:1 Ring AGI(agi://192.168.20.41/testtr |
| 06:49.17 | hrolf | The call is stuck for like 40 hours |
| 06:50.04 | hrolf | I did check the TCP connections on the 192.168.20.41 machine, and there are no connections from Asterisk, it is only listening on 457 |
| 06:50.08 | hrolf | 4573 port |
| 06:51.17 | hrolf | How do I debug this? |
| 06:53.06 | gerritfromsa | hrolf, I believe asterisk never received the BYE and got stuck , there is a remote possibility that you wont be able to debug because |
| 06:53.11 | gerritfromsa | it already happened |
| 06:53.38 | gerritfromsa | If you were running a pcap trace 40hours ago , you still had a chance |
| 06:54.13 | gerritfromsa | logs is in /var/log/asterisk/messages |
| 06:58.12 | hrolf | gerritfromsa: BYE? No I log all connections at the FastAGI application, I never received a call with the channel SIP/308-00000456. |
| 06:58.56 | gerritfromsa | So the request never made it to 192.168.20.41 ? |
| 06:59.00 | hrolf | gerritfromsa: Nope. |
| 06:59.46 | hrolf | gerritfromsa: Other strange thing is that I do netstat | less and get this: |
| 06:59.48 | hrolf | tcp 0 0 192.168.20.40:52307 ivr-server.ahiml-main.:4573 ESTABLISHED |
| 06:59.57 | hrolf | A connection is established |
| 07:00.22 | hrolf | ivr-server is where our FastAGI application is being hosted i.e. 192.168.20.41 |
| 07:00.57 | hrolf | but when I view the TCP connections in 41 machine, there are no connections established, just one connection on port 4573 which is listening for AGI connections. |
| 07:01.48 | gerritfromsa | Try to add the option in AGI if a time-out/error occurs to log and Hangup |
| 07:02.45 | hrolf | gerritfromsa: What option is it? |
| 07:05.03 | hrolf | gerritfromsa: And judging from how it is at present. The connection is established, no errors I believe ? |
| 07:05.23 | hrolf | gerritfromsa: how will it detect, if it ever occurs again in future? |
| 07:05.28 | gerritfromsa | If the AGI command returns -1 |
| 07:06.15 | gerritfromsa | I believe the connection you see is the socket , but does not represent a successful originate |
| 07:06.17 | hrolf | gerritfromsa: In this case, it hasn't returned yet. |
| 07:06.35 | gerritfromsa | Hence the timeout condition |
| 07:06.37 | hrolf | gerritfromsa: According to it, it is still connected. |
| 07:06.53 | hrolf | gerritfromsa: And no errors. |
| 07:07.34 | gerritfromsa | Did you create the agi script? |
| 07:08.30 | hrolf | gerritfromsa: Umm? Yes I created an application running on FastAGI, the IVR. |
| 07:08.47 | hrolf | on .NET. |
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| 07:10.08 | gerritfromsa | sorry then I'm lost |
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| 07:10.19 | gerritfromsa | Can't help with .NET |
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| 07:15.23 | ChannelZ | runs away |
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| 07:24.33 | linocisco | hi |
| 07:24.35 | linocisco | hi all |
| 07:24.58 | linocisco | I want to setup CCTV. is there any way to setup CCTV with asterisk based? |
| 07:26.23 | gerritfromsa | how do you want the cctv to interface with * ? |
| 07:27.05 | gerritfromsa | I've got asterisk to dial me when theres movement - something like that ? |
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| 07:34.10 | linocisco | I heard there are ip camera |
| 07:34.37 | gerritfromsa | ok |
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| 07:57.39 | magespawn | good morning all |
| 08:02.07 | phpboy | haazit? |
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| 08:23.39 | polysics | hi! just a minor thing, installing the latest 11 proceeds correctly if uuid-dev is not installed, but the resulting binary has no RTP capabilities |
| 08:23.49 | polysics | I assume that is by desgin, to decouple functionality |
| 08:24.00 | polysics | but it should be on by default as most people do want RTP :) |
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| 08:49.13 | makmak78 | Hi everybody. Is there anybody here that lives in Sweden? |
| 08:50.09 | makmak78 | Im looking for someone familiar with the asterisk code |
| 08:50.48 | makmak78 | Im actually looking for a solution to att PDD value to timeout |
| 08:51.04 | makmak78 | anybody with that info perhaps? |
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| 08:59.46 | gerritfromsa | pdd value ? |
| 09:00.20 | kaldemar | post dial delay |
| 09:00.29 | gerritfromsa | phpboy, just knew you where local when I sa haazit ! lol |
| 09:01.25 | gerritfromsa | makmak78, Add the D() value to your Dial() statement - lets say you want it to pause 0.5 sekonds after dialling |
| 09:02.11 | kaldemar | makmak78: asterisk can't really know that, if you really want the time between Dial app execution and caller getting an indication. |
| 09:02.13 | gerritfromsa | makmak78, use it like this exten => _X.,1,Dial(DAHDI/g1/${EXTEN},,rD(w)) |
| 09:02.43 | gerritfromsa | Every "w" represents 0.5 sec |
| 09:03.14 | kaldemar | D() sends DTMF after the callee has answered. |
| 09:03.20 | gerritfromsa | You can also pass DTMF this way after Dial |
| 09:03.38 | gerritfromsa | But the w - pauses |
| 09:04.12 | kaldemar | how does sending DTMF in the ear of the callee *after* an answer increase the timeout for app Dial? |
| 09:05.55 | gerritfromsa | Your NOT sending DTMF you're sending a PAUSE/DELAY |
| 09:06.19 | *** join/#asterisk atha (~athayde@unaffiliated/athayde) |
| 09:06.25 | kaldemar | at the point D() is used, the caller has already gotten both a ringing indication and an answer => it has no effect whatsoever on the timeout. |
| 09:07.01 | gerritfromsa | Thats what I understand under POST DIAL delay |
| 09:07.15 | gerritfromsa | Its comes after dialling? |
| 09:08.16 | kaldemar | PDD is the time the calling user experiences between entering the last digit in the phone and hearing a ringing indication. |
| 09:09.04 | gerritfromsa | and not ending with # ? |
| 09:09.04 | kaldemar | so it's really not even the time between a Dial and ringing indication. |
| 09:09.26 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:10.18 | gerritfromsa | This is dialplan related , so if you have 2 competing Pattern Matches on longer than the one you're dialing , you will have the 3 sec delay |
| 09:11.03 | gerritfromsa | But if you only have one match it dial immediately , so if you want a delay use Wait() |
| 09:11.08 | kaldemar | only with overlap dialing and if digit timeout is 3 seconds. |
| 09:11.21 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 09:11.26 | makmak78 | What i need is the time from originate to ring event |
| 09:11.53 | makmak78 | and add that value to ringtimeout |
| 09:11.56 | kaldemar | he does not want delay. he wants to increase the timeout for app Dial with the PDD. |
| 09:12.24 | PLMg | hey, does anyone know the default admin password for SPA303? (I know this is not really an asterisk question but I didn't find anything usefull on google) |
| 09:12.25 | kaldemar | that cannot be done from dialplan. |
| 09:12.26 | gerritfromsa | makmak78, you simply want to log this? |
| 09:12.44 | makmak78 | i want to both log it and add it to ring timoeut if possible |
| 09:13.16 | gerritfromsa | The difference is miiliseconds ? |
| 09:13.59 | gerritfromsa | Unless you're using DAHDI(or ZAPATA) |
| 09:14.35 | makmak78 | i know i can extract the pdd value and add it to cdr. my problem is to add the pdd value to ring timeout |
| 09:14.53 | gerritfromsa | makmak78, you can actually |
| 09:15.17 | *** part/#asterisk PLMg (PLMg@78.96.151.225) |
| 09:15.41 | makmak78 | Eg. we set 20 second timeout for originate. we check pdd and lets say it takes 5 sec to get first ring. then we need to add those 5 seconds to ring timeout so it doesnt hangup before 25 seconds |
| 09:15.42 | gerritfromsa | When you start the dialplan use SET(cdr(userfield)=THE TIME NOW) |
| 09:16.27 | makmak78 | and i need to know if it is duable in the asterisk code |
| 09:16.32 | makmak78 | doable* |
| 09:17.01 | gerritfromsa | using Dial() or originate via the AMI? |
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| 09:23.08 | makmak78 | in the asterisk soruce code |
| 09:23.24 | makmak78 | Im using originate |
| 09:23.52 | makmak78 | but as far as i know, if it is possible, then it is in the src code |
| 09:25.23 | kaldemar | you'll have to modify the source. |
| 09:26.11 | makmak78 | Yes, and my question is, does anybody have experience in that? |
| 09:26.14 | makmak78 | in here |
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| 09:52.42 | LooserOuting | gerritfromsa: hello gerritfromsa, you helped me two days ago with my fax problem. can i ask you 4 advise regarding a bug I reported ? |
| 10:00.15 | LooserOuting | I found a small function in the res_fax code, that may return a false value. I just found it by reading the code, i have problems now to report a bug because i don't run into an issue using asterisk and I git no debug log or something else. |
| 10:00.59 | *** join/#asterisk the_5th_wheel (~edd@105-236-54-21.access.mtnbusiness.co.za) |
| 10:09.04 | gerritfromsa | go ahead |
| 10:09.17 | gerritfromsa | sorry lots of lag |
| 10:10.16 | *** part/#asterisk StaRetji (~LittleAll@178.79.6.17) |
| 10:20.43 | LooserOuting | i dunno what to do. https://issues.asterisk.org/jira/browse/ASTERISK-22790 |
| 10:21.11 | LooserOuting | the bug was close |
| 10:21.17 | LooserOuting | +d |
| 10:22.36 | LooserOuting | i don'T even know if i am right :-) |
| 10:24.13 | Greenlight | LooserOuting: You're best bet is to ask in #asterisk-dev if its related to the code. Perhaps one of them will have a spare few minutes to take a look at some point, even if it's just to confirm it's a bug or not |
| 10:25.34 | LooserOuting | Greenlight: OK. thanks, i'll try. |
| 10:25.40 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-stchjsdwepywuplk) |
| 10:34.09 | LooserOuting | oh dear, maybe i'll ask when there is more action in asterisk-dev. maybe in a few hours |
| 10:37.06 | Greenlight | Yea, patience is needed on that one :) |
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| 12:27.32 | skrusty | quick question, if i have a srv record pointing to a master and failover proxy, lets say it's _sip._udp.my.domain.com, what should the A record for my.domain.com be? Does it matter? |
| 12:27.51 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:29.22 | Greenlight | You'd most often point the A record to the master server, to support devices which ignore SRV records |
| 12:30.01 | Greenlight | As a side note, Asterisk doesn't properly support SRV records either. |
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| 12:34.27 | skrusty | Greenlight: cheers :) |
| 12:34.46 | skrusty | and thanks for the heads up |
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| 13:44.49 | *** join/#asterisk ghost75 (~quassel@dslb-188-105-028-174.pools.arcor-ip.net) |
| 13:45.33 | ghost75 | voipmonitor works only on same host as * or with spanport right? |
| 13:51.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:51.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:12.32 | skrusty | ghost75: no, you can use ssh to extend monitoring to a remote host |
| 14:13.51 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 14:13.54 | *** join/#asterisk lnb (~lnb@CPE0002b3c8018e-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 14:15.14 | elred_ | Hi. I need to have a global MOH in a conference room (i am using Meetme) actually the MOH is started from beginning at each conference entering. Do i have the possibility to use another MOH's method (not mode=files ?) that will stream MOH globally ? |
| 14:15.23 | elred_ | Thanks you |
| 14:16.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ptdkyyezspnoaqro) |
| 14:16.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:20.43 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:22.48 | elred_ | I wonder using application= would bind a single process thus MOH global ? Thanks |
| 14:22.52 | gerritfromsa | elred_, Yes you can stream , it's described in the sample musiconhold.conf - check for [ulawstream] |
| 14:23.48 | gerritfromsa | You can use streamplayer or mpg123 |
| 14:24.15 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:24.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:24.56 | lnb | for some reason an analogue phone -> ATA now has no sound. Can dial out, and rings when calls come in. What might cause this? |
| 14:25.11 | lnb | Opened firewall completely but that did not do anything |
| 14:25.30 | lnb | this phone was working right before today |
| 14:27.37 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 14:27.38 | gerritfromsa | sound issues both ways or one directional ? |
| 14:30.30 | lnb | both ways |
| 14:31.45 | gerritfromsa | can you see the rtp flowing to and fro ? |
| 14:35.04 | lnb | how? from tcpdump? |
| 14:37.00 | gerritfromsa | jip |
| 14:48.02 | ghost75 | skrusty: software like ssh zenpack? |
| 14:49.42 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 14:50.10 | Ice_Strike | What does this error mean, Probation passed - setting RTP source address to .. |
| 14:50.58 | Chainsaw | Ice_Strike: It is not an error actually. |
| 14:51.12 | Chainsaw | Ice_Strike: It means Asterisk is now sure of both endpoints for the RTP (audio) stream. |
| 14:51.38 | Chainsaw | Ice_Strike: As in, it is sure that no further re-invites will occur, and that data is flowing correctly. |
| 14:52.21 | Ice_Strike | Ah I see, I have upgraded from 1.4 to 11 and I didnt see this before :) |
| 14:54.28 | Chainsaw | Ice_Strike: Quite a jump there. Good change though, you'll like 11. |
| 14:54.44 | Ice_Strike | Chainsaw Just a few changes and seem working so far. |
| 14:54.50 | Chainsaw | nods approvingly |
| 14:54.58 | Ice_Strike | Chnages to AMI and Dial Plan. |
| 14:55.09 | Chainsaw | Yeah, I'm just glad you weren't stuck on 1.2 |
| 14:55.33 | Chainsaw | That dial-plan is so different, with the priority jumping that getting away from it is proper hard work. (I went 1.2 -> 1.6.0, skipping 1.4) |
| 14:55.55 | Ice_Strike | Chainsaw Astrisk are hosted on dedicated server and the hardware phone are here.. how to test if latency is ok? |
| 14:56.06 | Ice_Strike | I don't use priority jumping |
| 14:56.26 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:56.34 | Chainsaw | Ice_Strike: Latency on audio streams? Generally an echo test on the old and then the new system to compare. |
| 14:56.43 | Chainsaw | Ice_Strike: Any benchmark is meaningless without a baseline. |
| 14:57.23 | Ice_Strike | I've just dialed to my mobile phone and test |
| 14:57.25 | Ice_Strike | seem fine |
| 14:58.18 | Chainsaw | Ice_Strike: Greenlight may have a more scientific approach to latency measurement. Runs a much bigger operation. |
| 14:59.01 | Ice_Strike | Chainsaw I had out of sync issue of recorded files only when after transferring the calls.. That only happen sometime. |
| 14:59.17 | Ice_Strike | Greenlight suggest me to change from Monitor() to MixMonitor() |
| 14:59.30 | Greenlight | nods |
| 15:00.56 | Ice_Strike | Minimum = 27ms, Maximum = 37ms, Average = 30ms |
| 15:01.02 | Ice_Strike | That is when I ping asterisk server |
| 15:01.56 | Greenlight | A lot of jitter... |
| 15:02.36 | Greenlight | Well, perhaps not enough to cause issues, but enough to make me wonder "what's causing that" |
| 15:03.32 | Greenlight | Leave mtr running for a while and see what results you get |
| 15:05.11 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:05.33 | Ice_Strike | wtf, wrong time Wed Oct 30 15:15:59 GMT 2013 |
| 15:05.35 | Ice_Strike | on linux lol |
| 15:06.18 | Greenlight | Careful with ntpupdate with large difference when asterisk is live |
| 15:06.26 | Greenlight | I've had.... issues before |
| 15:07.21 | Ice_Strike | I don't use auto update via ntpupdate |
| 15:11.03 | *** join/#asterisk danjenkins_ (~danjenkin@62.254.236.250) |
| 15:12.33 | lnb | Can hear myself in the packet capture using the analogue phone. Does that mean something is blocking the packets back to the analogue phone? |
| 15:15.52 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 15:15.55 | Assid | heya |
| 15:18.17 | Assid | ok.. need some suggestion here.. i currently have 2 asterisk servers.. both are interconnected via vpn. 1 of which is onsite and the other is hosted in the cloud. The onsite server has local users. The roaming users can also connect directly to this server. OR i could have them connect to the cloud based server.. and use IAX to interconnect |
| 15:18.19 | gerritfromsa | lnb, or something has gone faulty |
| 15:18.43 | *** join/#asterisk simonmox (~simonmox@5.133.168.42) |
| 15:18.55 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 15:19.11 | lnb | gerritfromsa: i have a wireless extension to the analogue phone, dialed out with it. same results. no sounds period |
| 15:19.27 | Greenlight | Assid: If you have the bandwidth on site, then why overcomplicate things with the offsite server? |
| 15:19.29 | *** join/#asterisk teeteewhy (~teeteewhy@no.ra.pe) |
| 15:19.36 | simonmox | Hey folks, any pointers where I should luck if Asterisk 11.5.1 bombs out when it uses > 3.5Gb |
| 15:20.02 | gerritfromsa | you means a cordless phone connected to an ATA? |
| 15:20.19 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 15:20.34 | Greenlight | simonmox: You'll need to be a bit more specific. "bombs out" ? |
| 15:20.55 | gerritfromsa | simonmox, 3.5GB of RAM? |
| 15:20.57 | Greenlight | And I'm guessing that number is memory usage? |
| 15:21.21 | Greenlight | I would suspect that the memory usage is another *symptom* and not the actual *cause* |
| 15:21.31 | simonmox | Sorry for not being clear. Yes, when the user memory hits 3.5Gb, the asterisk process crashes. |
| 15:21.42 | gerritfromsa | +1 |
| 15:21.42 | Assid | Greenlight: technically the onsite server doesnt have "THAT MUCH" bandwidth .. but on the other hand all the roaming users end up speaking with the onsite users only.. not with other roaming users |
| 15:21.50 | dym | Anyone well with SNOM and their telephone firmware? I have a SNOME 320 with 8.7.3.19 installed and on EVERY SINGLE change i try to apply i get a Security Warning and im logged out. I cant reset the phone, apply the current firmware again - nothing. |
| 15:22.20 | Greenlight | dym: My SNOM 300 on my desk is working okay with latest firmware... |
| 15:22.42 | dym | Greenlight: Thanks. Im happy to hear that :) |
| 15:22.44 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:22.50 | Greenlight | simonmox: And, what exactly are you *doing* when it "crashes" (we talking segfault here?) |
| 15:23.03 | lnb | gerritfromsa: the analogue (panasonic) can have up to 3 or 4 wireless (cordless extensions) |
| 15:23.14 | Greenlight | dym: Well it doesn't fix your issue though ;S |
| 15:23.38 | dym | Greenlight: Well spotted, sherlock! :D |
| 15:24.05 | gerritfromsa | So it's a base VOIP base station which supports 3-4 dect handsets than |
| 15:24.13 | lnb | gerritfromsa: no |
| 15:24.21 | Greenlight | No problems my dear Watson. |
| 15:24.50 | lnb | gerritfromsa: its a panasonic analogue phone -> Grandstream HT502 ATA -> freepbx server -> ITSP |
| 15:24.59 | simonmox | From the crash dumps it's confbridge that triggers the crash. From what I can understand it tries to destroy a channel that doesn't exist. |
| 15:25.10 | simonmox | The comment in the code: /* Try to allocate memory for a new conference bridge, if we fail... this won't end well. */ pointed to a memory issue |
| 15:25.13 | gerritfromsa | lnb, what is wireless ? |
| 15:25.28 | Greenlight | simonmox: Hmm.. what version again? |
| 15:25.35 | lnb | gerritfromsa: the cordless extensions |
| 15:25.43 | simonmox | I checked munin, which we have running and at every point asterisk segfaulted user memory just hit 3.5Gb |
| 15:25.48 | simonmox | It's 10.5.1 |
| 15:26.04 | gerritfromsa | have you tried a pot (Plain old Telephone) instead ? |
| 15:26.23 | Greenlight | simonmox: How much memory do you have on the system ? |
| 15:26.23 | lnb | gerritfromsa: no, but i will try it now |
| 15:26.30 | simonmox | 16Gb in the system. |
| 15:26.42 | Greenlight | Assuming 64bit ? |
| 15:26.59 | simonmox | Yes 64bit |
| 15:27.09 | simonmox | One sec, I'll upload the memory chart. |
| 15:27.20 | Greenlight | First thing I'd advise is to upgrade to the latest 10 build |
| 15:27.51 | gerritfromsa | dym, did you set an admin password? |
| 15:28.18 | gerritfromsa | brb |
| 15:28.27 | Greenlight | I'd be interested to find out what's consuming all that memory, it looks like the segfault is again a *symptom* of perhaps not being able to allocate any more memory |
| 15:29.00 | Greenlight | I know there were *issues* with ConfBridge, specifically ConfBridgeRecord (if memory serves) but those were fixed some months back |
| 15:29.23 | simonmox | yeah, that's what I figured. Strange it was exactly at 3.5Gb. |
| 15:29.25 | simonmox | https://www.dropbox.com/s/3twflnkrw9cacgj/memory-week.png |
| 15:29.56 | simonmox | That's the memory chart for the past 7 days. We had segfaults on 25th, 28th and 29th. |
| 15:30.18 | Greenlight | Yea I can see the spikes |
| 15:31.08 | Greenlight | My gut feeling is that this isn't a memory or memory leak issue, and that's merely another symptom. What we really need to see is what happened to cause it start to ramp up towards that |
| 15:31.17 | lnb | gerritfromsa: worked once, second call no sound. maybe a jack issue |
| 15:31.20 | simonmox | Segaylt seemed to be triggered by users leaving the confbridge, specifically calling destroy_conference_bridge function. I couldn't figure out where this function was being called from though. |
| 15:31.23 | Greenlight | eg, what happened *before* the actual crash |
| 15:31.45 | Greenlight | First things first, upgrade to latest build. |
| 15:32.00 | Greenlight | Are you doing any confbrdige recording ? |
| 15:32.12 | *** join/#asterisk LieutPants (~LieutPant@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:32.12 | *** mode/#asterisk [+o LieutPants] by ChanServ |
| 15:32.17 | simonmox | Yes, we are recording. |
| 15:32.44 | Greenlight | Using ConfBridgeRecord ? |
| 15:33.21 | simonmox | I think so, we just pass the record param in confbridge.conf - It uses mixmonitor |
| 15:33.23 | simonmox | ? |
| 15:33.27 | Greenlight | Ok, now, 100% definetly upgrade. There have been known issues around this. |
| 15:33.39 | simonmox | Okay, will do. Thanks a lot for the help. |
| 15:33.47 | *** join/#asterisk LieutPants (~LieutPant@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:33.47 | *** mode/#asterisk [+o LieutPants] by ChanServ |
| 15:38.22 | Ice_Strike | http://pastebin.com/PSHfiBzd |
| 15:38.23 | Ice_Strike | What is this? |
| 15:38.27 | Ice_Strike | Getting flooded |
| 15:38.42 | Ice_Strike | with sip set debug on |
| 15:38.49 | ghost75 | friendly scanner :> |
| 15:40.17 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:40.56 | Ice_Strike | Thats better |
| 15:40.59 | Ice_Strike | iptables -A INPUT -s 182.151.213.1xx -j DROP |
| 15:41.00 | Ice_Strike | lol |
| 15:50.22 | elred_ | <gerritfromsa> elred_, Yes you can stream , it's described in the sample musiconhold.conf - check for [ulawstream] |
| 15:50.22 | elred_ | <gerritfromsa> You can use streamplayer or mpg123 |
| 15:50.41 | elred_ | that force me to put on a media streaming server when i want to have shared moh |
| 15:50.56 | elred_ | not possible to that in asterisk you mean ? External requirement , |
| 15:55.12 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 15:56.31 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:56.31 | *** join/#asterisk CrashHD (~na@68-67-80-153.wavecable.com) |
| 15:58.49 | *** join/#asterisk Defraz (~Defraz@24-116-129-19.cpe.cableone.net) |
| 15:59.28 | lnb | anyone know jack wiring? |
| 16:00.13 | lnb | sees the jack analogue phone is plugged into has only one blue with white band colored wire screwed in one side |
| 16:00.33 | lnb | there are 3 more wires not attached to the jack |
| 16:00.59 | drmessano | lol |
| 16:01.08 | lnb | it aint funny! |
| 16:01.10 | lnb | heh |
| 16:01.12 | drmessano | so connect the white/blue to the other terminal |
| 16:01.22 | drmessano | Sorry |
| 16:01.23 | drmessano | Blue |
| 16:01.33 | drmessano | Blue/white + Blue is a pair |
| 16:01.44 | lnb | its not solid blue |
| 16:01.52 | drmessano | What colors are there |
| 16:01.58 | lnb | has white rings every so often |
| 16:01.59 | drmessano | Im not gonna guess |
| 16:02.06 | lnb | let me take a picture and upload it |
| 16:02.16 | drmessano | You cant just name the 3 other colors? |
| 16:02.30 | [TK]D-Fender | Tom! |
| 16:02.31 | [TK]D-Fender | Dick! |
| 16:02.33 | [TK]D-Fender | Harry! |
| 16:02.36 | drmessano | lol |
| 16:02.42 | drmessano | *list |
| 16:03.15 | drmessano | Im sure the pic will be harder to discern than just listing the other 3 wires, but typing is too hard |
| 16:04.14 | lnb | i am not near that jack... one sec i will load pic on pc |
| 16:04.59 | *** join/#asterisk zpotoloom (~tom@tom.data.ee) |
| 16:07.07 | lnb | drmessano: there is a white/orange rings, orange/white rings, white/blue rings unattached |
| 16:07.17 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 16:07.44 | drmessano | blue is your missing wire |
| 16:07.50 | drmessano | Sounds like its not there |
| 16:07.56 | drmessano | Clipped off maybe |
| 16:08.34 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:09.23 | *** join/#asterisk serafie (~erin@nat/digium/x-xokqhsbkxolbbuyb) |
| 16:09.30 | *** join/#asterisk serafie1 (~erin@nat/digium/x-dwklegayiwrzhgoh) |
| 16:09.37 | lnb | sold blue? |
| 16:10.13 | ChannelZ-Wk | Lord I am listening to the loudest MOH evar |
| 16:10.34 | ChannelZ-Wk | And it's lute music or something |
| 16:14.43 | lnb | drmessano: http://www.servaris.com/images/voip/broken_wires.jpg |
| 16:15.17 | lnb | its the jack on the left side that is borked |
| 16:15.37 | drmessano | That white/blue goes with your blue/white |
| 16:15.41 | lnb | there is a blue/white rings wire attached (kind of blurry) |
| 16:15.59 | lnb | on same screw? |
| 16:16.08 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 16:16.08 | drmessano | No, the one across from it |
| 16:16.12 | lnb | ok |
| 16:16.14 | lnb | i go try |
| 16:16.21 | lnb | hopefully i wont blow the house up |
| 16:16.24 | *** join/#asterisk fatemeblue (~fateme@static.94.244.47.78.clients.your-server.de) |
| 16:19.02 | fatemeblue | can any body help me, i have install asterisk on centos on vmware and configure sip.conf but my account faild to register with 408 error |
| 16:20.52 | [TK]D-Fender | fatemeblue: "sip set debug on" <- * CLI |
| 16:20.58 | [TK]D-Fender | fatemeblue: Do you see the packets arrive? |
| 16:21.28 | fatemeblue | i set debug on |
| 16:21.43 | fatemeblue | there is lot of thing written there |
| 16:21.53 | [TK]D-Fender | PASTEBIN <- |
| 16:21.55 | [TK]D-Fender | ~pb |
| 16:21.55 | 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. |
| 16:21.56 | fatemeblue | it got confuse |
| 16:21.57 | [TK]D-Fender | ^^^ |
| 16:23.19 | fatemeblue | i mean the log file is lorf |
| 16:23.21 | fatemeblue | lorg |
| 16:24.23 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:24.36 | [TK]D-Fender | I am not talking about log files. |
| 16:24.41 | [TK]D-Fender | Use Asterisk CLI live for this |
| 16:27.03 | fatemeblue | aha |
| 16:27.08 | fatemeblue | ok i enable it |
| 16:27.24 | fatemeblue | and which command should i use to see packet |
| 16:27.59 | [TK]D-Fender | [12:20][TK]D-Fenderfatemeblue: "sip set debug on" <- * CLI |
| 16:33.07 | fatemeblue | yes i enable sip debug |
| 16:33.19 | fatemeblue | how should i check packet arrive |
| 16:33.33 | *** join/#asterisk JamKo (~JamKo@unaffiliated/jamko) |
| 16:35.02 | LooserOuting | i get this error: ERROR[2298]: res_config_mysql.c:224 find_table: Failed to query database 'asterisk', table 'sipfriends' columns: Can't create/write to file '/var/tmp/mysql.mpmNyU/#sql_6c5d_0.MAI' (Errcode: 2) |
| 16:35.39 | LooserOuting | asterisk uses the table. i can register and make calls |
| 16:36.01 | LooserOuting | this is asterisk 11.2.1 |
| 16:36.24 | JamKo | Greetings. Does anyone know a good reason why Asterisk has not been patched in 1.8 to generate a series of T.38 no-signal packets to the destination media server, after the 200 ok with session description? |
| 16:37.01 | JamKo | Currently it just sends 1 T.38 no signal after the 200, which is not enought for most SBC and other media gateways. |
| 16:37.36 | WIMPy | Welcome to the SIP compatibility lottery. |
| 16:37.41 | JamKo | I have it patched to send a flurry of t.38 no-signals after the 200, and resolves most issues. |
| 16:38.13 | JamKo | I have seen this issue mentioned more than once in the bug posts, but it never gets accepted. Reason given, Asterisk shouldn't have to do that. |
| 16:39.07 | JamKo | Essentially saying everything else is broken. So if we have a fix to make it more cross platform compatible, why not implement a patch? |
| 16:39.50 | LooserOuting | it's simpler to maintain a software if you don'T have a lot of workarounds in it |
| 16:40.17 | LooserOuting | if the problem is on th other side then they shold fix it |
| 16:40.21 | WIMPy | It's simpler if you can't use it. |
| 16:42.16 | LooserOuting | don't get it wrong i am no developer. But let me ask another questions. Why don't you talk to the manufacturer ? |
| 16:42.28 | JamKo | It's been awhile. I wouldn't hold my breath on the others resolving the issue at this point. What's the big deal? Asterisk has a bunch of workaround configs for poor NAT setups, ALGs etc. |
| 16:42.40 | JamKo | Why not on T38? |
| 16:42.48 | JamKo | It's just like a keep alive. |
| 16:42.50 | JamKo | easy. |
| 16:43.29 | *** join/#asterisk Alex_h (~AlexHold@178.78.119.76) |
| 16:44.18 | *** join/#asterisk Defraz (~Defraz@209.141.122.3) |
| 16:44.23 | LooserOuting | the thing is: they can do it. but they must not. and the manufacturer definitly have to do it. |
| 16:44.46 | LooserOuting | sorry bad english |
| 16:44.57 | JamKo | To clarify I'm speaking about t38 passthrough setups in asterisk. |
| 16:45.07 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 16:45.16 | LooserOuting | the thing is: they can do it. but they could not. and the manufacturer definitly have to do it. |
| 16:46.14 | JamKo | I have Level3 capturing from their end on the sbc media gateway. Maybe they will have the leverage to force a patch, but I certainly don't. lol. |
| 16:46.19 | lnb | same crap |
| 16:46.22 | LooserOuting | i don't understand the problem in detail. |
| 16:46.30 | lnb | can dial/get call but no sound |
| 16:46.43 | lnb | i think the jack is borked |
| 16:47.37 | Alex_h | weird problem with ChanSpy, had this working previously, chanspy launches but then never attaches to the channel being spied on. Asterisk logs show the launch then nothing, in my test lab i see the chanspy app launch then attach, i did a diff on the config files from a known working backup and cannot see a difference, anyone ever had this? |
| 16:48.32 | *** part/#asterisk polysics (~Adium@host176-66-dynamic.4-87-r.retail.telecomitalia.it) |
| 16:48.34 | Alex_h | phones being used are in correct contexts, targeting correct channels/extensions etc |
| 16:48.41 | Penguin | alex_h: Different asterisk versions? |
| 16:48.52 | Alex_h | nope same |
| 16:51.48 | JamKo | LooserOuting: When asterisk sets up a T.38 passthrough session, it will send T.38 no-signal packets from it's original source rtp port while everyone gets their 200s in line. After the 200s, and the new T38 media ports have all |
| 16:52.03 | LooserOuting | JamKo: refering to t38 rec. '"No signal" indicator may be sent whenever there is no signal in TDM input. For example, it may |
| 16:52.03 | LooserOuting | be used when the modem is changed from [ITU-T V.21] to [ITU-T V.17], or from [ITU-T V.17] to |
| 16:52.03 | LooserOuting | [ITU-T V.21] one.' |
| 16:52.52 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:52.57 | JamKo | been negotiated, asterisk sends 1 solitary T.38 no-signal from it's new t38 port, to the negotiated port to whereever it is sending the call. Most SBS, Sonus appliances need more than one packet recevied |
| 16:53.12 | JamKo | in order to keep the session open. Usually the one packet results in a reinvite back to voice from the carriers. |
| 16:53.42 | Penguin | jamko: its |
| 16:54.26 | LooserOuting | Did you see the "MAY" in my post ? |
| 16:55.17 | LooserOuting | JamKo: I didn't read the hole document but it doesn't look like asterisk is doing something wrong |
| 16:56.03 | JamKo | Right it's not doing anything wrong, but it's also not working with the standard implentations used by large carriers. |
| 16:56.48 | LooserOuting | JamKo: that's not true. i am working with big carries too. we dont't have this problem. |
| 16:57.55 | WIMPy | Did I mention the word "lottery"? |
| 16:58.08 | JamKo | Asterisk not sending NAT keep alives would also be considered "not doing something wrong" but we all know some setups need it. |
| 16:58.13 | LooserOuting | what is in the jackpot ? |
| 16:58.21 | WIMPy | There are some bigger carriers where you can't even make voice calls with Asterisk. That's the SIP world. |
| 16:58.26 | JamKo | LooserOuting: What version may I ask? |
| 16:58.46 | *** join/#asterisk monsterco (~monsterco@bas6-toronto47-1176197537.dsl.bell.ca) |
| 16:59.25 | LooserOuting | we used all branches |
| 16:59.45 | LooserOuting | but right now we are using the 11 branch |
| 16:59.57 | *** part/#asterisk monsterco (~monsterco@bas6-toronto47-1176197537.dsl.bell.ca) |
| 17:02.40 | LooserOuting | We all know that large carriers won't make updates if these aren't it's notvery important. also they take month for that. please don'T get me wrong but that's your problem :-) |
| 17:03.21 | LooserOuting | we got some of these ourselves. |
| 17:03.43 | LooserOuting | most does't support V.34 |
| 17:04.18 | JamKo | Of course it is, and it's patched on my side to work. Just thought it would be nice for asterisk users to have some greater T.38 interoperability built into Asterisk. |
| 17:04.18 | LooserOuting | but that's another story |
| 17:05.32 | JamKo | Asterisk also sends 3 no-signals from it's original rtp port in the 10-20k range, before it switches over to it's new 4K port. That's questionable behavior. Why send 3 from the wrong port, and then only 1 from the correct port? |
| 17:06.23 | drmessano | lnb: dial out but no sound is not a jack issue.. It's passing audio just fine |
| 17:06.37 | *** join/#asterisk flapjacks (~flapjacks@wsip-184-183-148-254.ph.ph.cox.net) |
| 17:07.08 | drmessano | lnb, if it was borked, you wouldn't have a dial tone, wouldn't be able to dial out. Wrong variable |
| 17:10.03 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 17:10.31 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 17:11.40 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:16.57 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:204d:6287:4c21:c6c9) |
| 17:22.11 | *** join/#asterisk monsterco (~monsterco@bas6-toronto47-1176197537.dsl.bell.ca) |
| 17:22.17 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:23.05 | monsterco | <PROTECTED> |
| 17:28.32 | gerritfromsa | Why not use the fax machine thats already connected to the ATAs to fax out? |
| 17:32.26 | *** join/#asterisk charkee (~charkee@122.55.36.17) |
| 17:32.27 | gerritfromsa | elred_, was wondering if you can't map a network drive and use the symbolic link from each amchine to that drive ? |
| 17:33.10 | gerritfromsa | lnb, have you tried the 2nd port of the ATA ? Does it do the same ? |
| 17:33.40 | gerritfromsa | brb |
| 17:41.05 | *** join/#asterisk eduzimrs (~eduzimrs@mail.aytycrm.com.br) |
| 17:45.38 | monsterco | gerritfromsa - fax machine not available on all floors - also user finds it easier to use computer |
| 17:46.09 | monsterco | do you know of any third party web portals or email fax modules for Asterisk? |
| 17:48.13 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginm.net) |
| 17:49.37 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 17:50.43 | danfromuk | Hi. After giving up with misdn and freepbx, i've decided to try to build it manually like i'm used to. I'm trying to choose chan_misdn from menuselect, but its missing some dependants. |
| 17:50.58 | danfromuk | Where does isdnnet comes from? and also suppserv? |
| 17:51.52 | WIMPy | You're still using the abandoned version of mISDN? |
| 17:52.15 | danfromuk | No. I downloaded it from misdn.eu |
| 17:52.57 | WIMPy | That (or the one in the standard kernel) doesn't work with chan_misdn. |
| 17:56.05 | danfromuk | Where should I look for another version? The manufacturer's misdn is corrupt. |
| 17:56.20 | *** join/#asterisk zafu (~pif@84-74-26-25.dclient.hispeed.ch) |
| 17:56.26 | *** join/#asterisk serafie (~erin@nat/digium/x-aeucfhdrcywljydk) |
| 17:56.35 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 17:56.47 | zafu | hi, is something wrong with this? exten => _0Z.,n,GoToIf($["${CALLERID(num):0:3}" = "+41"]?0041${EXTEN:1},1) |
| 17:57.11 | WIMPy | Well, it's senseless. |
| 17:57.26 | WIMPy | No it isn;t. Forget what I said. |
| 17:57.42 | Alex_h | its missing a : after the 1st label |
| 17:57.57 | zafu | even is label is empty : is required? |
| 17:58.01 | Alex_h | yes |
| 17:58.03 | zafu | aha |
| 17:58.48 | danfromuk | WIMPy: should i be looking for misdn1 or 2? |
| 17:58.57 | Alex_h | so if it resolves to true, it would be.... exten => _0Z.,n,GoToIf($["${CALLERID(num):0:3}" = "+41"]?0041${EXTEN:1},1:) |
| 17:59.04 | Penguin | No, the label is not required to have a : after it. |
| 17:59.09 | ghost75 | somebody using voipmonitor? |
| 17:59.10 | Alex_h | oh |
| 17:59.13 | Alex_h | really? |
| 17:59.15 | zafu | Penguin: yes |
| 17:59.23 | Penguin | But if you have a false label, you have to have the : before it. |
| 17:59.28 | WIMPy | danfromuk: I'd go for mISDN2 or dahdi. Depending on what features you need. |
| 17:59.43 | Alex_h | my mistake i believed it was for both empty false and true labels |
| 17:59.46 | zafu | my problem is if the GoToIf fails then the dial stops |
| 18:00.04 | Penguin | Show us. |
| 18:00.06 | Penguin | ~pb |
| 18:00.06 | 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. |
| 18:00.07 | zafu | I'd like the 'fail' case to just keep going |
| 18:00.20 | danfromuk | WIMPy: I can't get dahdi to recognise this card, so i'm trying misdn |
| 18:01.04 | Penguin | That's what the false case already doese. |
| 18:01.19 | Penguin | If you only specify :falselabel, then that is what the true case does as well. |
| 18:01.52 | Penguin | If you do not specify the label for true or false, dialplan will keep going when the label isn't there. |
| 18:01.59 | zafu | my aim is the rewrite an outoing number else keep going in the dialplan |
| 18:02.10 | Penguin | pastebin |
| 18:02.12 | zafu | yep |
| 18:03.33 | zafu | http://pastebin.com/T4ka9EeC |
| 18:04.53 | Penguin | Where's the rest of it? |
| 18:05.04 | Penguin | I want to see your dial plan. |
| 18:05.12 | zafu | ok |
| 18:05.31 | Penguin | You already said it fails. I need to see the dial plan to know why it fails... so you can fix it. |
| 18:10.57 | fatemeblue | how should i check packet arrive in * cli> afte enabling sip debug |
| 18:16.26 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginm.net) |
| 18:17.40 | zafu | ah, $EXTEN is no longer valid, now it's ${EXTEN}Â ? |
| 18:18.26 | WIMPy | It has always been that way. |
| 18:18.29 | [TK]D-Fender | it was never valid |
| 18:18.38 | zafu | *blush* |
| 18:18.49 | [TK]D-Fender | ${variableORfucntion()} |
| 18:19.14 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 18:21.17 | danfromuk | WIMPy: On misdn.eu, it says "From Linux Kernel 2.6.27 on, mISDN v2 will be already included by kernel.org". That seems to imply that its already installed on centos 6.4. But asterisk menuselect lists it as uninstalled. |
| 18:21.45 | danfromuk | I can't seem to get it to install, even following quite a few guides. some of which are out of date. |
| 18:26.23 | *** join/#asterisk eagles0513875 (eagles0513@gateway/shell/trekweb.org/x-tepcrkfhdwgudbhi) |
| 18:26.50 | eagles0513875 | can Asterisk do video and audio conferencing? |
| 18:29.22 | [TK]D-Fender | eagles0513875: follow-the-speaker only |
| 18:29.28 | [TK]D-Fender | eagles0513875: or on-on-one call |
| 18:31.30 | eagles0513875 | ok :-/ |
| 18:31.38 | eagles0513875 | thanks [TK]D-Fender |
| 18:32.46 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:34.56 | Penguin | danfromuk: zgrep MISDN /proc/config.gz |
| 18:35.40 | WIMPy | danfromuk: I already told you that it's not supported by chan_misdn. |
| 18:36.04 | WIMPy | It's also explained on the link I gave. |
| 18:37.01 | danfromuk | Sorry, i noticed that. However, even after i download and installed misdn, its still not allowing me to select it. |
| 18:37.17 | danfromuk | I think this one is actually going to make me cry if i dont get it sorted soon. |
| 18:37.47 | WIMPy | The old one? |
| 18:38.20 | danfromuk | Penguin: gzip: /proc/config.gz: No such file or directory |
| 18:38.37 | danfromuk | the only one thats available from misdn.org |
| 18:38.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.95) |
| 18:38.59 | danfromuk | thats the only one that i can find that actually doesnt show up with any errors |
| 18:39.31 | WIMPy | Yes, I think the old one has vanished. |
| 18:39.44 | danfromuk | Is misdn not available on yum repo? |
| 18:39.57 | WIMPy | But you can use the version from the Digium repositories. |
| 18:40.01 | danfromuk | WIMPy: correct, any links to the old one take me to do new one |
| 18:40.40 | danfromuk | Tried that. I get this http://pastebin.com/uBPvJuVF |
| 18:40.45 | WIMPy | Or the version by irroot. |
| 18:41.13 | danfromuk | Whats irroot? |
| 18:41.28 | WIMPy | who |
| 18:41.38 | WIMPy | He used to hang around here quite a lot. |
| 18:42.21 | WIMPy | But as I said before, I'd go with v2 and LCR or dahdi. |
| 18:42.47 | WIMPy | Is it a card with a CCD chip or something else? |
| 18:43.17 | danfromuk | I'm currently trying to do this with asterisk 1.8. Do you think i'd have the same problem with any other version? |
| 18:43.24 | danfromuk | No idea. its this |
| 18:43.25 | danfromuk | http://quadbri.phoniceq.com/ |
| 18:43.40 | WIMPy | The Asterisk version shouldn't make much of a difference. |
| 18:44.05 | danfromuk | I offered to replace a client's asterisk 1.4 with a freepbx install. now they've been offline for abiout 7 hours and i cant get it back up. can't see my kids till its sorted. |
| 18:44.18 | danfromuk | So i know it works with 1.4 (and the right files) |
| 18:44.40 | WIMPy | Not the best picture, but I think there is the Cologne Cathedran on there. So it should work with any driver. |
| 18:45.47 | WIMPy | So you need to have it running with FreePBX? |
| 18:46.37 | danfromuk | Yes. But for now i've given up on freepbx, and just trying it the traditional way. I downloaded the asterisk source files to try to build manually. |
| 18:46.44 | danfromuk | which is when i found that misdn was unavailable |
| 18:47.03 | mjordan | keep in mind that mISDN requires a rather old version of linux |
| 18:47.16 | mjordan | what are you trying to build it on? |
| 18:47.18 | danfromuk | it is a cologne. i think dahdi reported that |
| 18:47.35 | danfromuk | centos 6.4 which came with the freepbx install |
| 18:47.46 | mjordan | yup. That won't work. |
| 18:47.47 | WIMPy | Yes, I have no Idea what the latest supported Linux for the old version might be. But it did work on 3.x versions. |
| 18:48.07 | danfromuk | argh. |
| 18:48.09 | danfromuk | ok. |
| 18:48.49 | danfromuk | Ok, so to review.... Dahdi won't work because the driver requires is qozap which requires zaptel and some other stuff which i couldnt get working. |
| 18:49.05 | danfromuk | and misdn wont work with this version of linux |
| 18:49.11 | WIMPy | Did you read the link I gave you? |
| 18:49.26 | danfromuk | I did. Let me read again. |
| 18:49.40 | WIMPy | With dahdi you use dahdi_hfcs. |
| 18:49.45 | danfromuk | You told me to forget it |
| 18:49.57 | WIMPy | And with mISDN you use LCR and chan_lcr. |
| 18:50.19 | WIMPy | I'd forget about the old mISDN, just like the developers did some years ago. |
| 18:50.44 | lnb | drmessano: you here? |
| 18:50.45 | danfromuk | So i should be trying to get dadhi working? |
| 18:51.09 | WIMPy | Should be easy to patch. |
| 18:51.26 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-30.usfamily.net) |
| 18:51.35 | WIMPy | I usually prefer LCR, but as I said in the beginning, it depends on the features you want/need. |
| 18:52.02 | danfromuk | features? at the moment, i'll settle for making and receiving calls :-) |
| 18:52.12 | danfromuk | When i try to use dadhi, i get "driver should be 'qozap' but is actually 'hfc4s8s_l1'" |
| 18:52.40 | danfromuk | the manufacturer only provides this http://quadbri.phoniceq.com/driver/bristuff/ |
| 18:52.50 | WIMPy | Err. What is that? Is that a name from mISDN v1? |
| 18:52.53 | drmessano | lnb, for the moment |
| 18:53.31 | WIMPy | If you have a recent kernel, modprobe hfcmulti. |
| 18:54.12 | danfromuk | Did have hfcmulti at one point today. |
| 18:54.15 | lnb | bought a new jack. tried a few combo's but no dial tone. do you know which colors on the jack the blue/white/blue bands go to? |
| 18:54.25 | danfromuk | just rebooting the server |
| 18:54.43 | WIMPy | wonders how many competing drivers there might be on that system by now. |
| 18:55.16 | lnb | on this new jack there is yellow, green, red, black |
| 18:55.21 | WIMPy | Try to load either hfcmulti or wcb4xxp. |
| 18:56.19 | danfromuk | wcb4xxp is blacklisted at the moment |
| 18:56.46 | danfromuk | in dahdi.blacklist.conf |
| 18:56.55 | *** join/#asterisk acidfoo (~nib@unaffiliated/acidmen) |
| 18:57.16 | danfromuk | i think i called a script that did that. commented it out. |
| 18:57.58 | *** join/#asterisk Jamuel (~Adium@c-67-180-156-186.hsd1.ca.comcast.net) |
| 18:58.05 | danfromuk | ok, still the same hfc4s8s_l1 |
| 18:58.16 | WIMPy | Might be a good idea to first make sure nothing is loaded. |
| 18:58.33 | WIMPy | rmmod that and then try the others. |
| 18:59.02 | WIMPy | Where does it come from? Did you build it or are you on a system that still has the old stuff? |
| 18:59.46 | danfromuk | I didnt build it. It could have come with freepbx i suppose. its a fresh install of freepbx |
| 19:00.30 | danfromuk | do i need to reboot each time? |
| 19:00.55 | WIMPy | I have no clue what it would/could do regarding your systems drivers/utilities (packages). |
| 19:01.07 | WIMPy | Probably not. |
| 19:01.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.124) |
| 19:01.39 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginm.net) |
| 19:01.52 | WIMPy | I've had issues with cards not being initialized propperly, but for the moment that doesn't matter. We just want to know which driver will recognize it. |
| 19:02.53 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.125.124) |
| 19:02.56 | danfromuk | weird. after rmmod hfc4s8s_l1, i did a reboot, and it still showing up |
| 19:03.01 | *** part/#asterisk fatemeblue (~fateme@static.94.244.47.78.clients.your-server.de) |
| 19:03.16 | WIMPy | Sure. |
| 19:03.17 | *** join/#asterisk fatemeblue (~fateme@static.94.244.47.78.clients.your-server.de) |
| 19:03.27 | danfromuk | sure? |
| 19:03.33 | WIMPy | It's just like killing a task. |
| 19:03.39 | danfromuk | oh |
| 19:03.45 | danfromuk | ok, 1sec |
| 19:04.00 | fatemeblue | any know what is the problem of unmonitorid sip user? |
| 19:04.02 | WIMPy | If you don;t remove (or balacklist) it from your boot scrips it will come back. |
| 19:04.43 | WIMPy | But for the moment it should be fine if you just temporarily rmmod it and try to modprobe the others. |
| 19:05.27 | danfromuk | ok, this is what i get now |
| 19:05.29 | danfromuk | http://pastebin.com/BXB6nti6 |
| 19:05.40 | WIMPy | Ypu probably need to remove the package that contains the old stuff and replace it with the current modules and tools. |
| 19:06.50 | WIMPy | >>Try to load either hfcmulti or wcb4xxp. |
| 19:07.07 | WIMPy | Just do it manually for now |
| 19:08.24 | drmessano | lnb, green and red is line 1 on traditional phone jacks |
| 19:09.04 | drmessano | lnb, your blue/white pair would be wired to green/red. One wire to each |
| 19:09.17 | lnb | ok thanks |
| 19:09.26 | danfromuk | No good. Same result. http://pastebin.com/keFxdfKj |
| 19:09.41 | lnb | i did take the pots phone direct to ATA and same thing there... no sound.. which is what, no rtp? |
| 19:09.56 | lnb | one one fxs port |
| 19:10.02 | *** part/#asterisk fatemeblue (~fateme@static.94.244.47.78.clients.your-server.de) |
| 19:10.04 | lnb | the other fxs port is good |
| 19:10.14 | drmessano | NAT settings perhaps |
| 19:10.21 | WIMPy | >>Try to load either "modprobe hfcmulti" or "modprobe wcb4xxp". |
| 19:10.57 | lnb | nat on $ext_if from !($ext_if) to any -> ($ext_if) static-port |
| 19:11.18 | danfromuk | dahdi_hardware says this btw http://pastebin.com/RVnHPzdR |
| 19:12.00 | lnb | both fxs ports go to same freepbx server, just different sip /rtp ports |
| 19:12.02 | WIMPy | "modprobe hfcmulti", "modprobe wcb4xxp" |
| 19:13.03 | runfromnowhere | So is it possible to get behavior similar to Audiohook with Monitor, where a call will be followed from transfer to transfer in a single recording file? |
| 19:13.10 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 19:13.11 | danfromuk | WIMPy: http://pastebin.com/iavki0jr |
| 19:13.38 | drmessano | lnb, why would you change the port range? |
| 19:14.04 | drmessano | lnb, port range is negotiated by the client and has no relation to the ports used on the server |
| 19:14.33 | WIMPy | Ok, so dahdi won't recognize the ID. |
| 19:14.40 | WIMPy | Try "misdn_info" after loading hfcmulti. |
| 19:15.39 | danfromuk | That did something http://pastebin.com/4vLZ1SYp |
| 19:16.04 | WIMPy | Ok, so that driver is working. |
| 19:16.38 | WIMPy | So either you patch wcb4xxp to recognize the PCI ID of your card or you install LCR and use that. |
| 19:16.53 | *** part/#asterisk monsterco (~monsterco@bas6-toronto47-1176197537.dsl.bell.ca) |
| 19:17.25 | danfromuk | One sec. I'm currently using hfcmulti which can recognise the card. not wcp4xxp. |
| 19:17.42 | WIMPy | Exactely. |
| 19:17.47 | danfromuk | Is there a problem with using hfcmulti if it can recognise the card? |
| 19:18.04 | WIMPy | No. |
| 19:18.23 | WIMPy | But you need LCR to use it and then connect to Asterisk. |
| 19:18.27 | Penguin | How can you be using a 2.6 or higher kernel and not have /proc/config.gz? |
| 19:19.04 | WIMPy | Because a (insert some swear word) configured it. |
| 19:19.27 | lnb | drmessano: we were having issues with faxing before, it was suggested to make sure on the server the ports were 506x and rtp 10k-15k |
| 19:19.38 | WIMPy | danfromuk: Do you use any port in NT mode or all as TE? |
| 19:20.02 | danfromuk | Two connect to the isdn provider. One connects to a fax server. |
| 19:20.40 | WIMPy | Hmm. Has that ever been stable with mISDN1? |
| 19:21.19 | WIMPy | If you use LCR you can route those calls directely in the kernel without Asterisk. |
| 19:21.21 | drmessano | lnb, source ports != destination ports |
| 19:21.41 | lnb | drmessano: changed it... now 300/300 99.238.64.55 D N A 5060 OK (23 ms) |
| 19:21.53 | danfromuk | The documentation seems to say that LCR requires misdn |
| 19:22.19 | drmessano | lnb, if this is a two port ATA, the listening ports should typically be 5060 on Port 1 and 5061 on Port 2 |
| 19:22.43 | drmessano | lnb, which is the factory default |
| 19:23.13 | drmessano | lnb, and the rtp ports are pretty irrelevant |
| 19:23.14 | WIMPy | You have mISDN running. |
| 19:23.14 | lnb | wll |
| 19:23.34 | lnb | well ... dial out, no ring tone, but cell phone rings |
| 19:23.35 | WIMPy | hfcmulti is the mISDN2 driver. |
| 19:23.43 | danfromuk | Good point. I was getting mixed up with asterisk not seeing it. |
| 19:23.51 | drmessano | lnb, are both ports configured to 5060? |
| 19:24.10 | lnb | what do you mean both ports? |
| 19:24.13 | drmessano | Because that means both SIP clients on that ATA are using the same port, and thats not going to work |
| 19:24.21 | drmessano | lnb, both ports on the ATA |
| 19:24.26 | lnb | no no |
| 19:24.28 | drmessano | Your ATA config is HOSED |
| 19:24.48 | drmessano | If one port is working and one is not, you've changed something you should not have. |
| 19:24.53 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 19:25.26 | drmessano | lnb, beyond setting a proxy, user, pass, and removing some feature codes, there is no tweaking to make an ATA "work" |
| 19:25.56 | lnb | drmessano: fxs0 5063/10300, fxs1 (phone) 5060/10400 |
| 19:26.21 | drmessano | O.o |
| 19:26.35 | lnb | drmessano: the itsp was questioning why were fax would go direct to them, it would be a 55000+ ports |
| 19:26.40 | danfromuk | WIMPy: sorry about this. now i can't compile lcr. do you compile it from within the asterisk source directory? |
| 19:26.41 | danfromuk | : error: asterisk/buildopts.h: No such file or directory |
| 19:27.10 | drmessano | lnb, that has nothing to do with the source ports on the ATA you changed |
| 19:27.17 | WIMPy | No, from anywhere. |
| 19:27.25 | drmessano | [15:21:21] <drmessano> lnb, source ports != destination ports |
| 19:27.47 | WIMPy | Is Asterisk already installed? |
| 19:27.54 | danfromuk | Yes |
| 19:27.59 | lnb | drmessano: if you run pf nat, the way I showed you above, you can make source = destination |
| 19:28.23 | lnb | unless of course, the ports are already used on the remote server |
| 19:28.26 | WIMPy | That file should be in /usr/include/ |
| 19:28.29 | drmessano | lnb, which still has absolutely nothing to do with the source ports on the ATA |
| 19:28.42 | Penguin | Just because they can be the same doesn't really mean it's relevant. |
| 19:28.46 | lnb | well i put the local source port back to default (5062) |
| 19:28.55 | lnb | for fxs1 (analogue phone) |
| 19:29.07 | lnb | and on server its.... |
| 19:29.25 | danfromuk | WIMPy: its not there. |
| 19:29.40 | WIMPy | Is it anywhere? |
| 19:29.45 | danfromuk | probably because of freepbx messing things around. |
| 19:29.46 | danfromuk | 1sec |
| 19:29.57 | lnb | its not registering |
| 19:30.42 | drmessano | Source port has nothing to do with destination |
| 19:31.03 | danfromuk | find / -name buildopts.h returned nothing |
| 19:31.35 | WIMPy | So that Asterisk you have there is not the ode you built from source? |
| 19:31.40 | WIMPy | one |
| 19:31.56 | danfromuk | No, i never completed that rebuild because i couldnt select misdn. |
| 19:32.07 | danfromuk | ok, i'll rebuild. one second. |
| 19:32.07 | drmessano | lnb, then you have something else configured incorrectly. |
| 19:32.28 | WIMPy | Ah, then there's the usual -dev stuff missing from the install, I guess. |
| 19:32.34 | lnb | for sure.. just have to find it |
| 19:34.03 | drmessano | lnb, networking 101.. Most SIP endpoints do not use the same RTP range as what is configured on an Asterisk server. ALL multi-line SIP endpoints use source ports other than 5060 for second line and beyond. They all work without changing those two things |
| 19:34.24 | drmessano | So the notion that a device must share an RTP range with the Asterisk server and all use 5060 is a false one |
| 19:34.54 | Penguin | This one time, I set my asterisk rtp.conf range to the same range as my phones... |
| 19:35.16 | Penguin | But then I changed it back because I exhausted my asterisk rtp ports too quickly. |
| 19:36.54 | ngharo | why do all firewall setups tell to open 10000-20000 for RTP; instead couldn't use the sip_conntrack module? |
| 19:36.57 | ngharo | anyone try that? |
| 19:37.38 | WIMPy | ngharo: Sure. Just do it. But don't load the NAT module. |
| 19:38.06 | drmessano | ngharo, i've been told that newer firewalls seem to handle the RTP just fine, without forcing it. |
| 19:38.22 | ngharo | roger. I might try it later |
| 19:38.57 | lnb | drmessano: thank you for helping me with wiring |
| 19:39.04 | WIMPy | I've relied on the conntrack module for years without issues. |
| 19:39.16 | ngharo | WIMPy: good to know! |
| 19:39.48 | drmessano | WIMPy, that's awesome |
| 19:39.57 | drmessano | I've always "just opened them" |
| 19:40.38 | Penguin | Which nat module are you saying to explicitly not use? |
| 19:40.52 | WIMPy | nf_nat_sip |
| 19:41.22 | Penguin | nf_nat_sip 4464 0 |
| 19:41.29 | Penguin | Too late! It's loaded. |
| 19:41.35 | WIMPy | That does the "ALG" stuff and will probably cause major issues. |
| 19:41.37 | lnb | hmmm |
| 19:41.44 | lnb | fsx1 won't register |
| 19:42.27 | Penguin | I don't have a sip_conntrack module loaded, but I have an nf_conntrack_sip loaded. |
| 19:42.40 | WIMPy | That's the one. |
| 19:43.35 | WIMPy | The one that makes the RTP "RELATED" to SIP. |
| 19:44.05 | danfromuk | WIMPy: ok, i've installed from source, now i get a different make error http://pastebin.com/hUWUrYhb |
| 19:44.52 | WIMPy | What verison of Asterisk and LCR? |
| 19:45.01 | Penguin | I have four modules that are *_sip |
| 19:45.16 | drmessano | So the nf_conntrack_sip is the BAD one? |
| 19:45.21 | danfromuk | asterisk 1.8.23.1 and the latest version of lcr i think |
| 19:45.30 | Penguin | nf_nat_sip, nf_conntrack_sip, nf_nat, nf_conntrack |
| 19:45.31 | WIMPy | drmessano: no. |
| 19:45.42 | drmessano | ok.. nf_nat_sip is? |
| 19:45.49 | drmessano | and nf_conntrack_sip is good? |
| 19:45.50 | WIMPy | yes |
| 19:45.57 | WIMPy | exactely |
| 19:46.00 | Penguin | Well, I guess I should have said related to *_sip... I grepped for _sip. |
| 19:46.00 | drmessano | Ok, got it |
| 19:47.06 | WIMPy | danfromuk: Hmm. strange. Either the version detection has become broken or the detection failed for some other reason. Any hints when running configure? |
| 19:47.10 | Penguin | So if I have always had nf_nat_sip loaded and calls always work, seems like I should leave it as is. |
| 19:47.24 | danfromuk | WIMPy: configure was clean |
| 19:47.38 | WIMPy | Penguin: Does that box do NAT? |
| 19:47.43 | Penguin | Yes it does. |
| 19:48.04 | Penguin | It's the router for the voice network. |
| 19:48.33 | danfromuk | Do you think that i should start from scatch with a fresh install, without freepbx? Although the whole point of this exercise was to allow the client to manage their own callflow. |
| 19:48.40 | WIMPy | Well, maybe it's doing it right then. |
| 19:48.59 | Penguin | It's a Vyatta router, which is really just Debian. |
| 19:49.06 | drmessano | Penguin, are port forwarding the RTP ports at all? |
| 19:49.18 | Penguin | Yes, I forward 10000-20000. |
| 19:49.24 | drmessano | Then that is why |
| 19:49.38 | WIMPy | danfromuk: I'm not sure what happens, but it seems to try to buld chan_lcr for Asterisk<1.8. |
| 19:49.52 | drmessano | We're talking about having the firewall handle the RTP ports |
| 19:50.02 | Penguin | Oh, if I forward the ports, the kernel module doesn't do anything? |
| 19:50.07 | drmessano | Correct |
| 19:50.10 | Penguin | Ah. |
| 19:50.21 | drmessano | If you forward the ports, it's dumb forwarding |
| 19:50.23 | WIMPy | danfromuk: You don't have multiple Asterisk versions installed now, do you? |
| 19:50.31 | Penguin | I can discontinue forwarding just to test it. |
| 19:50.43 | danfromuk | I made sure to install the same version that freepbx had installed |
| 19:50.50 | drmessano | Penguin, you can, but you may need to unload that module, as WIMPy is stating |
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| 19:50.59 | danfromuk | let me reboot, maybe there were locked files or something |
| 19:51.04 | drmessano | I would suspect may=definitely |
| 19:51.10 | Penguin | I'm not sure, but Vyatta probably loads it when doing NAT. |
| 19:51.44 | Penguin | I'll see if I can unload it. |
| 19:51.53 | drmessano | nf_nat_sip seems pretty specific, and if it's known to cause issues, you probably want to unload or blacklist it |
| 19:52.36 | Penguin | I unloaded it manually, and it seems to have cleared all other modules related to sip. |
| 19:52.54 | drmessano | Hrm |
| 19:53.11 | Penguin | So now I guess I have to manually load nf_conntrack_sip. |
| 19:53.23 | WIMPy | Strange. |
| 19:53.27 | danfromuk | WIMPy: no luck |
| 19:53.48 | danfromuk | im going to try an earlier version of lcr |
| 19:53.50 | Penguin | Okay, loaded that one back. |
| 19:55.11 | WIMPy | danfromuk: configure should end up setting "AST_1_8_OR_HIGHER". |
| 19:55.19 | danfromuk | WIMPy: 1.5 has the same error |
| 19:55.31 | danfromuk | How do i check? |
| 19:55.55 | WIMPy | grep the Makefile |
| 19:57.07 | danfromuk | Can't find AST_1_8 |
| 19:57.20 | runfromnowhere | So before I go and do it myself - has anyone backported MixMonitor send/recieve file splitting to Asterisk 1.8? |
| 19:58.02 | WIMPy | danfromuk: It checks asterisk/channel.h. So either it found a wrong version or something else it going wrong. |
| 19:58.38 | WIMPy | As a dirty workaround you might be able to make -D... but the reason for the check failing might cause other issues. |
| 19:59.32 | danfromuk | Whats make -D? |
| 19:59.57 | WIMPy | You can set defines on the command line. |
| 20:01.27 | danfromuk | Do you know what it says AST_1_8_OR_HIGHER to? |
| 20:01.33 | danfromuk | sets* |
| 20:01.50 | WIMPy | nothing |
| 20:02.33 | danfromuk | I dont follow. make AST_1_8_OR_HIGHER="" ? |
| 20:03.07 | WIMPy | In the Makefile it should end up in AST_CFLAGS |
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| 20:03.57 | WIMPy | But I worry that you won't end up with the right version anyway. |
| 20:04.14 | WIMPy | There must be a reason it wasn't detected correctly. |
| 20:04.19 | danfromuk | I've just looked through the configure file, and i cant find any reference to AST_1_8...... |
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| 20:04.43 | WIMPy | On the old version or the new? |
| 20:04.50 | danfromuk | old one. |
| 20:05.01 | danfromuk | one sec while i get the latest one and see if its there |
| 20:05.02 | WIMPy | It's probably too old. |
| 20:05.33 | WIMPy | Speaking in version numbers Asterisk 1.8 support is very new. |
| 20:06.04 | WIMPy | Someone was too lazy to make release versions (and increase numbers), I guess. |
| 20:06.28 | danfromuk | Its not in the newest one either. |
| 20:06.38 | WIMPy | o.O |
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| 20:07.01 | danfromuk | I also dont see it in the make file |
| 20:07.38 | WIMPy | Well, that'd come fromthe configure script. |
| 20:08.01 | danfromuk | only if its actually in the configure file to start with :-) |
| 20:09.16 | WIMPy | checks |
| 20:09.45 | lnb | drmessano: reset ATA back to defaults. Now calls have sound. But the phone shows up at port 60978 |
| 20:10.11 | WIMPy | Ok back to a few days back.... Did you notice version 1.10? Or did you go for 1.7? |
| 20:10.18 | lnb | also removed static-port from nat line in PF |
| 20:10.41 | danfromuk | WIMPy: didnt spot 1.10. |
| 20:10.44 | danfromuk | one moment |
| 20:11.14 | WIMPy | The thing about version numbers and sorting. |
| 20:13.36 | danfromuk | Ok, this time the Make ended like this http://pastebin.com/XZzvZNVi |
| 20:15.14 | WIMPy | Ok, I guess you mISDN-user tools might be out of date. |
| 20:16.04 | WIMPy | Or even the wrong ones. (i.e. from mISDN 1). |
| 20:16.33 | WIMPy | Wait, no, you had misn_info so you do have them. |
| 20:17.40 | abradley | I've tried setting up a virtual extension to cascade immediate to a cell phone with no success. When I dial the virtual extension, I just get busy signal. How should I troubleshoot this? |
| 20:18.45 | WIMPy | All extensions are virtual, as there is no link to devices in Asterisk. |
| 20:21.19 | WIMPy | Err, and to answer the question: Look at the *CLI what happens. |
| 20:21.21 | danfromuk | just updated misdn user tools just in case. no difference. |
| 20:22.33 | WIMPy | Ok, let's see if that's really in the user part. |
| 20:23.38 | danfromuk | WIMPy: I discovered where the hfc4s8s came from. Well, at least why its used. DMES HFC-4S/8S: found adapter HFC-4S Evaluation Board (hfc4s8s_0) at 0000:02:07.0 |
| 20:24.36 | WIMPy | That tells you why? |
| 20:25.31 | WIMPy | But I guess it might be a good idea to get rind of anything related to the old mISDN. |
| 20:26.02 | WIMPy | Having two incompatible versions installed in paralled is probably not the best situation. |
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| 20:26.26 | drmessano | lnb, are the ATA's external? |
| 20:26.36 | drmessano | lnb, this one, specifically |
| 20:26.48 | lnb | drmessano: ATA is here in my officxe |
| 20:26.53 | lnb | Server is remote |
| 20:27.15 | drmessano | Sounds more to me like the RTP ports dont match what you have configured in the firewall |
| 20:27.38 | drmessano | or you've got the extension misconfigured |
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| 20:29.26 | danfromuk | WIMPy: i think i'm giving up for now. |
| 20:30.38 | WIMPy | At least you could get the fax working. |
| 20:31.16 | WIMPy | ... with what you've got so far. |
| 20:31.50 | WIMPy | Err, no. Wrong. Forget that. |
| 20:34.16 | WIMPy | Do you have multiple copies (versions) of mlayer3.h? |
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| 20:41.19 | lnb | drmessano: Registered SIP '300' at 99.238.64.55:59119 |
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| 21:48.39 | danfromuk | WIMPy: are you there? |
| 21:50.05 | danfromuk | WIMPy: if you see this, thanks for all your help. ive given up and told them its not possible. ive spent too long on this. |
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| 22:17.18 | WIMPy | danfromuk: You don;t give up on the finish line. |
| 22:36.02 | runfromnowhere | Well if this works, anyone want the patch? |
| 22:36.35 | WIMPy | For what? |
| 22:37.02 | runfromnowhere | Fingers crossed, I've backported MixMonitor's ability to record multiple channels from Asterisk 10 to Asterisk 1.8 (1.8.9 series to be specific) |
| 22:37.57 | WIMPy | And that was easier than upgrading? |
| 22:38.20 | runfromnowhere | I'd say it took 3 hours? |
| 22:38.31 | runfromnowhere | Almost definitely faster and less traumatic than a full upgrade, yeah |
| 22:41.10 | Chainsaw | runfromnowhere: Neat, please share. I have a backport of the T38 gateway for 1.8 as well. |
| 22:41.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.30) |
| 22:42.00 | runfromnowhere | Well, I'll share if I get it working :) |
| 22:42.13 | runfromnowhere | Nobody needs my busted code if I can't get my act together! |
| 22:42.20 | Chainsaw | runfromnowhere: Thanks. T38 backport and other "honestly, why didn't you ship 1.8 with this" patches for 1.8.24.0 here: http://mirror.meleeweb.net/pub/linux/gentoo/distfiles/gentoo-asterisk-patchset-1.17.tar.bz2 |
| 22:42.39 | runfromnowhere | Awesome :) |
| 22:43.05 | runfromnowhere | Unfortunately I'm dealing with a situation where I'm running FreePBX on top of Asterisk so a straight upgrade is a big problem for me |
| 22:43.11 | WIMPy | Even if there's no chance for the stuff to be released, jira might still be a good place to collect the patches. |
| 22:43.16 | Chainsaw | runfromnowhere: chainsaw@gentoo.org please, if it should be completed to your liking. |
| 22:43.28 | Chainsaw | WIMPy: Why, are they going to actually apply their patch backlog? |
| 22:43.34 | ghost75 | freepbx on top of asterisk oO |
| 22:44.02 | WIMPy | People might have a good chance to find them there. |
| 22:44.20 | runfromnowhere | ghost75: I didn't set it up :( |
| 22:44.26 | runfromnowhere | Yeah I'd want to share this if I can |
| 22:44.31 | runfromnowhere | Again, if it works - so far no dice |
| 22:45.35 | Chainsaw | WIMPy: It's obviously correct things like these bitrotting away that really make me despair: https://issues.asterisk.org/jira/browse/ASTERISK-17185 |
| 22:45.35 | LieutPants | [ASTERISK-17185] [Status: Open] [patch] SIP CHANNEL(rtpqos,audio,...) variables missing. - https://issues.asterisk.org/jira/browse/ASTERISK-17185 |
| 22:46.56 | Chainsaw | WIMPy: Or claiming to have SIP over SSL support without applying this: https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 22:46.57 | LieutPants | [ASTERISK-18345] [Status: Open] sips connection dropped by asterisk with a large INVITE - https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 22:48.01 | Chainsaw | WIMPy: Being able to actually *use* correct SSL certificates... https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 22:48.02 | LieutPants | [ASTERISK-18345] [Status: Open] sips connection dropped by asterisk with a large INVITE - https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 22:48.07 | Chainsaw | WIMPy: I could go on. For hours. |
| 22:48.22 | Chainsaw | https://issues.asterisk.org/jira/browse/ASTERISK-17727 even. |
| 22:48.23 | LieutPants | [ASTERISK-17727] [Status: Open] [patch] TLS doesn't get all certificate chain - https://issues.asterisk.org/jira/browse/ASTERISK-17727 |
| 22:48.27 | Chainsaw | Copy/paste is hard. Sorry. |
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| 22:50.29 | runfromnowhere | OK so now what I need to find is....what compile-time options my asterisk was compiled with |
| 22:50.31 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ptdkyyezspnoaqro) |
| 22:50.38 | WIMPy | I was just talking about users being able to find solutions, not about Digium using them for releases. |
| 22:52.48 | Chainsaw | WIMPy: Still remains disappointing that I can rattle off a list that complete and that convincing, years after the fact. |
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| 22:55.07 | WIMPy | I'm not going to argue that. |
| 22:55.15 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:55.32 | Chainsaw | WIMPy: That's disappointing. I was so ready. |
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| 22:58.18 | prometheanfire | Chainsaw: hihi |
| 22:58.27 | prometheanfire | goes to get some popcorn |
| 22:58.29 | Chainsaw | prometheanfire: Hey. You missed it I'm afraid. |
| 22:58.32 | prometheanfire | damn |
| 22:59.07 | Chainsaw | prometheanfire: I listed 17185, 17727 & 18345. |
| 23:00.30 | runfromnowhere | Welp, as of now I have no way to tell whether or not my module works |
| 23:00.45 | runfromnowhere | Because Asterisk won't load them due to "different compile-time parameters" and I have no way of knowing what the original compile time parameters are |
| 23:02.21 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.84) |
| 23:02.28 | Chainsaw | runfromnowhere: I couldn't find a convenient "core show" for that, indeed. |
| 23:03.43 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:04.30 | runfromnowhere | I have to say that working with this application has left me....hesitant about continuing to run it |
| 23:05.55 | prometheanfire | wonders when digium will get rid of their NIH syndrom |
| 23:06.12 | WIMPy | Oh, that's completely normal. |
| 23:06.18 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 23:06.19 | WIMPy | NIH? |
| 23:06.25 | prometheanfire | not in the projects I'm used to |
| 23:06.31 | runfromnowhere | "Not Invented Here" syndrome |
| 23:07.14 | runfromnowhere | The philosophy that anything that wasn't done in-house is clearly inferior and therefore should be thrown away in favor of something locally developed. |
| 23:09.21 | runfromnowhere | I mean I understand that you might want to make sure I don't try to load modules that were compiled under entirely different circumstances, but if that's what you want to do then why make it impossible for me to find out what the proper environment is? I mean, the application clearly has a record of it - it checks against it. |
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| 23:12.15 | Chainsaw | runfromnowhere: It's probably a compiler version check. |
| 23:12.29 | danfromuk | WIMPy: i couldnt see the finish line |
| 23:13.54 | runfromnowhere | Chainsaw: Nope. It's a check of all CFLAGS and BUILDDEPS used during the compile process |
| 23:14.10 | Chainsaw | runfromnowhere: Seems harsh to want to match *everything*. |
| 23:14.30 | runfromnowhere | It is extremely harsh |
| 23:14.36 | Chainsaw | runfromnowhere: Then again, I normally patch things in and then compile. So it can't hurt me. |
| 23:14.45 | runfromnowhere | Well if you can recompile and do a full reinstall, sure |
| 23:15.00 | runfromnowhere | Where I'm sitting now (which, I totally admit, is far from an ideal place) a compile and reinstall might just break the whole system |
| 23:15.04 | WIMPy | danfromuk: ok, but it was more than half way. |
| 23:17.42 | WIMPy | These places seem to be very popular in here. |
| 23:21.18 | WIMPy | danfromuk: did you see my question about versions of mlayer3.h? |
| 23:22.26 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 23:24.12 | runfromnowhere | sighs |
| 23:24.17 | runfromnowhere | Looks like it's going to be a full upgrade for me.... |
| 23:24.21 | gartral | ~book |
| 23:24.21 | 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 |
| 23:31.10 | gartral | arrrrrrgh! >.< |
| 23:33.18 | Chainsaw | runfromnowhere: Well, good luck. Talk later. |
| 23:33.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:35.11 | *** part/#asterisk prometheanfire (~promethea@gentoo/developer/prometheanfire) |
| 23:38.20 | *** join/#asterisk SGjunior (~sgjunior@out-pq-129.wireless.telus.com) |
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| 23:55.07 | *** join/#asterisk Shadowpillar (~wut@cpe-98-148-188-96.socal.res.rr.com) |
| 23:55.50 | Shadowpillar | quick question, I want to have a certain CID/outbound number when I call a certain area code. how would I accomplish this? |
| 23:56.08 | Shadowpillar | I was thinking based on the dial pattern, and I'm trying to figure out the pattern I'd use |
| 23:56.36 | Shadowpillar | example, call 310 area code, the callee would see a 310 area code, etc |
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| 23:58.50 | *** join/#asterisk barbosa2 (~juliano.b@177.158.1.247) |
