| 00:02.22 | iztech | woot, one of the trunks registered |
| 00:02.26 | iztech | thanks |
| 00:05.02 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 00:05.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 00:05.13 | leowt | ive got asterisk with sip clients. Wen connecting to a client outside the NAT i am able to talk to the 1st client, and the other ones get sip connection but no sound. |
| 00:05.22 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6cef::8899) |
| 00:05.24 | leowt | anyone has a clue about what is happening? |
| 00:07.06 | [TK]D-Fender | leowt: Improper NAT setup like I just explained to iztech |
| 00:07.13 | ChannelZ | deja vu |
| 00:07.29 | jpsharp | Again? |
| 00:07.48 | [TK]D-Fender | Deja moo : the feeling you've heard all this bull before.. |
| 00:07.50 | leowt | [TK]D-Fender: why im i able to call the first outside client? |
| 00:07.59 | leowt | i mean |
| 00:08.02 | leowt | have sound |
| 00:08.26 | [TK]D-Fender | leowt: show your actual setup and call attempts. |
| 00:08.31 | ChannelZ | You can't control everyone else's network. And it's important to know does sound work in either direction or just one? Which one? |
| 00:08.54 | leowt | either direction |
| 00:09.16 | leowt | the first caller network is identical to my third one |
| 00:09.23 | leowt | including sip client and OS |
| 00:10.21 | leowt | [TK]D-Fender: asterisk server with 2 sip clients in the same network |
| 00:10.32 | leowt | and then a router to internet, ant the other 4 clients |
| 00:10.34 | leowt | *and |
| 00:11.37 | [TK]D-Fender | leowt: I did not mean "give me a story about them", I mean "show me your configs and debug from actual call attempts" |
| 00:11.59 | leowt | [TK]D-Fender: sip.conf? |
| 00:12.11 | [TK]D-Fender | that would be a relevant config file... |
| 00:12.30 | leowt | just a sec |
| 00:14.40 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
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| 00:15.16 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 00:15.29 | leowt | http://pastebin.com/yQm19EUB |
| 00:18.21 | leowt | MYDDNS was replaced just to show you |
| 00:18.35 | leowt | just saying =P |
| 00:19.50 | ChannelZ | is your * behind NAT as well? |
| 00:19.56 | leowt | yep |
| 00:19.57 | ChannelZ | None of your peers are nat=yes |
| 00:20.21 | leowt | ChannelZ: ive made that, and didnt make a difference |
| 00:20.21 | [TK]D-Fender | directmedia=yes ; Asterisk by default tries to redirect the <- BAD |
| 00:20.27 | [TK]D-Fender | Automatic fail. |
| 00:20.41 | [TK]D-Fender | Next, permanently trash the 1200 lines of junk comments in there |
| 00:20.46 | ChannelZ | Did you port-forward a range of RTP ports as specified in rtp.conf? |
| 00:21.00 | leowt | 10000 to 20000 |
| 00:21.02 | leowt | yes |
| 00:21.37 | ChannelZ | and the firewall Asterisk is behind isn't preventing it from spewing UDP on pretty much any port (because you can't always control what the remote end is going to ask for) |
| 00:22.19 | [TK]D-Fender | [19:20][TK]D-Fenderdirectmedia=yes ; Asterisk by default tries to redirect the <- BAD |
| 00:22.40 | ChannelZ | very |
| 00:22.50 | leowt | [TK]D-Fender: directmedia=no |
| 00:22.52 | leowt | no sound |
| 00:22.54 | ChannelZ | because everyone is probably telling everyone else to connect to fake LAN IPs that don't exist |
| 00:23.39 | [TK]D-Fender | leowt: I also asked for the CALL DEBUG. but your sip.conf is a mess. clean all the comments out. |
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| 00:23.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:23.44 | leowt | a sec |
| 00:23.54 | [TK]D-Fender | leowt: And then provide the updated version along with the debug requested |
| 00:24.09 | leowt | sorry? |
| 00:24.28 | leowt | the version of *? |
| 00:24.34 | [TK]D-Fender | SIP.CONF |
| 00:24.52 | leowt | http://pastebin.com/Zy0xYEfx |
| 00:25.00 | leowt | its iqual |
| 00:25.05 | leowt | but directmedia=no |
| 00:26.53 | [TK]D-Fender | [Jan 22 00:19:29] WARNING[28254]: chan_sip.c:22276 handle_request_invite: Failed to set an alternate media source on glared reinvite. Video may not work properly on this call. |
| 00:26.59 | [TK]D-Fender | Directmedia = BAD |
| 00:27.06 | [TK]D-Fender | you are allowing REINIVITES. |
| 00:27.12 | [TK]D-Fender | that's what Directmedia is. |
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| 00:27.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 00:27.44 | leowt | but, what should i put in directmedia then? |
| 00:27.51 | [TK]D-Fender | NO |
| 00:28.08 | leowt | already done that |
| 00:28.12 | ChannelZ | blows his rape whistle |
| 00:28.16 | [TK]D-Fender | Apparently not in all the right places |
| 00:28.35 | [TK]D-Fender | clean the junk out of your config and pastebin it. |
| 00:28.57 | leowt | ok, sec |
| 00:45.40 | leowt | http://pastebin.com/xfj2PRzG |
| 00:50.29 | [TK]D-Fender | "sip set debug on" <- |
| 00:50.32 | [TK]D-Fender | waiting to see the call. |
| 00:50.42 | leowt | ? |
| 00:52.15 | leowt | ow |
| 00:52.20 | leowt | jiust a sec |
| 00:53.43 | leowt | im seeing that the caller that works have his ip |
| 00:53.52 | leowt | but the other no |
| 00:54.02 | saint_ | [TK]D-Fender: I did a sip trace for my audio which is not working. can you give me a hint ? :) http://pastebin.com/B56ry98M |
| 00:54.07 | leowt | izo@192.168.0.16:61585 |
| 00:54.14 | leowt | this is a local ip |
| 00:55.08 | saint_ | [TK]D-Fender: XXXXX is my local set behind a private network , YYYY is the ip of my asterisk server, so is moon.light.com |
| 00:55.08 | leowt | [TK]D-Fender: http://pastebin.com/2dFrVSB1 |
| 00:55.24 | leowt | i can see that this client shows a local ip |
| 00:55.39 | leowt | and the one that its working is showing his public ip |
| 00:56.27 | [TK]D-Fender | saint_: that is not a complete call. I do not see the full dialplan processing |
| 00:56.50 | [TK]D-Fender | leowt: And that isn't even a call |
| 00:56.54 | saint_ | [TK]D-Fender: i did sip set debug on , is there anything else to do ? |
| 00:57.07 | ChannelZ | leowt: the retransmitting is a bother.. |
| 00:57.22 | leowt | [TK]D-Fender: isnt there the error? |
| 00:57.22 | [TK]D-Fender | saint_: it isn't COMPLETE |
| 00:57.35 | leowt | ChannelZ: sorry? |
| 00:57.52 | saint_ | [TK]D-Fender: that is a full copy / past from the console. all i scambled was the public IP and fqdn |
| 00:58.17 | ChannelZ | leowt: your last paste doesn't really show much but it does show that Asterisk has re-transmitted the same packet multiple times because either it's not escaping your network or the other end isn't replying/the reply isn't making it back to you |
| 00:59.04 | leowt | ChannelZ: i can see that the other client rizo@ is showing a local ip |
| 00:59.18 | leowt | and not maching the ip on the top |
| 01:00.12 | leowt | but what causes this? |
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| 01:00.48 | leowt | the client that works is showing the external ip on top and the <sip:renato@ddns> with domain and not a local ip |
| 01:00.49 | ChannelZ | "rizo" is behind NAT and isn't aware of its external IP. that's one problem but if the peer is set nat=yes AND Asterisk gets some RTP from the actual IP (the 94.* one) it should all work out. |
| 01:01.05 | leowt | the peer is set nat=yes |
| 01:01.32 | ChannelZ | But another problem seems to be that your server is unable to send SIP to that peer, or it is and their reply is getting chewed up somewhere. |
| 01:01.44 | ChannelZ | >> Retransmitting #3 (NAT) to 94.132.85.47:61585 |
| 01:04.02 | leowt | going to try from a 3g in a phone |
| 01:09.34 | saint_ | [TK]D-Fender: so i just ran a sip trace on 2 calls , 1st one no audio, 2nd one audio (it usually works like that. first one no audio, 2nd one has audio).. they look exactly the same, even the rport , the only thing that change are the sequence vakyes and other random tags. any idea why i do not have audio all the time ? |
| 01:17.11 | leowt | brb |
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| 01:17.54 | saint_ | anyone can share a voicemail.conf file that handle Digium phones ? |
| 01:19.59 | sawgood | handle Digium series phones differently say then a Polyocom or Cisco? |
| 01:21.42 | saint_ | I don t know. I just setup a basic file with a vaey basic D70=>1234,saint,,, and my D70 does not do anything with that .. |
| 01:23.17 | saint_ | either that, or i screwed up what to put in front of voicemail in the D70 configuration |
| 01:24.21 | sawgood | I can pastebin a copy of one for you |
| 01:25.15 | saint_ | and I keep receiving in the console: Received SIP subscribe for peer without mailbox: D70 |
| 01:25.41 | saint_ | sawgood: I'll take it.. or you can just write 1 line here.. is D70 => 1234,saint,,, good ? |
| 01:25.43 | sawgood | Does that phone have an entry in /etc/asterisk/voicemail.conf? |
| 01:25.57 | saint_ | that's what I have in the voicemail.conf |
| 01:26.01 | saint_ | the D70 => xxxx |
| 01:26.25 | saint_ | in the D70 configuration, for the voicemail, I have: sip:d70:1234@asterisk_server |
| 01:26.47 | sawgood | [default] |
| 01:26.47 | sawgood | 342 => 10755,After Hours VM,support@ippbxsupport.com,,attach=yes|saycid=no|envelope=no|delete=no |
| 01:26.54 | sawgood | hows something like that? |
| 01:26.57 | saint_ | yeah, so i m pretty good |
| 01:27.11 | saint_ | I wonder if it's because my extension starts with a letter |
| 01:27.30 | sawgood | Well, I've asked bout that in the past, and found it should be ok |
| 01:28.13 | *** join/#asterisk leowt (~leowt@bl17-27-216.dsl.telepac.pt) |
| 01:28.26 | saint_ | I changed the voicemail number to 999 |
| 01:28.32 | saint_ | and I changed the voicemail number in the D70 to 999 |
| 01:28.43 | saint_ | now when I press MSGS it works, but it asks for my mailbox |
| 01:28.49 | saint_ | is there a way to automatize this ? |
| 01:29.07 | saint_ | and I still see the Received SIP subscrive for peer without mailbox: D70 in the logs |
| 01:35.19 | saint_ | i can t find anywhere how to setup the message waiting indication too on the D70.. |
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| 01:39.52 | saint_ | anyone know how to install iksemel-devel with yum ? |
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| 02:09.49 | mathi | hi |
| 02:09.56 | mathi | when I try to "make" libpri |
| 02:09.59 | mathi | it says : |
| 02:10.22 | mathi | q921.c:811:7: error: variable 'tei' set but not used [-Werror=unused-but-set-variable] |
| 02:18.37 | saint_ | anyone has gtalk / google voice working correctly ? |
| 02:18.37 | saint_ | when i try to call my system, I receive in the logs: Channel 'Gtalk/+1xxxxxxx-a454' sent into invalid extension 's' in context 'default', but no invalid handler |
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| 02:23.58 | leowt | [TK]D-Fender: localnet was bad |
| 02:24.02 | leowt | now it works |
| 02:24.06 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 02:24.17 | leowt | local ip gets replaced by dns |
| 02:24.42 | leowt | but i still dont understand why the first caller had the external ip instead of local ip |
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| 02:29.21 | leowt | and, do this mean that i got to put every possible local ip on the localnet=? |
| 02:29.28 | ruben231 | hi guys i ahve installed asterisk 1.4 on ubuntu-server 12.04 LTS and somehow wanted to email my voicemail directly, how do i setup it..any idea..? |
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| 03:51.32 | din3sh | I have installed 2 HP DL380 servers with digium cards, I get "PRI got event: HDLC Abort (6) on Primary D-channel of span 1" & "HDLC Bad FCS (8) on Primary D-channel of span 1" |
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| 04:20.21 | ruben231 | <PROTECTED> |
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| 04:27.39 | din3sh | your email entries in voicemail.conf are not working? |
| 04:31.21 | ruben231 | yes its not working |
| 04:36.56 | nix8n82 | You may have to setup an email relay |
| 04:44.59 | igcewieling | ruben231: Asterisk uses sendmail (or whatever MTA you have installed) to send the mail. Check there. |
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| 04:51.00 | ruben231 | igcewieling: but the problem i dont know how to setup sendmail |
| 04:51.47 | igcewieling | ruben231: learn. Are you sure your distro uses sendmail and not postfix or exim, both of which install a sendmail binary for compatability? |
| 04:52.48 | ruben231 | i just install teh basic ubuntu serevr 12.04 LTS |
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| 04:57.28 | kaldemar | ruben231: ubuntu has easy-to-follow howto's for mails. |
| 04:57.59 | kaldemar | ruben231: you're asking in the wrong place. |
| 05:11.08 | *** join/#asterisk voxter_ (~voxter@d23-16-70-150.bchsia.telus.net) |
| 05:11.39 | ChannelZ | hint: you might look at msmtp because if you aren't already running a mail server, chances are you don't want to. |
| 05:20.54 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 05:21.24 | deo_ | hi guys.. can we check in our asterisk server if one of pstn lines is not accessible or inactive?? |
| 05:21.30 | deo_ | trough the terminal... |
| 05:34.36 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 05:34.39 | DarthExpeditor | hey |
| 05:34.43 | DarthExpeditor | anyone still up? |
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| 06:43.57 | ChannelZ | neeewwwwwp |
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| 06:55.46 | ChannelZ | deo_: A DAHDI channel or..? |
| 06:57.05 | deo_ | nope ChannelZ |
| 06:57.14 | deo_ | a PSTN line |
| 06:58.07 | [TK]D-Fender | There are bout a dozen or more kind of PSTN lines |
| 06:58.21 | [TK]D-Fender | so that answer isn't saying much... |
| 06:58.51 | ChannelZ | core show channels |
| 06:58.57 | [TK]D-Fender | deo_: And doesn't describe how * even interfaces with it |
| 06:58.58 | ChannelZ | maybe. Not exactly sure what you're looking for. |
| 06:59.11 | [TK]D-Fender | ChannelZ: I I got this one :) |
| 07:02.49 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:11.07 | [TK]D-Fender | OR.. they could just drop off without answering.... |
| 07:11.30 | ChannelZ | ZzzZzzzzzZzzZzz |
| 07:11.49 | ChannelZ | narcolepsy maybe |
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| 07:40.39 | WIMPy | din3sh: Interrupt issues or configuration error. |
| 07:40.49 | WIMPy | Or a bad line. |
| 07:41.01 | deo_ | hi guys [TK]D-Fender ChannelZ ...sorry late reply |
| 07:41.15 | [TK]D-Fender | You've got a minute or two before I'm out |
| 07:41.21 | [TK]D-Fender | And it';s already extremely late here |
| 07:41.32 | deo_ | oopss okay maybe next time :) |
| 07:43.41 | [TK]D-Fender | I sai you have a minute or two |
| 07:43.43 | [TK]D-Fender | Don't waste them |
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| 07:44.36 | ChannelZ | He's Snoopy dancing. I just know it. |
| 07:47.25 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:51.59 | [TK]D-Fender | .... AND I'M GONE </meme> |
| 07:52.29 | ChannelZ | I don't know that one. |
| 08:06.58 | deo_ | ChannelZ: what i mean is a regular PSTN lines from telco... i find it hard to check if those lines are working.. theyre currently attached directly to a TDM400 Digium Card.. |
| 08:07.10 | deo_ | and i dont have any physical access to this server.. :( |
| 08:07.22 | ChannelZ | so that is a DAHDI channel. |
| 08:07.35 | deo_ | yes indeed it is :) |
| 08:07.46 | deo_ | sorry late reply |
| 08:08.02 | ChannelZ | well you told me "no" when I asked an hour ago but no matter |
| 08:08.46 | deo_ | please disregards those ChannelZ ... it is not meant to that.. :( |
| 08:08.56 | ChannelZ | You can check for alarms on the card with 'dahdi show status' but that sort of depends on what the problem really is |
| 08:09.34 | deo_ | i thought the problem is that line is not working already.... it may be that telco disconnect that line... |
| 08:09.35 | ChannelZ | if the lines are just not working between you and the telco there's not necessarily any way to know but to place a call on the channel and discover it doesn't work |
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| 08:11.01 | deo_ | dahdi show status shows ok |
| 08:13.44 | ChannelZ | well it's a pain in the ass but you can use dahdi_monitor on the shell to record a channel, then make a call out that channel, then download the recording and listen to it and see if the line is just dead silence or something |
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| 08:14.45 | pbxbrian | Hey there, anyone familiar with sangoma vega gateways? |
| 08:15.06 | pbxbrian | Is it possible to make them send either Remote-Party-ID or P-Asserted-Identity ? |
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| 08:28.01 | Guest90715 | hi all |
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| 08:28.41 | fukuda76140 | i'm a problem to configure T38modem with hylafax |
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| 08:39.12 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:41.06 | fukuda76140 | FaxGetty[24052]: /dev/ttyT38-1: Can not open modem |
| 08:42.14 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b4c3:d291:18b9:fd80) |
| 08:43.22 | *** join/#asterisk schmidts (~schmidts@fw.commpany.at) |
| 08:43.26 | schmidts | good morning |
| 08:44.47 | fukuda76140 | hi |
| 08:51.56 | fukuda76140 | help please |
| 08:55.36 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 08:57.33 | *** join/#asterisk din3sh (~din3sh@41.136.82.36) |
| 08:58.15 | din3sh | anybody here using CEL table/data for billing? |
| 08:58.35 | WIMPy | yes |
| 08:58.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:59.13 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:00.09 | fukuda76140 | i can't create a virtual modem t38modem |
| 09:00.48 | fukuda76140 | with faxstat -s i have : Modem ttyT38-1 (): Waiting for modem to come ready |
| 09:22.30 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.120) |
| 09:30.48 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 09:30.54 | bombev | hi all |
| 09:35.15 | bombev | I have strange issue here: http://pastebin.ca/2305784 look at line 89 |
| 09:36.45 | bombev | I dont understand why rom: <sip:48420212244024@........ I have two area codes first one 48-Poland, 420-Czech |
| 09:38.50 | schmidts | bombev from where did you get this call? maybe its just incoming like this |
| 09:39.21 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-163-094.pools.arcor-ip.net) |
| 09:40.15 | bombev | this call is from mine phone server |
| 09:41.00 | bombev | I have created diff context for that number : 48223821882 |
| 09:41.43 | bombev | http://pastebin.ca/2305785 |
| 09:42.25 | bombev | the real caller id is: 420212244024 but i am receiving 48420212244024 and no idea why |
| 09:45.56 | *** join/#asterisk PbxMan (c335d968@gateway/web/freenode/ip.195.53.217.104) |
| 09:45.58 | PbxMan | morning |
| 09:46.47 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 09:47.47 | wdoekes | bombev: if x.x.150.20 is your phone server then you're sending/adding the 48 there.. look there or upstream, not in the receiving asterisk |
| 09:48.30 | danfromuk | Hi. I'm starting to realise that there are issues with the way that sip.conf realtime has been implemented. I was wondering if anyone could tell me what the advantages of sip.conf realtime are compared to automatically generating the sip.conf file and issuing a sip reload ? |
| 09:50.26 | bombev | wdoekes *.*.150.20 is my trunk ip |
| 09:50.34 | bombev | not the IP of the phone server |
| 09:52.17 | wdoekes | start looking upstream. if you're the one placing the call with the trunk: what are you sending? |
| 09:53.07 | din3sh | WIMPy:how u track down/bill transfered calls for example? |
| 09:53.42 | WIMPy | I don't |
| 09:54.02 | *** join/#asterisk alsuren__ (~dlaban@80.169.133.251) |
| 09:54.12 | din3sh | transfers are not billed, or maybe you don't need to!? |
| 09:54.16 | bombev | wdoekes where can I upstream |
| 09:54.17 | bombev | ? |
| 09:55.02 | WIMPy | How do you do billing with Asterisk at all? There's no billing information in the CDRs. |
| 09:56.13 | fling | Hello! I need a fine how-to on SugarCRM <-> Asterisk integration. |
| 09:56.27 | wdoekes | bombev: you're receiving a call with XYZ in the CLI. where would you find the problem? with the sender, right? |
| 09:57.30 | bombev | wdoekes yes |
| 09:58.18 | wdoekes | the only sip trace you've shown, is the call-reception, not any sending |
| 09:59.14 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.120) |
| 10:00.31 | din3sh | either you develop your own billing interface using php/mysql or use software like Call Accounting Mate to parse the CDR. |
| 10:00.48 | bombev | wdoekes ok i will check the sender :) |
| 10:00.56 | wdoekes | that would make sense |
| 10:01.14 | wdoekes | and when you see that you're sending +42, you'll realise that the problem is between the sender and the recipient |
| 10:01.32 | wdoekes | and then you call the next hop after the sender to ask why your +42 gets translated |
| 10:02.50 | wdoekes | (or you can play with different methods of sending CLI, e.g. 0042 or using different headers: RPID, PAI) |
| 10:05.15 | bombev | how can i use those diff methods |
| 10:10.14 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 10:13.55 | fukuda76140 | who use T38modem here ? |
| 10:14.49 | *** join/#asterisk bjornts (~BTS@2001:700:200:10:19d9:6bad:456a:a34a) |
| 10:16.57 | *** join/#asterisk timahvo1 (~rogue@41.90.226.148) |
| 10:24.48 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 10:25.14 | *** join/#asterisk mathi (Matthew@ip-62-235-214-115.dsl.scarlet.be) |
| 10:25.16 | mathi | hi |
| 10:25.25 | mathi | I can't install libpri |
| 10:25.33 | mathi | here is the problem described: https://bugs.launchpad.net/ubuntu/+source/libpri/+bug/1102723 |
| 10:25.43 | mathi | no idea what to do next |
| 10:28.12 | hebber | mathi: it might help to download the Latest Version - 1.4.14 |
| 10:28.26 | hebber | mathi: http://www.asterisk.org/downloads/libpri |
| 10:28.33 | mathi | hebber, i'll try that now |
| 10:28.56 | WIMPy | That's not an error. |
| 10:29.03 | WIMPy | And you can probably ignore it. |
| 10:29.27 | WIMPy | But a current version is surely a good idea anyway. |
| 10:29.43 | *** join/#asterisk beebeeep (~beebeeep@thanatos.migalin.net) |
| 10:34.58 | mathi | hebber, with 1.4.14 I have another error: fatal error: dahdi/user.h: No such file or directory |
| 10:35.14 | mathi | (in pridump.c:45:24) |
| 10:35.27 | *** join/#asterisk timahvo1 (~rogue@41.90.226.148) |
| 10:36.12 | WIMPy | You need to install dahdi first. |
| 10:37.08 | mathi | WIMPy, that's not what is written here: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#Installing_id423812 |
| 10:37.14 | mathi | "With LibPRI installed, we can now install DAHDI." |
| 10:37.27 | WIMPy | It's outdated. |
| 10:37.32 | mathi | great ... |
| 10:37.41 | mathi | so now we need to do the opposite ? |
| 10:37.51 | WIMPy | yes |
| 10:37.59 | mathi | how do you know? |
| 10:37.59 | mathi | :P |
| 10:38.17 | WIMPy | You just told me :-) |
| 10:38.34 | WIMPy | But it's not the first time that issue came up. |
| 10:40.13 | mathi | should update the docs! |
| 10:42.20 | WIMPy | The 4th edition is already up for review. |
| 10:42.52 | WIMPy | http://ofps.oreilly.com/titles/9781449332426/index.html |
| 10:44.58 | mathi | then I have another problem whene xecuting command: |
| 10:45.07 | mathi | sudo apt-get install linux-headers-`uname -r` |
| 10:45.47 | mathi | Unable to lcoate package linux-headers-3.5.0-22-generic |
| 10:46.40 | WIMPy | Interesting. What Distro? |
| 10:46.51 | mathi | Lubuntu 12.10 |
| 10:46.59 | WIMPy | Having asked that, you might have more luck with that one in a channel for your distro. |
| 10:48.45 | mathi | ok I just asked, I hope i'll have an answer |
| 10:49.05 | mathi | I miss Windows |
| 10:49.35 | WIMPy | No dahdi for windows. |
| 10:50.04 | WIMPy | And teh old rule is: Make sure yor get Hardware that's supported by Linux. |
| 10:50.23 | WIMPy | Although dahdi may be neccessary. |
| 10:55.38 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 11:01.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:07.02 | *** join/#asterisk conashu (59217a9a@gateway/web/freenode/ip.89.33.122.154) |
| 11:07.24 | conashu | la buda? |
| 11:07.29 | conashu | sorry |
| 11:08.22 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:10.23 | *** part/#asterisk conashu (59217a9a@gateway/web/freenode/ip.89.33.122.154) |
| 11:26.10 | mathi | WIMPy, I willa dd a comment in the new book, ok ? |
| 11:31.56 | *** join/#asterisk leowt (~leowt@233.163.108.93.rev.vodafone.pt) |
| 11:33.18 | leowt | hi there, do i have to put every possible local network on localnet? or there is another way? |
| 11:43.49 | iztech | hey anyone here set up SipStation i am getting this error |
| 11:43.54 | iztech | [Jan 22 03:37:44] NOTICE[6536][C-00000002]: chan_sip.c:25184 handle_request_invite: Call from '' (184.72.227.214:5060) to extension '2137778888' rejected because extension not found in context 'incoming'. |
| 12:10.55 | *** join/#asterisk srp_ (~abhishek@115.115.144.218) |
| 12:13.16 | srp_ | Hello, I am trying to register SIPML 5 web sip client with asterisk, but Ive been getting 'Failed to connect to server' in the client. Is there any special configuration to be made on the client ? |
| 12:18.09 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 12:28.29 | *** join/#asterisk davlefouAMD (~david@197.15.234.38) |
| 12:30.13 | fukuda76140 | iztech your context is not create |
| 12:32.14 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:32.15 | iztech | fukuda76140: just mucked around my extensions.conf still with no luck |
| 12:32.29 | iztech | there is something simple i am missing |
| 12:32.36 | fukuda76140 | you use a virtual modem ? |
| 12:33.01 | iztech | sorry? |
| 12:33.27 | iztech | virtual? |
| 12:33.46 | fukuda76140 | you are using a modem virtual (like iaxmodem or T38modem) ? |
| 12:33.53 | iztech | no |
| 12:34.03 | fukuda76140 | ok |
| 12:34.25 | fukuda76140 | i'm sorry, i don't know |
| 12:34.36 | iztech | thx for trying |
| 12:35.08 | fukuda76140 | i'm beginner on asterisk |
| 12:37.28 | fukuda76140 | me i'm a problem to install and configure T38modem |
| 12:38.58 | ectospasm | iztech: you need an exten => ... line which matches the extension 2137778888 |
| 12:39.19 | ectospasm | possibly something like exten => _NXXNXXXXXX,... |
| 12:40.32 | iztech | in [incoming] section? |
| 12:41.10 | *** join/#asterisk mjordan (~mjordan@216.186.152.216) |
| 12:41.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:43.21 | iztech | exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}) |
| 12:43.27 | iztech | like that? |
| 12:44.34 | SeRi | iztech: #1 do you know who is 184.72.227.214? |
| 12:44.49 | SeRi | ec2-184-72-227-214.compute-1.amazonaws.com. |
| 12:45.00 | SeRi | Thats the system that it seems to be trying to csll you |
| 12:45.11 | SeRi | s/csll/call/ |
| 12:45.56 | iztech | heh, let me check |
| 12:45.59 | iztech | my logs |
| 12:46.48 | iztech | yeah, that's fine |
| 12:50.50 | iztech | http://pastebin.com/wJtq0wjs |
| 12:51.02 | iztech | thats what i have in my [incoming] in extensions.conf |
| 12:53.38 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 12:58.07 | ectospasm | iztech: you need to have an extension that matches your DID. So if the caller dials 2137778888 to reach you, you need "exten => 2137778888,1,..." |
| 12:58.35 | iztech | k, let me try |
| 12:58.36 | iztech | thanks |
| 13:02.23 | iztech | woot |
| 13:02.26 | iztech | thanks so much |
| 13:02.39 | iztech | good lord that took me a long time |
| 13:02.53 | ectospasm | don't worry about it, it took me two weeks to figure that out. |
| 13:03.46 | ectospasm | If you have multiple DIDs that need to be handled by the same dialplan code, use a pattern _NXXNXXXXXX, or similar. << iztech |
| 13:04.29 | mathi | i created a user/group "asteriskpbx", and changed the file permissions as specified in the docs, but now I have to log in into this user every time I want to run asterisk ? |
| 13:05.44 | ectospasm | mathi: the init script (in /etc/init.d/asterisk) should handle that. |
| 13:06.12 | *** part/#asterisk mjordan (~mjordan@216.186.152.216) |
| 13:08.45 | iztech | do you know what exten => s,n stands for? |
| 13:10.04 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 13:10.47 | iztech | so i did exten => 2137778888,1,Dial(SIP/101) |
| 13:10.58 | iztech | which worked |
| 13:19.42 | mathi | ectospasm, thanks ineed, i had to put sudo in front |
| 13:20.07 | mathi | should I use WAV, ALAW and GSM encodings ? |
| 13:20.47 | ectospasm | mathi: depends, what codec are your users/callers going to be using? |
| 13:21.08 | mathi | ectospasm, uhm... they are going to use normal phones |
| 13:21.51 | ectospasm | mathi: the term "normal phones" doesn't compute. Do you mean POTS/analog, or VoIP/SIP phones? |
| 13:22.03 | mathi | ectospasm, POTS/analog |
| 13:22.35 | ectospasm | mathi: so go with alaw/ulaw, which ever companding your telco uses. Check their documentation. |
| 13:23.28 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 13:23.34 | mathi | ectospasm, if I specify also GSM, Asterisk will first try to use ALAW because it knows it is better quality ? |
| 13:24.15 | ectospasm | mathi: if all you have is GSM, and your phones are using ALAW, Asterisk will transcode from GSM to ALAW. |
| 13:24.34 | ectospasm | It does not hurt 'cept for disk space to have multiple versions of an audio file |
| 13:24.53 | ectospasm | ...you don't specify the codec in dialplan, Asterisk selects the best one at runtime |
| 13:25.03 | ectospasm | ...at least that's how it's supposed to work |
| 13:27.35 | mathi | ah i see thanks) |
| 13:28.24 | *** join/#asterisk deo_ (~deo@112.198.90.5) |
| 13:40.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:40.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:46.41 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 13:46.49 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 13:48.08 | *** join/#asterisk ashgotti (d1e22c23@gateway/web/freenode/ip.209.226.44.35) |
| 13:49.18 | ashgotti | Hello all. I've been doing a bit of research to see if Asterisk would be the proper application for one of our clients. He runs a residence and would like to set up a touch screen kiosk in the lobby where visitors can look up residence, dial them, and be given access like most condos or apartments. Does Asterisk have this feature set? |
| 13:51.29 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 13:51.58 | ectospasm | ashgotti: it can do that, but it will take work. |
| 13:52.39 | ashgotti | ectospasm> We're not looking to do the work ourselves and will contract it out. We just don't want to go down a rabbit hole only to find out that it's not really the right tool. |
| 13:53.03 | ectospasm | ashgotti: Asterisk was specifically designed as the basis for projects like this. |
| 13:53.19 | ashgotti | They already have a voip system in place so that's what lead me to Asterisk. Do you know of any other applications that could do this, maybe more directly? |
| 13:53.19 | ectospasm | ...but it's not going to do all of it out of the box with minimal configuration. |
| 13:53.44 | *** join/#asterisk mihamina (~mihamina@41.190.237.66) |
| 13:53.46 | ashgotti | ectospasm> of course there will be configuration, maybe some customization |
| 13:54.01 | ashgotti | is there a forum or website for asterisk consultants or developers that we can engage? |
| 13:54.02 | ectospasm | I don't know of any turnkey system which will offer what you want. |
| 13:54.09 | beardy | Not anything without "special" hardware and software, and cabling, and costs. |
| 13:54.49 | ectospasm | yeah, the "glue" tying the touchscreen to the PC isn't handled by Asterisk. |
| 13:54.59 | ashgotti | beardy> I understand. They already have 3 kiosks. They had a partner for the software but it felt through. That's where we come in. |
| 13:55.19 | *** join/#asterisk serafie (~erin@nat/digium/x-qgubcyroywdddngs) |
| 14:00.34 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-14-212.broadband.tenet.odessa.ua) |
| 14:00.40 | ashgotti | I think I found it, Asterisk Exchange |
| 14:03.58 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-pqkmfqzzvxnvnaep) |
| 14:04.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 14:19.13 | fukuda76140 | what's the diferrence between iaxmodem and T38modem ? |
| 14:20.14 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 14:21.25 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-myraolntzgqzpjkw) |
| 14:21.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:25.09 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:25.30 | jmetro | one uses iax protocol im guessing. |
| 14:37.08 | Penguin | This is a real bother that all the files in the download site have dates of 28-Nov-2012. How am I supposed to track dates now? |
| 14:45.17 | mjordan | Penguin: which site? |
| 14:45.48 | Penguin | http://downloads.asterisk.org/pub/telephony/asterisk/releases/ |
| 14:45.53 | Penguin | Look at the dates. |
| 14:45.57 | mjordan | Penguin: nevermind. Yeah, I see it now |
| 14:46.01 | Penguin | |
| 14:46.11 | Penguin | It's jacked. |
| 14:47.05 | jmetro | oooh |
| 14:47.13 | jmetro | i didnt know what you meant at first, but that looks bad. |
| 14:47.57 | *** join/#asterisk nix8n82 (~AndChat27@216.67.131.253) |
| 14:50.48 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 14:55.15 | Katty | hello my asterisk does not work at all how to fix?? is urgent plz answer thx |
| 15:00.34 | jmetro | helly my critter cam only has sqirrls plz help i need birds. |
| 15:02.35 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:06.04 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:06.05 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:06.53 | din3sh | I have a 30-channel E1, I have this occasional problem where calls to any DID/number on this E1 does not even reach the * box. PSTN Caller hears an announcement from telco saying number is unavailable. |
| 15:07.45 | din3sh | Also I get HDLC errors on the span, are these 2 problems related? how to test/troubleshoot? |
| 15:09.54 | Katty | jmetro: there were BLUE BIRDS earlier :> :> :> |
| 15:10.00 | Katty | jmetro: 2 males and a female |
| 15:11.29 | jmetro | where do you live geographically? I never see blue birds or robins as much as your cam seems to |
| 15:11.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:12.01 | Katty | southern missouri |
| 15:12.08 | Penguin | We probably have more robins here than most other birds. |
| 15:12.12 | Katty | i've not seen any robin's on the feeder yet, but they tend to go back to the forest when it's cold. |
| 15:12.30 | Katty | and in all honesty, i've never had /any/ bluebirds on a feeder up until the next week. |
| 15:12.36 | Katty | not a single one in four years. |
| 15:12.52 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:12.57 | jmetro | i saw a red robin a couple times, and a cardinal just perching there for a while. |
| 15:13.10 | Katty | i'm not sure if it's the new area i've moved to...or if finding food in the frozen ground is particularly difficult this year...bluebirds eat mostly insects and fruit |
| 15:13.31 | Katty | strange. i've not seen a single robin since this past summer |
| 15:13.59 | jmetro | cause you don watch your own cam =p |
| 15:14.10 | Katty | http://1.bp.blogspot.com/-C7JFQC8zBLA/Tfkf6SmEsRI/AAAAAAAAA3I/HQWROAWW-Lg/s1600/house_finch.JPG <- you sure it wasn't one of those guys? |
| 15:14.37 | Katty | http://www.allaboutbirds.org/guide/PHOTO/LARGE/american_robin_6.jpg <- that's the native robin around here |
| 15:14.50 | Katty | a female bluebird would also be close |
| 15:14.54 | jmetro | definitely not a finch |
| 15:15.17 | Katty | http://www.allaboutbirds.org/guide/PHOTO/LARGE/eastern_bluebird_4.jpg <- female bluebird. |
| 15:15.22 | Katty | they can look more brown than blue on the back |
| 15:15.31 | Katty | the thing about a robin is that bright yellow beak. |
| 15:15.31 | jmetro | i saw two of the american robins last week |
| 15:15.40 | jmetro | together |
| 15:15.42 | Katty | insane!!! |
| 15:15.49 | Katty | but very welcome |
| 15:16.02 | jmetro | they were perched next to eachother on the hanging feeder |
| 15:16.19 | jmetro | wait thats a stand. The standing feeder. |
| 15:16.31 | Katty | i've never seen a robin eat seeds. |
| 15:16.40 | Katty | but they will be back, surely |
| 15:17.00 | Katty | the plan is to put up a window box over to the right, on that window |
| 15:17.09 | Katty | and in the window box put a small bird bath |
| 15:17.14 | Katty | along with some herbs around it |
| 15:18.03 | Katty | the little gray and white bird that comes in to steal a peanut and leave is a tufted titmouse. |
| 15:18.15 | Katty | i believe they migrate. |
| 15:18.35 | Katty | the black ones with the speckles are starlings. they're the birds you see in huge flocks flying in the sky |
| 15:18.53 | Katty | i'm not much of a fan of them, they tend come in 2 or 3 at a time and scare all the smaller birds off. |
| 15:18.54 | *** join/#asterisk bchia (~Adium@nat/digium/x-bzcemnkhsaffluit) |
| 15:19.09 | jmetro | Starlings are mean, according to the redwall books =p |
| 15:19.16 | Katty | i wouldn't doubt it |
| 15:19.19 | Katty | blue jays certainly are. |
| 15:19.23 | Penguin | They're trash. Kill 'em all. |
| 15:19.30 | Katty | i've seen a blue jay chase every other bird off the feeder. |
| 15:19.40 | Katty | except for a squirrel. |
| 15:19.57 | *** part/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 15:20.01 | Katty | the squirrels don't seem to care about the coming and going of birds. |
| 15:20.29 | Katty | bout the only thing that bothers them is the random car that drives by or if the cat swats the window...or if another squirrel gets too close for comfort. |
| 15:22.44 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 15:23.51 | Greenlight | AFternoon all. I'm getting an odd error on Asterisk 11. [Jan 22 16:26:41] WARNING[14386]: rtp_engine.c:1471 ast_rtp_instance_bridge: Got hangup while attempting to bridge 'SIP/minotaur-0000fa85' and 'SIP/70229-0000de81'. This is happening for about 10% of the bridges I try to initiate, and annoyingly it just seems to hangup both channels. Any ideas what might cause this? The only change I |
| 15:23.52 | Greenlight | think has been made is to use MixMonitor rather than Monitor |
| 15:24.11 | Penguin | That's a warning, not an error. |
| 15:25.32 | Qwell | Katty: squirrels aren't birds. just sayin'. :p |
| 15:25.45 | Greenlight | Ok, a warning. The bridge failes nontheless... |
| 15:26.38 | Greenlight | *fails |
| 15:27.09 | Penguin | What does the sip debug reveal? |
| 15:27.33 | Katty | Qwell: *hee* |
| 15:29.04 | Greenlight | I can't really enable SIP debg it's quite a busy server (200k+ calls a day) |
| 15:29.26 | Greenlight | Is that normal for a failed bridge to hangup both sides? |
| 15:29.53 | Penguin | I wouldn't have thought so, but I just don't know. |
| 15:30.51 | Greenlight | Fair enough. And are there any known issues with using MixMonitor on a channel that's being bridged? (Use ManagerBridge btw if it makes a difference) |
| 15:37.33 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
| 15:39.11 | *** join/#asterisk phonebuff (~chatzilla@23.31.148.65) |
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| 15:53.47 | BriGuy | Does anyone have experience with running Asterisk on VMware - but using RTP proxy on physical boxes? I wanted to know if that resolves the call quality issues... |
| 15:55.12 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
| 15:56.42 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 15:56.45 | Greenlight | As long as you're not *doing* anything with the audio (recording, voicemail etc) and it's just passing through, ideally in same codec, it'll be okay |
| 15:56.55 | _abc_ | Does anyone know if a cisco phone can be rebooted using asterisk sccp commands? |
| 15:57.02 | _abc_ | from debug mode or such in console |
| 15:57.21 | Greenlight | We can 500 simultenous channels over a virtualised box with no issues |
| 15:57.24 | Greenlight | *ran |
| 15:57.55 | BriGuy | We are recording VMs and recordings then passing them off to NFS - even using dahdi will that be an issue on a VM platform? |
| 15:58.09 | _abc_ | http://www.nmedia.net/~mklein/reboot.pl found it... |
| 15:59.31 | _abc_ | uhh telnet interface on 7940 phone not responding |
| 15:59.43 | _abc_ | I hate cisco 'clever' setups which lock out everything |
| 16:00.16 | Greenlight | With just recording you can get some success as it should use the timing from the source, and not rely on the local timing source, but I'd not recommend against it. I've personally seen issues with recordings when the local timing source goes waaaay out, which can happen on VM's |
| 16:00.21 | Penguin | If you are using chan_sccp, there's sccp restart on the CLI. If you are using chan_skinny, I wouldn't know (becuase I do not use it). |
| 16:00.25 | Rumbles | Hello, I'm having trouble installing asterisk-dahdi through the centos repo on asterisk.org, have some dependencies been broken in recent changes? |
| 16:01.36 | Rumbles | managed to get dahdi to install from source, and /etc/init.d/dahdi status shows the lines fine, but unable to load chan_dahdi.so in asterisk |
| 16:01.37 | BriGuy | how do you handle call recording/VMs on the VMs then? |
| 16:03.09 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 16:03.14 | Greenlight | I've never used a VM based Asterisk to do recordings on. |
| 16:03.15 | Rumbles | this is Asterisk 1.8.20 btw |
| 16:04.05 | leifmadsen | we do it all the time |
| 16:04.09 | leifmadsen | works fine in the ramdisk |
| 16:04.18 | _abc_ | Penguin: sccp. Thanks. btw good info here: http://www.voip-info.org/wiki/index.php?page_id=542 |
| 16:04.18 | Greenlight | Yea, if you are doing it, use a ramdisk |
| 16:04.29 | leifmadsen | http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html |
| 16:05.04 | BriGuy | good idea - thanks for the input Greenlight & leifmadsen |
| 16:05.05 | phonebuff | Speaking of Cisco -- The BLF patch that was done ofr 1.8.12 fails with 1.8.19. Does anyone know if there are plans to get this excepted into the the code, and/or where it broke between 18.12 & 1.8.19 ? |
| 16:05.31 | Greenlight | I'm lead to beleive it's more efficent to use /dev/shm which doesn't have the overhead of ext filesystem, but that's probably nitpicking |
| 16:05.31 | leifmadsen | you'll have to elaborate on what patch and where it is |
| 16:05.59 | phonebuff | https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-201850 |
| 16:06.37 | phonebuff | https://issues.asterisk.org/jira/secure/attachment/43780/gareth-1.8.12.1.patch |
| 16:06.38 | jmetro | theres filesystems besides zfs? |
| 16:07.02 | phonebuff | https://issues.asterisk.org/jira/secure/attachment/44090/gareth-1.8.19.0.patch |
| 16:07.07 | leifmadsen | phonebuff: ya that looks like a feature patch -- will only be applied to trunk whenever it is marked as done |
| 16:07.07 | Greenlight | I've heard rumours.... |
| 16:07.20 | leifmadsen | phonebuff: you still haven't figured out how to fix the conflicts? |
| 16:08.07 | WIMPy | din3sh: Yes, those issues are most probably related. As I said earlier: Interrupt issues, wrong configuration or a bad line. |
| 16:08.20 | _abc_ | Penguin: hmm okay that reset the device but not from cold start, it keeps its old ip, I need it to get a new dhcp lease.... any ideas on how to make it do that? |
| 16:08.47 | Penguin | Try sccp reset instead. |
| 16:08.48 | phonebuff | No I have not, figure I am going to have to go back to 1.8.12 source and compare that against 1.8.19 source and then see if I can find the errors. patch -p0 ......... tells me 4 out of 144 hunks failed |
| 16:08.59 | _abc_ | Penguin: it's skinny not sccp, sorry |
| 16:09.06 | _abc_ | Penguin: reset exists in skinny too |
| 16:09.28 | phonebuff | but the make errors look more like a comment bracket or something is way out of place |
| 16:09.50 | phonebuff | Just two many error messages. |
| 16:10.02 | leifmadsen | phonebuff: sounds about right, the code likely changed and you'll have to look at the failed hunks and then apply them manually, then export the patch for that version |
| 16:10.27 | leifmadsen | if you can't figure it out, you'll have to wait for the reporter to update it, or hire a consultant/developer to create the patch for you |
| 16:11.32 | phonebuff | Leif: if I do that for 1.8.19 is it likely to be broken again almost immediately for 1.8.20 ? |
| 16:12.30 | phonebuff | Leif: I have no issue working the code other than time, I am not however familiar with the patch routine and diff files, can you point me to some learning material. |
| 16:12.35 | *** join/#asterisk gr0mit (~tim@2a01:9c41:ffff:30:340d:eda2:2505:72cd) |
| 16:13.27 | *** join/#asterisk Becker54 (~becker54@c-98-219-183-45.hsd1.pa.comcast.net) |
| 16:13.41 | _abc_ | Penguin: mea culpa, seems to work fine, the reboot is 10 times faster than before, I have a new dhcpd online heh |
| 16:13.50 | Becker54 | Hello room |
| 16:13.59 | _abc_ | Penguin: I rebooted it about 5 times and it was so fast I did not have time to see it happen so I thought it didn't |
| 16:13.59 | Penguin | phonebuff: http://jungels.net/articles/diff-patch-ten-minutes.html |
| 16:14.10 | _abc_ | Penguin: all ok on skinny rebooting ciscos heh |
| 16:14.16 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:14.16 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:15.23 | Penguin | _abc_: I think it is sccp reset that doesn't actually do a reboot, and it happens in a fraction of a second. The sccp restart should take at least a second and you should be able to see that happen. |
| 16:15.26 | phonebuff | Great I will do some reading -- Since the apply failures arein sip.c I have a feeling that I maintaining / applying this patch may be a very long term effort. |
| 16:15.37 | Penguin | I could have the two commands reversed -- I do that all the time. |
| 16:15.51 | Becker54 | I was wondering if anyone could assist me in getting SIP messaging working or find information on why it is not? |
| 16:16.21 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 16:16.49 | phonebuff | By the way, it's the use of diff output by patch . :-) |
| 16:17.05 | leifmadsen | phonebuff: it depends entirely where changes happen, so that isn't a question one can answer |
| 16:17.20 | _abc_ | Penguin: I tried both, again, skinny has the same options |
| 16:17.28 | Penguin | Good to know. |
| 16:17.30 | _abc_ | Penguin: neither takes more than 2 seconds with my settings |
| 16:18.30 | _abc_ | Penguin: do you have a sccp/skinny config (cnf) template for ciscos on non-sip? |
| 16:18.43 | Penguin | phonebuff: When I have had to port a patch from one version of code to another, I will often look at the code side by side using sdiff. |
| 16:18.46 | _abc_ | I have a hard time locating such. If you have a template, could you share it? |
| 16:19.05 | phonebuff | Well from my reading and looking at the code so far, it appears that both some of the latest revs per the change logs, and many of the changes needed in this patch are i the same areas (memory management) |
| 16:19.49 | Penguin | _abc_: Are you referring to the SEP<MAC>.cnf.xml files? |
| 16:21.02 | Rumbles | does anyone here know if asterisk-dahdi can be install on CentOS 5.9 via the asterisk.org repo? I'm trying to install it and it's asking for an old kernel-i686 |
| 16:21.09 | phonebuff | By the way on a side note Penguin be careful of the Cisco 9 dot firmware for 7975s and others, had to fall back to an 8 dot release to get the phone to handshake with 1.8.19 |
| 16:21.50 | _abc_ | Penguin: yes, but I prefer the generalized XMLDefaul |
| 16:21.53 | _abc_ | t.cnf.xml |
| 16:21.57 | Rumbles | i.e. asking for "Processing Dependency: kernel-i686 = 2.6.18-308.24.1.el5 for package: kmod-dahdi-linux" when I am using kernel 2.6.18-348.el5PAE |
| 16:23.22 | phonebuff | Leif: Is there a note in the Wiki that talks about the process to get patches incorporated ? |
| 16:23.26 | Penguin | _abc_: http://www.voip-info.org/wiki/view/SCCP-HOWTO2 |
| 16:24.03 | Becker54 | Sorry can't help you Rumbles are you using a VM or real hardware? |
| 16:24.10 | Rumbles | t |
| 16:24.14 | Rumbles | sorry, real hardware |
| 16:24.20 | _abc_ | Penguin: I think I have that, it is not applicable to skinny |
| 16:24.34 | Rumbles | first time we've had this issue but it's the first time we have had to use CentOS 5.9 |
| 16:25.17 | Becker54 | what is the repo address? |
| 16:25.21 | Penguin | _abc_: I don't see why it wouldn't be. chan_skinny and chan_sccp are only the asterisk channel drivers; the .cnf.xml files are for the phones' firmware. |
| 16:25.29 | Rumbles | thnx Becker54, 1 sec |
| 16:25.48 | _abc_ | Penguin: it's related to everyone explaining how to upgrade to sip and pretty little else |
| 16:25.51 | *** join/#asterisk minotaur01 (~minotaur0@24.215.3.50) |
| 16:26.06 | _abc_ | Penguin: do you know, for example, how to enable telnet in the cnf file? |
| 16:26.11 | Rumbles | from the repo file: baseurl=http://packages.asterisk.org/centos/$releasever/asterisk-1.8/$basearch/ |
| 16:26.12 | Penguin | _abc_: Sounds like you read the wrong thing. |
| 16:26.16 | _abc_ | no |
| 16:28.31 | Becker54 | I am looking hang on |
| 16:28.44 | Qwell | There is no package for that kernel yet. |
| 16:28.50 | Qwell | Red Hat broke an interface. Again. |
| 16:29.05 | Rumbles | grand |
| 16:29.05 | Penguin | I'm not even sure that the sccp firmware has telnet. I've never seen a setting for it at all. |
| 16:29.11 | _abc_ | ah |
| 16:29.25 | _abc_ | just a sec I have a reference, maybe I misread |
| 16:29.28 | Penguin | The sip firmware has telnet. I know that. |
| 16:29.40 | Rumbles | thanks Qwell |
| 16:30.03 | Rumbles | now I just need to figure out a way to get it to load :/ |
| 16:30.33 | *** join/#asterisk minotaur01 (~minotaur0@HMTNON14-1176243898.sdsl.bell.ca) |
| 16:30.42 | Rumbles | Qwell, any idea when the package will be out? |
| 16:30.50 | Qwell | Once DAHDI 2.6.2 gets released. |
| 16:30.59 | Becker54 | that stinks sorry rumbles |
| 16:31.43 | navaismo | Hi does asterisk 11.2 set/send the profile-level-id in the sdp by default? https://supportforums.cisco.com/community/netpro/collaboration-voice-video/telepresence/blog/2011/01/14/video--telepresence-sip-h264-profile-level-id |
| 16:31.49 | Rumbles | yeah, but not much I can do to help that.... These phone servers are meant to go live in 16 hours :/ |
| 16:32.23 | Penguin | _abc_: But yeah, http://www.voip-info.org/wiki/view/SCCP-HOWTO2 is only for configuring the phones with sccp/skinny, and has nothing to do with sip. If you think it is trying to get you to switch over to sip, you read the wrong thing. |
| 16:32.33 | Becker54 | I am new to Asterisk and trying to figure out SMS in asterisk 11. I have Asterisk 11 running and added the dialplan and edited the sip.conf file accordingly but SMS still does not work. When I disconnect from PBX SMS works without PBX. Scratching my head. |
| 16:32.50 | _abc_ | Penguin: no, I was referencing another article on the web |
| 16:32.56 | Rumbles | I'm using sangoma cards so I'll try getting woomera working |
| 16:33.00 | _abc_ | Penguin: you seem to be right, there may be no telnet on skinny |
| 16:33.12 | _abc_ | Penguin: it does have a dandy web page in the phone, though |
| 16:33.25 | _abc_ | why they could not add a cgi button there for reset and such? |
| 16:33.33 | _abc_ | The mysteries of large company products |
| 16:33.45 | Penguin | You probably don't need either, though. Most things can be done with the config files and a restart/reset from the CLI. |
| 16:34.19 | _abc_ | I hope so. I have to wait for someone to get back to the relevant office so I can verify that the phone can really do what it should when cold booted by unplugging. |
| 16:34.38 | _abc_ | Otherwise I might get a *phone* call in the morning about it (from another phone >:) |
| 16:34.47 | _abc_ | thanks for the tips for now |
| 16:35.02 | _abc_ | wanders away to find some food and patch the always empty coffee cup |
| 16:35.35 | *** part/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 16:35.50 | Becker54 | I am new to Asterisk and trying to figure out SMS in asterisk 11. I have Asterisk 11 running and added the dialplan and edited the sip.conf file accordingly but SMS still does not work. When I disconnect from PBX SMS works without PBX. Scratching my head. |
| 16:37.37 | Becker54 | Anybody out there |
| 16:38.23 | Greenlight | Why would I get a "BridgeAction Suceess" message over the AMI, yet get a Bridge failure message in the CLI... for the same bridge? |
| 16:39.19 | WIMPy | Becker54: Patience. This is not the 1.99/minute hotline. But You need to tell from and to where you're trying to get SMS working. |
| 16:40.34 | Becker54 | Sorry I am not real familiar with IRC. I used it once 10 years ago. Not really up with the etiquette. |
| 16:40.39 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 16:41.33 | *** join/#asterisk rue_work2 (~rue_mohr@24-207-100-190.eastlink.ca) |
| 16:42.09 | rue_work2 | turns out connecting the mgcp machine to the network to be able to post the errors is going to be a job and a half |
| 16:42.17 | rue_work2 | its an isolated staticly addressed network |
| 16:42.18 | Becker54 | I have 2 WiFi SIP handsets that have SIP SMS enabled. They will send messages to each other when the PBX is not reachable. However, when SIP is registered SMS stops. I am running Asterisk 11 |
| 16:45.06 | *** join/#asterisk scubes13 (~scubes13@39.sub-70-193-15.myvzw.com) |
| 16:45.13 | Becker54 | I ran a packet capture and saw the phones communicate "Request: MESSAGE sip:6000:10.1.1.1 (text/plain) |
| 16:45.32 | Becker54 | The phone that sent was ext 6000 and asterisk is 10.1.1.1 |
| 16:45.35 | *** join/#asterisk SteveWilliams (~chatzilla@59.162.182.220) |
| 16:45.42 | SteveWilliams | hello everyone... |
| 16:45.42 | *** join/#asterisk nix8n82 (~AndChat27@216.67.131.253) |
| 16:46.37 | Becker54 | there was no activity to EXT 6001 which I sent the text to. So I am thinking the server is the problem |
| 16:48.08 | SteveWilliams | i have an asterisk server with asterisk 1.6.2.16-68.... it has two ethernets.... eth1 has a public ip address and eth0 has a private ip address... i would like no sip registrations on eth1... how do i do it....please help... |
| 16:49.46 | WIMPy | You can bint to the internal address. But that will completely disable SIP on other addresses. |
| 16:51.12 | Becker54 | I have mine set that way. For ex. my external is 192.168.1.x and my internal(for asterisk) is 10.1.1.X. I just set the bindaddress to my NIC IP intended for asterisk only |
| 16:51.55 | *** join/#asterisk timahvo1 (~rogue@41.80.120.86) |
| 16:52.51 | SteveWilliams | WIMPy: sorry... i am new to asterisk... but where do i do that.... and would that hinder me from registering my asterisk server as a sip client to other sip server in any way... |
| 16:52.55 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:53.08 | phonebuff | Steve: If it were me I would get the box off the Internet entirely and put a Firewall between it and the world, them place rules for what is and is not allowed. www.pfsense.org |
| 16:53.25 | phonebuff | Otherwise look at iptables -- |
| 16:53.26 | WIMPy | SteveWilliams: sip.conf and yes. |
| 16:53.55 | WIMPy | SteveWilliams: Yu can use permit/deny for your peers or ACLS if you use a current version of Asterisk. |
| 16:54.26 | SteveWilliams | WIMPy: i have asterisk 1.6.2.16-68 will permit/deny work?? |
| 16:54.55 | WIMPy | yes, but not ACLs. |
| 16:55.12 | WIMPy | But you should get a more recent version anyway. |
| 16:55.18 | rue_work2 | ok, I have an mgcp problem |
| 16:55.27 | rue_work2 | I cant define two gateways |
| 16:55.29 | SteveWilliams | WIMPy: yes, you're right.... |
| 16:55.36 | rue_work2 | if I do, I get an error for the first gateway defined |
| 16:56.16 | rue_work2 | maximum retries exceeded for transaction # on [gateway] |
| 16:56.20 | rue_work2 | where # is a number |
| 16:56.38 | rue_work2 | and [gateway] is the name of the gateway that was defined FIRST in mgcp.conf |
| 16:56.56 | rue_work2 | because I tied swapping them |
| 16:57.08 | SteveWilliams | WIMPy: what if i block access to port 5060 on eth1 which has a public ip address... will that hinder me from registering my asterisk server as a sip client to other sip server |
| 16:57.08 | rue_work2 | the last defined gateway always works fine |
| 16:58.35 | rue_work2 | I understand to get help with my question I need to post every config file and every log file, so I'll just work on that now |
| 16:58.41 | WIMPy | SteveWilliams: Depends on how exactely you do it. |
| 16:59.47 | SteveWilliams | WIMPy: through iptables.... the command could be: iptables -A INPUT -i eth1 -p tcp --dport 443 -m state --state NEW,ESTABLISHED -j DROP |
| 16:59.57 | WIMPy | If you yse connection tracking to allow replies to what you send out, it will be ok, if you just shut that port there will be no sip at all. |
| 17:00.07 | SteveWilliams | WIMPy: sorry... port would be 5060 not 443 |
| 17:00.12 | WIMPy | Looks good. |
| 17:00.31 | WIMPy | err, yes and protocol udp. |
| 17:00.48 | WIMPy | wait |
| 17:01.00 | WIMPy | ESTABLISHED DROP? What kind of combination is that? |
| 17:01.24 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
| 17:01.35 | SteveWilliams | WIMPy: oops... something wrong... ?? sorry new to linux admin as well... new job |
| 17:02.02 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
| 17:02.33 | WIMPy | Better drop everything and accept ESTABLISHED (and RELATED). |
| 17:03.04 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 17:03.17 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:03.21 | SteveWilliams | WIMPy: okay... will try that out... thanks.. |
| 17:03.48 | WIMPy | that's not port specific, BTW. |
| 17:04.21 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 17:05.50 | SteveWilliams | WIMPy: okay.... i see that blocking connections on 5060 for eth1 still allows sip registration... |
| 17:08.21 | apb1963_ | Anyone have experience with the Cisco SPA 502G phone? I'm thinking about getting one to use with asterisk |
| 17:12.03 | jmetro | we use spa 509s. Work awesomely. |
| 17:12.03 | *** part/#asterisk sorressean (~tyler@tds-solutions.net) |
| 17:13.04 | apb1963_ | ok |
| 17:13.20 | apb1963_ | not quite the same thing, but nice to know :) |
| 17:14.31 | jmetro | the added blf's are pretty righteous. |
| 17:14.42 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:16.34 | Rumbles | Qwell, I don't want to bug you, and if you don't have a solid date don't worry, but do you have a rough eta for dahdi 2.6.2? Just so I can let the guy who's setting the next servers up about this issue. |
| 17:17.03 | Greenlight | Is anyone around that's familiar with some of the source code. I'm trying to track down an issue I've started experiencing when Bridging after I've moved from Monitor to MixMonitor. In features.c, in the ast_bridge_call function, there is a section of code which seems to do something if the Monitor app is running, but there's nothing about MixMonitor - was wondering if that could have an |
| 17:17.03 | Greenlight | effect, or if there's a good reason ? |
| 17:17.53 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:e90b:7171:1947:d35e) |
| 17:17.57 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 17:19.43 | mjordan | Greenlight: MixMonitor works differently than Monitor. MixMonitor works through a feature known as audiohooks that manipulate frames on the channel. Monitor (being rather old and crusty) didn't have such a feature when it was written, and so it's kind of shoe horned into various locations. |
| 17:20.12 | mjordan | Greenlight: the code in ast_bridge_call is to initiate the Monitor when the bridge is started. In the case of MixMonitor, that already happens when the application is called and the frame hooks are attached to the appropriate channel. |
| 17:22.09 | Greenlight | Ahh okay, so what seems to be happening in around 5% or so of the bridges is that ast_rtp_instance_bridge fails reporting that one oft the channels has hungup. I initiate the bridge at pretty much the same time as I request the MixMonitor - could this cause some sort of race condition? I also allow audiohook inheritance by setting it as soon as I've had a repsonse to say MixMonitor has started |
| 17:23.40 | mjordan | You get the error message "Got hangup while attempting to bridge '[chan]' and '[chan]'"? |
| 17:24.09 | Greenlight | Exactly, which I tracked down to that function |
| 17:25.36 | Greenlight | My guess is that ast_channel_bridge which gets called inside ast_bridge_call points to ast_rtp_instance_bridge ? |
| 17:25.44 | mjordan | Greenlight: hm. |
| 17:26.05 | mjordan | there is a possible way MixMonitor could hang up the channel when it's attached |
| 17:26.29 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:26.41 | Greenlight | And would that be related to if there was a bridge in progress ? |
| 17:27.09 | mjordan | ah. Well, not really a hang up. MixMonitor is going to attempt to break a native bridge if it detects that the channels are in one |
| 17:27.32 | Greenlight | Well they're same tech, same codec so they woujl |
| 17:27.35 | Greenlight | *would be |
| 17:27.40 | mjordan | because it needs the audio frames in order to record - so a packet to packet bridge has to be broken |
| 17:27.51 | Greenlight | Yea, makes sense, I never even considered that |
| 17:28.19 | Greenlight | *could* that perhaps cause the check inside ast_rtp_instance_bridge (ast_check_hangup) to return false? |
| 17:28.40 | mjordan | that is probably what's happening. One of the channels has a soft hangup flag to tell it to break the native bridge, but the native bridge hasn't fully started yet |
| 17:28.53 | mjordan | it checks that they aren't hung up, which includes the soft hangup flag, and bails |
| 17:29.02 | mjordan | so yes, that sounds like a bug and a race condition |
| 17:29.07 | Greenlight | Yea, that makse perfect sense. |
| 17:29.25 | Greenlight | Soo...... a *quick* and bloody dirty fix is set the outgoing side to a differnet codec? |
| 17:29.32 | Greenlight | Then work on the issue ? |
| 17:30.08 | Greenlight | Which, would force a *proper* bridge, if I understand how it works? |
| 17:30.11 | mjordan | or just put an option on the Dial statement that prevents a native bridge in the first place |
| 17:30.18 | mjordan | such as a DTMF feature |
| 17:30.31 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:30.49 | Greenlight | Ah yea, perfect! |
| 17:31.12 | mjordan | ast_check_hangup should probably not consider the soft hangup flag AST_SOFTHANGUP_UNBRIDGE as a proper hang up, but I'm not sure what all that will effect without digging into it |
| 17:31.17 | mjordan | please do report this as an issue |
| 17:31.18 | mjordan | :-) |
| 17:32.06 | Greenlight | I'll do the dirty fix and then have a delve into it, if it's easy enough to fix I'll submit the fix when I raise the issue |
| 17:32.13 | mjordan | that'd be awesome |
| 17:32.19 | Greenlight | Thankyou ever so much for your help, it's much appriciated! |
| 17:32.26 | mjordan | np - thanks for digging into it! |
| 17:33.01 | leifmadsen | +1 |
| 17:35.32 | Katty | my every day here is a fight :< |
| 17:35.44 | Katty | mayhaps it's time to think about going elsewhere |
| 17:36.35 | nuken | does anybody have some experience with sip trunks between asterisk and cisco routers ? |
| 17:36.43 | nuken | i'm unable to use codec ilbc |
| 17:37.00 | Greenlight | I've a few installations with Asterisk <-> Cisco. All alaw though. |
| 17:37.01 | nuken | calls from cisco -> asterisk, work fine with ilbc |
| 17:37.12 | nuken | asterisk -> cisco, just alaw |
| 17:37.31 | Greenlight | Im my experiance Cisco tend to try and have their own way of doing things, and think everyone else should adapt to them ;/ |
| 17:37.54 | Greenlight | When does it fail ? |
| 17:38.37 | nuken | i'm montoring with snmp the links |
| 17:39.03 | nuken | from my softphone to asterisk server, ilbc is always used |
| 17:39.21 | Greenlight | Isnt't that just a codec preference issue ? |
| 17:39.31 | Rumbles | does anyone know if I can compile chan_dahdi.so from source, to correct the issue I am having due to the kernels available on CentOS 5.9? I'm not getting far with my google search to see if it is possible |
| 17:39.32 | Greenlight | Have you tried setting asterisk to only allow alaw |
| 17:40.11 | nuken | I've tryied setting asterisk to use just ilbc, that is what i want |
| 17:40.22 | Greenlight | Yea, sorry, just ilibc |
| 17:40.39 | nuken | this way, asterisk can't complete the call |
| 17:40.48 | nuken | busy chanel to cisco |
| 17:41.09 | Greenlight | Grab a sip trace, but sounds like Cisco isn't allow that codec for calls in that direction |
| 17:41.16 | WIMPy | Rumbles: You do need the headers for your kernel. No way around that. |
| 17:41.28 | Greenlight | Rumbles: Not sure, but you're only a wget and a make away from finding out... |
| 17:41.45 | nuken | sip trace? with debug ccsip in router ? |
| 17:42.53 | Rumbles | I have kernel headers, just wanted to make sure it was possible |
| 17:43.22 | nuken | Greenlight, please, take a look at this config, is that i'm using in cisco router... http://pastebin.com/xBTyd7Kj |
| 17:43.35 | nuken | this is what you use too ? |
| 17:43.41 | Rumbles | where can you download the source from Greenlight ? |
| 17:44.02 | WIMPy | rumbles: Try it |
| 17:44.12 | WIMPy | http://downloads.asterisk.org/pub/ |
| 17:44.56 | Greenlight | nuken: I've always worked with a 3rd party who configured the Cisco side of things, sorry |
| 17:45.06 | Rumbles | thanks WIMPy |
| 17:46.26 | nuken | no problem Greenlight , thanks anyway |
| 17:46.56 | Rumbles | make install bombed out on 2.6.1 :/ |
| 17:47.41 | WIMPy | What? You can make but can't make install? |
| 17:47.58 | Rumbles | ah, I only tried make install :) |
| 17:48.03 | Rumbles | does a make |
| 17:48.09 | Greenlight | Yea, that'd help :) |
| 17:48.18 | Rumbles | same error |
| 17:48.29 | WIMPy | ~pb |
| 17:48.29 | 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. |
| 17:48.56 | Rumbles | http://pastebin.com/BeGqmqWp |
| 17:49.06 | Greenlight | Do you have kernel-devel ? |
| 17:49.16 | Greenlight | Think you need that for the headers |
| 17:49.27 | Katty | why does asterisk flash green lights at me????? |
| 17:49.33 | Rumbles | damnit, thanks Greenlight |
| 17:49.51 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:44e0:7317:b5db:64aa) |
| 17:49.55 | Greenlight | Katty: It's saying "hello" ... |
| 17:50.01 | Rumbles | still bombed out with the same error |
| 17:50.05 | WIMPy | Do you need the xpp stuff? If not, disable it. |
| 17:51.07 | nuken | SIP/2.0 488 Not Acceptable Media, this is a codec error correct ? |
| 17:51.09 | Rumbles | no idea what it is tbh |
| 17:51.19 | Rumbles | googles it |
| 17:51.29 | WIMPy | xpp is for Astribank |
| 17:51.45 | Greenlight | mjordan: Damn, still getting hangups and failed bridges even with different codec and DTMF features. Could MixMonitor still be doing a soft hangup even under those circumstances ? |
| 17:51.52 | *** join/#asterisk igcewieling1 (~igcewieli@user-24-214-153-32.knology.net) |
| 17:51.55 | WIMPy | No need to buid drivers for hardware you don't have. |
| 17:52.00 | Rumbles | and how do I disable it? :) |
| 17:52.16 | Rumbles | no idea what Astribank is.. :) |
| 17:52.22 | mjordan | Greenlight: looks like it. Is the hangup occurring in a different place? |
| 17:52.28 | talntid | I make deposits into astribank :) |
| 17:52.29 | igcewieling1 | Has anyone experienced a problem where when you save a voicemail message the message stays in the INBOX? |
| 17:52.41 | WIMPy | Didn't dahdi have dome configuration thing? It's been a long time since I used it. |
| 17:53.00 | Rumbles | oh, if that's for asterisk-usb connectivity I need it, we have a Digium failover switch |
| 17:53.00 | Greenlight | mjordan: Still seems to be the same place, in that fuction. The error is the same, and it's when I try to bridge. |
| 17:53.21 | WIMPy | Rumbles: Does that use xpp? |
| 17:53.51 | igcewieling1 | when you do a "make config" in DAHDI it prints the command you can use to generate the file which only loads drivers for hardware you have |
| 17:54.11 | Rumbles | I have no idea, but on the page I'm looking at talking about astribank it shows a failover switch |
| 17:54.17 | Rumbles | let me check with my colleague |
| 17:54.19 | Rumbles | see if he knows |
| 17:54.41 | Greenlight | WARNING[16088]: rtp_engine.c:1471 ast_rtp_instance_bridge: Got hangup while attempting to bridge 'SIP/minotaur-00015064' and 'SIP/70273-0001430a' |
| 17:54.46 | mjordan | Hm. I wonder why it's still going into the native bridge. |
| 17:54.51 | Greenlight | Yea - same place |
| 17:55.16 | Greenlight | The SIP/minotaur trunk is forced ulaw, everything else is alaw |
| 17:56.30 | Greenlight | I also set/enable AudioHook inheritence, not sure if that could effect things |
| 17:56.43 | mjordan | nah |
| 17:56.55 | mjordan | there's several race conditions here that are kind of annoying. |
| 17:56.57 | mjordan | I'd do this: |
| 17:57.07 | *** join/#asterisk ziz212 (~chatzilla@61.245.172.27) |
| 17:57.17 | ziz212 | hi frends |
| 17:57.30 | mjordan | in rtp_engine, after the check for the hangups, check to see if the reason why you're getting hung up is due to the AST_SOFTHANGUP_UNBRIDGE flag |
| 17:58.06 | ziz212 | how can I write some thing in ring groups in dial plan |
| 17:58.14 | Greenlight | Does (ast_check_hangup return that reason ? |
| 17:58.14 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 17:58.33 | mjordan | nope, hold on :-) |
| 17:58.38 | Greenlight | Heh :) |
| 17:58.50 | Greenlight | Sorry, I'm painfully unfamiliar with the code |
| 17:59.28 | Greenlight | ziz212: Dial(SIP/200&SIP/201) ? |
| 17:59.32 | ziz212 | let say I need to write some funtionlaity when call hit a ring gorup.. how can I write that in dial plan. I mean how to point the exact location..or command. pls help me |
| 17:59.46 | mjordan | what version are you on? |
| 17:59.47 | ziz212 | let say I am haning ring roup 1000 |
| 17:59.50 | Greenlight | 11 |
| 18:00.06 | Greenlight | 11.0.1 to be precise |
| 18:00.47 | mjordan | kk |
| 18:00.53 | mjordan | I'll pastebin something in a second |
| 18:01.12 | igcewieling1 | ziz212: in extensions.conf in whatever context the call arrives in exten => 1000,1,Dial(SIP/peer1&SIP/peer2&SIP/peer3) This should be covered in the Asterisk Book |
| 18:01.15 | Rumbles | okay, spoken with the colleague, so the issue with xpp is down to an error in one of the files, if you comment it out it works... he knows this because he has already compiled dahdi-linux-complete from source... so dahdi is installed, but we have no channel driver, any idea how I can compile a chan_dahdi.so without compiling the entirity of asterisk? |
| 18:01.18 | ziz212 | it is 1.8.12 |
| 18:01.24 | kaldemar | ziz212: you're the one who should know the location in your dialplan. as for the commands, you must elaborate on the "thing". |
| 18:01.30 | Greenlight | I suppose it's fair to say that what i'm doing may be a niche case, I could I suppose fire my bridge request *after* I get the confirmatoin that mixmonitor has started, or vice versa. Although I'd like to fix the underlying issue if possible |
| 18:01.49 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.171.41) |
| 18:02.05 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 18:02.21 | ziz212 | Oh ... sorry for disturbing you all |
| 18:02.24 | WIMPy | Rumbles: No go. It needs to fit exactely. Either you install it from the same packet source or you do it yurself. |
| 18:02.25 | ziz212 | I got it |
| 18:03.28 | ziz212 | thanks guys to remembering basics... it is a shame to ask that question. |
| 18:03.33 | Rumbles | how come chan_dahdi.so isn't created as part of compiling dahdi? Is there no other way apart from compiling asterisk again? |
| 18:04.09 | ChannelZ | chan_dahdi is asterisk-side not dahdi-side |
| 18:04.15 | Rumbles | actually, I know the answer for that, dahdi can be used without asterisk and visa vesa... |
| 18:04.57 | WIMPy | indeed |
| 18:04.58 | Rumbles | so there is no way to create a valid chan_dahdi.so without compiling asterisk from source? |
| 18:05.27 | WIMPy | What way could there possibly be? |
| 18:05.37 | Rumbles | magic? :) |
| 18:05.50 | WIMPy | It's always from source. Either by yourself or someone else doing it for you. |
| 18:05.55 | Rumbles | true |
| 18:05.55 | WIMPy | Doesn't compute. |
| 18:06.26 | Rumbles | okay, thanks for all your help WIMPy :) |
| 18:08.24 | igcewieling1 | Asterisk is one of those programs where you can spend all your time fighting with it and be miserable or you can accept Asterisk's oddities and be happy. |
| 18:10.29 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:10.32 | igcewieling1 | Like my issue with "can't delete new messages". If I don't get a fix I'll keep downgrading until the problem goes away. |
| 18:11.19 | mjordan | igcewieling1: what's the ASTERISK issue? |
| 18:11.24 | ChannelZ | eh? |
| 18:11.50 | Greenlight | mjordan: The changes to try and prevent native bridgedo seem to have greatly reduced the frequency of the bridging issues, although it is still occuring, so I think we're on the right path |
| 18:12.11 | igcewieling1 | mjordan: this is a 2nd hand report. The user reports "when we delete voicemail 5 mins later the messages are back as new messages". I doubt it is really 5 mins, but there is some problem somewhere. |
| 18:12.21 | Greenlight | Although it's subjective |
| 18:12.23 | mjordan | Greenlight: yay progress. Compiling a patch now that should at least dump out some more info if it happens (and will try to avoid it) |
| 18:12.52 | ChannelZ | igcewieling1: are you using IMAP storage or something? |
| 18:12.53 | Rumbles | WIMPy, something my colleague just mentioned, can you do "make menu-select" or something along those lines and just compile parts of asterisk? i.e. just the channel driver I need? |
| 18:12.59 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 18:13.01 | igcewieling1 | ChannelZ: no. |
| 18:13.43 | WIMPy | Rumbles: yes, but all parameters must be exactely the same as for the Asterisk Version tou're using. |
| 18:13.47 | mjordan | Greenlight: http://pastebin.com/jYBaT4uW |
| 18:13.53 | Rumbles | oic, thanks :) |
| 18:14.19 | WIMPy | So unless you did it from the same source on the same system, you need some amount of luck. |
| 18:14.52 | Greenlight | Hmm if there is a soft hangup flag, do we still want to go to done, should the logic not be to continue ? |
| 18:15.08 | igcewieling1 | Asterisk's SRV support is a perfect example of "go with the flow, don't fight it" sort of thing. 8-| |
| 18:15.56 | Greenlight | Or can the rest of that function be skipped ? |
| 18:16.23 | Greenlight | it looks like it does some useful stuf |
| 18:16.28 | din3sh | I have a 30-channel E1, I have this occasional problem where calls to any DID/number on this E1 does not even reach the * box. PSTN Caller hears an announcement from telco saying number is unavailable. ] |
| 18:16.29 | din3sh | Also I get HDLC errors on the span, are these 2 problems related? how to test/troubleshoot? |
| 18:17.14 | WIMPy | din3sh: I answered that one twice. |
| 18:18.09 | Greenlight | Like: http://pastebin.com/zT37tKpV ? |
| 18:21.13 | mjordan | well... most of that function is trying to set up a native bridge. |
| 18:21.35 | mjordan | If we're trying to bail out of that, it's all not needed - in fact, there is code that explicitly prevents trying to get into this function when there are audiohooks in the first place |
| 18:22.22 | mjordan | if you continue on, then *hopefully* the local/remote native bridges will get the message that you need to be broken anyhow |
| 18:23.46 | Greenlight | Ahh - so this function is *only* to setup a native (packet-to-packet?) bridge? |
| 18:24.19 | Greenlight | So, where abouts would the normal bridge be setup |
| 18:24.22 | mjordan | yup |
| 18:24.24 | mjordan | features.c |
| 18:24.32 | mjordan | (the bridging code is ... interesting) |
| 18:24.45 | mjordan | and, to some extent, channel.c |
| 18:24.55 | Greenlight | Right, so ast_bridge_call |
| 18:25.18 | *** join/#asterisk scubes13 (~scubes13@39.sub-70-193-15.myvzw.com) |
| 18:25.21 | mjordan | the problem is there's lot of kinds of bridges. There's the Asterisk managed kind (features), then channel specific (channel) which acts as a wrapper around the technology specific bridges (RTP, DAHDI, etc.) |
| 18:25.39 | mjordan | preference is to use the technology specific as much as possible, since there's less stuff in the way interpreting packets |
| 18:26.03 | Greenlight | Right, so that's what it's doing here: res = ast_channel_bridge(chan, peer, config, &f, &who); |
| 18:26.11 | Greenlight | That's tryuing to use the tech-specific bridge ? |
| 18:26.32 | Greenlight | In this instance, the rtp bridge function |
| 18:27.58 | mjordan | correct, there's a block of code in ast_channel_bridge that attempts to set up the native technology bridge |
| 18:28.07 | Greenlight | But if we return AST_BRIDGE_COMPLETE won't it think that a native bridge has worked, and not bother trying a normal bridge |
| 18:28.14 | mjordan | nope |
| 18:28.21 | mjordan | It shouldn't |
| 18:28.22 | mjordan | :-) |
| 18:28.25 | Greenlight | "interesting" you said :) |
| 18:28.37 | mjordan | it's probably one of the most complex things Asterisk does |
| 18:29.23 | Katty | ^- other than making toast. |
| 18:31.28 | mjordan | Katty: totally possible if you have a VoIP enabled toaster. |
| 18:31.32 | Greenlight | Oh, I see it's actually passing "res" back up to the caller |
| 18:31.44 | Greenlight | In features.c |
| 18:32.13 | Katty | http://www.linuxscrew.com/wp-content/uploads/2007/11/pdrm0388.JPG <- it WILL work, eventually. |
| 18:32.30 | mjordan | Katty: awesome |
| 18:32.35 | Greenlight | Wow |
| 18:33.01 | WIMPy | lame |
| 18:33.04 | Katty | that's not my photo. i can't take the credit. |
| 18:33.10 | WIMPy | Built in toaster is much better. |
| 18:34.21 | Katty | i wonder if they make voip toasters. |
| 18:34.39 | WIMPy | 5ΒΌ" toaster |
| 18:34.46 | WIMPy | And pizza slice |
| 18:34.53 | Greenlight | Might get a burnt ear |
| 18:35.34 | Katty | hmm. |
| 18:35.39 | Katty | hmm |
| 18:35.45 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:35.45 | Katty | i'm going to build one |
| 18:35.52 | Katty | with a raspberry pi |
| 18:36.07 | WIMPy | http://www.google.com/images?q=acorn+rocketship&hl=en&sa=X&oi=image_result_group |
| 18:36.14 | WIMPy | That's what the 90s looked like. |
| 18:37.07 | Greenlight | mjordan: So, the only thing I'm not fully understanding is that *if* the native brigdge hasn't happened yet, why is MixMonitor issueing a softhangup ? |
| 18:38.59 | Greenlight | Prior to the bridging the channel that's getting MixMonitor'd is briged to a Local/ channel |
| 18:39.19 | mjordan | I suspect what's happening is that the bridge NFLAG is getting set prior to it going very far into that loop |
| 18:39.24 | mjordan | AST_FLAG_NBRIDGE |
| 18:39.43 | *** join/#asterisk volga629 (~volga629@host7.pythian.com) |
| 18:39.56 | mjordan | and that is what happens on line 8040 in channel.c |
| 18:40.47 | Greenlight | I've a different line 8040, what's the text ? |
| 18:40.56 | mjordan | ast_set_flag(ast_channel_flags(c0), AST_FLAG_NBRIDGE); |
| 18:41.03 | mjordan | whoops, sorry, looking at trunk |
| 18:41.26 | Greenlight | Got it now |
| 18:42.20 | mjordan | so, my guess is the race condition goes something like this: a pbx_thread checks for audiohooks and other stuff, finds none, and sets the NBRIDGE flag. Context switch to MixMonitor (on a different thread) who puts the audiohooks on the channel (missed a race there), sees the NBRIDGE flag (race) and sends the softhangup. Context switch back to pbx_thread, enter into the rtp_engine callback for the bridge, who sees that the channel has been |
| 18:43.56 | Greenlight | Ahh cause the channels aren't locked until inside the rtp_engine callback |
| 18:43.57 | Greenlight | Okay |
| 18:45.21 | *** join/#asterisk jkroon (~jkroon@41.13.4.245) |
| 18:45.59 | Greenlight | But that NBRIDGE flag shouldn't be getting set if different codecs, right? |
| 18:48.04 | mjordan | the codec check unfortunately happens in the rtp_engine portion of the native bridge setup, after it checks for hangup |
| 18:48.15 | mjordan | IIRC, anyway :-) |
| 18:48.23 | Greenlight | Yea, just noticed that it compares tech in channel.c, but not codec |
| 18:49.17 | Greenlight | This also answeres why my "test" customer using DAHDI had everything working |
| 18:49.43 | Greenlight | And: |
| 18:49.44 | Greenlight | !ast_channel_monitor(c0) && !ast_channel_monitor(c1) && |
| 18:50.40 | Greenlight | Okay - going to apply that patch this evening, and hopefully that sorts things |
| 18:52.08 | mjordan | good luck! There's at least several nasty race conditions in this whole thing, so it'd be great to get it fixed in a release |
| 18:52.17 | Greenlight | Might also swap around the order I request things via AMI; at present I request the Bridge, then immediately after request MixMonitor. |
| 18:52.26 | mjordan | please let me know how it goes |
| 18:52.27 | mjordan | ah |
| 18:52.40 | mjordan | MixMonitor before bridge would probably alleviate some of it |
| 18:53.04 | Greenlight | Still potentially a race issue, depending how quick it gets that audiohook in there |
| 18:53.59 | Greenlight | But at a guess the call to MixMonitor will lock the channel fairly quickly, and start doing it's thing? |
| 18:54.16 | Greenlight | Fingers crossed... will let you know how it goes tomorrow! |
| 18:54.19 | Greenlight | Thanks again |
| 18:55.25 | iztech | since last night i was able to get outgoing and incoming calls working, but there is no audio coming, only going out - any obvious clues? |
| 18:55.42 | Greenlight | NAT/firewall |
| 18:56.25 | iztech | yes its behind nat |
| 18:56.29 | mjordan | Greenlight: yes to the first question, and good luck |
| 18:56.52 | iztech | sorry abt that |
| 19:06.15 | *** join/#asterisk greenwolf (c087e3a3@gateway/web/freenode/ip.192.135.227.163) |
| 19:06.59 | greenwolf | sup everyone... |
| 19:15.04 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:15.29 | iztech | hey guys Sipstation has stated that it would work without a static IP, from what i am reading this could be the issue. Audio is going but NO audio in. will bindaddr=0.0.0.0 help am i on the right track |
| 19:28.10 | greenwolf | put your private IP there |
| 19:28.19 | greenwolf | and also use bindport=5060 |
| 19:28.36 | greenwolf | externip={outside IP here} also |
| 19:29.50 | iztech | that all goes in the [general] section? |
| 19:30.19 | *** join/#asterisk ks3 (ks3@alderaan.digitallotus.com) |
| 19:35.39 | igcewieling1 | I never recommend using a bindip |
| 19:36.38 | igcewieling1 | iztech: one of the the best ways to screw up your asterisk is to use bindaddr, don't use it at all |
| 19:37.34 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:37.49 | *** join/#asterisk greenwolf_ (c087e3a3@gateway/web/freenode/ip.192.135.227.163) |
| 19:38.03 | iztech | well Sipstation states that this works without a static IP, that's why I paid for it - but they offer no support, nor do they provide any sample sip.conf - so i have been mucking around for 4 days or so |
| 19:38.35 | igcewieling1 | iztech: is the server behind nat? |
| 19:38.46 | iztech | yes |
| 19:38.58 | iztech | 10.0.1.33 |
| 19:39.06 | igcewieling1 | you need externip localnet and directmedia=no |
| 19:39.32 | igcewieling1 | you also need to disable any SIP ALG on your router and should set qualify=yes |
| 19:39.35 | iztech | in [general] i have directmedia=no |
| 19:39.48 | igcewieling1 | iztech: and localnet and externip? |
| 19:40.16 | iztech | qualify=yes is set |
| 19:40.59 | iztech | i did not haver externip |
| 19:41.00 | igcewieling1 | <PROTECTED> |
| 19:41.20 | igcewieling1 | localnet= would specify your INTERNAL network and netmask |
| 19:41.32 | igcewieling1 | put in everything and try again |
| 19:41.56 | iztech | yeah i don't have localnet |
| 19:43.34 | iztech | k, let me try |
| 19:43.54 | iztech | thx - you guys have been super helpful - lets hope it works |
| 19:45.53 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 19:47.14 | iztech | k, so i have local, directmedia=no, qualify=yes, and i should remove bindaddr? |
| 19:51.28 | igcewieling1 | <PROTECTED> |
| 19:52.34 | iztech | k, got it |
| 19:58.26 | iztech | still no love |
| 19:58.55 | iztech | so localnet is 10.0.1.1/255.255.255.0 correct? |
| 19:59.45 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 20:01.12 | mathi | hey |
| 20:01.42 | mathi | is it possible to playback a sound like it dials ? |
| 20:02.07 | mathi | somehow simulate a dialing |
| 20:03.03 | *** join/#asterisk greenwolf (c087e3a3@gateway/web/freenode/ip.192.135.227.163) |
| 20:03.14 | navaismo | you mean send dtmf or the ring or the tone |
| 20:03.27 | navaismo | ? |
| 20:03.31 | igcewieling1 | iztech: try 10.0.1.0/255.255.255.0 assuming that is your internal network |
| 20:03.50 | mathi | not dtmf, the tone I guess |
| 20:04.21 | iztech | router is 10.0.1.1 |
| 20:04.36 | igcewieling1 | iztech: localnet is a NETWORK not a host |
| 20:04.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:04.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:05.02 | igcewieling1 | mathi: something like Playtones(dial) ? |
| 20:05.14 | iztech | k, thx |
| 20:06.02 | mathi | i'll try thank you |
| 20:10.59 | *** join/#asterisk doctorray (~ray@72.26.99.19) |
| 20:11.33 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 20:12.23 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:13.33 | iztech | didn't work |
| 20:13.42 | doctorray | Is there a level of function VOLUME() that would be considered "muted"? I'm trying to see if I can duplicate MeetMe's "talk only" feature in the new ConfBridge without code patching. |
| 20:13.46 | iztech | let me try some port mapping |
| 20:13.48 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:14.49 | WIMPy | Hmm. Some comfort clicks when dialling wouldn't be bad. |
| 20:18.18 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-187-53.red.bezeqint.net) |
| 20:19.40 | dr0ck | doctorray: theres MUTEAUDIO() |
| 20:20.46 | doctorray | dr0ck: oh! that should work.. looks like you can even select the direction |
| 20:21.13 | greenwolf | has anyone used Erlang language with asterisk |
| 20:21.36 | greenwolf | agi scripting with erlang by any chance anyone? i would imagine it would be awesome for its use |
| 20:21.55 | doctorray | dr0ck: now I just need to figure out how to call that through ami. back to reading docs |
| 20:22.20 | iztech | igcewieling1: thanks so much - with the port forwarding it worked |
| 20:22.53 | iztech | this channel is amazing, thanks for helping out a n00b |
| 20:23.44 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 20:23.44 | greenwolf | iztech: great ppl here and were all willing to help out as long as your willing to learn and get knowledge by reading on your own also |
| 20:24.36 | iztech | yeah did a lot of reading, hopefully i can post something on net about sipstation, usually its super easy to configure when you have freepbx |
| 20:25.11 | iztech | but i am running it on a server with zoneminder and so i just installed asterisk 11 |
| 20:25.16 | greenwolf | are you using freepbx still or have you switched completely to vanilla asterisk? |
| 20:25.30 | iztech | no freepbx on this machine |
| 20:25.35 | iztech | vanilla |
| 20:27.30 | iztech | there aren't too many sip providers that work with non static ips, i was able to set up call centric on this a lot easier than sipstation |
| 20:28.04 | iztech | i would have used phonebooth but you have to have static ips with them |
| 20:29.10 | iztech | still have to get vm working on it |
| 20:33.12 | rrittgarn | anybody ever have channels that refuse to hangup? I have about 6 channels throwing errors trying to playback but I cannot end them. |
| 20:38.00 | ChannelZ | What flavor? |
| 20:40.23 | rrittgarn | SIP channels asterisk 10 |
| 20:40.59 | *** join/#asterisk Natureshadow_ (nik@shore.naturalnet.de) |
| 20:42.40 | ChannelZ | Have you done any SIP debugs? Do you have many Retransmits, or notice peers not responding to BYEs (or are they sending them but Asterisk is not responding?) |
| 20:43.08 | rrittgarn | wasn't able to get into the console... looks like there was a loop in the dialplan that kept calling playback |
| 20:43.33 | rrittgarn | the console was moving faster than i could issue commands, i ended up using asterisk -rx '' to try and end the channel that was looping |
| 20:43.49 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:44.16 | rrittgarn | whats weird is the channel that was looping was a channel that was hung up. Like i did a test call into the box, heard my playback() and hung up before pressing any options... it then tried again and again on a dead channel |
| 20:45.37 | ChannelZ | It kept executing the same priority of the extension (your Playback?) |
| 20:48.54 | rrittgarn | yeah |
| 20:49.26 | igcewieling1 | rrittgarn: is any leg of the call POTS FXO? |
| 20:49.35 | rrittgarn | i eventually got it to stop by pulling the playback from the dialplan and reloading vai asterisk -rx |
| 20:49.52 | rrittgarn | igce: no, was a call in from my SIP Trunking Provider |
| 20:50.12 | rrittgarn | My system has dropped back down to idle, but it was still very odd |
| 20:51.19 | ChannelZ | hmm. Can't say I've ever heard of that before. |
| 20:54.45 | rrittgarn | on an unrelated note: What's the best way to match on a number with and without a leading + ? Trying to combine two lines into one if possible in my incoming context for say: 18009001234 and +18009001234. Possible or no? |
| 20:55.37 | rrittgarn | i would assume exten => _.18009001234 but that seems like it could be problematic (see semi-unrelated situation above) |
| 20:55.44 | WIMPy | You can't. Use Goto(). |
| 20:56.05 | rrittgarn | was afraid of that.... thanks |
| 20:56.07 | ChannelZ | If only we had regex extensions! |
| 20:56.13 | WIMPy | And you can't have anything after a ".". |
| 20:56.39 | WIMPy | But if you have lots of them you can use patterns and Goto or switches. |
| 21:09.43 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 21:15.38 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 21:16.05 | *** join/#asterisk Tarcert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 21:16.09 | saint_ | hi all... |
| 21:16.41 | iztech | guys should this ring both the extension and the cell phone at the same time exten => _NXXNXXXXXX,1,Dial(SIP/101&SIP/VOIP_Provider/*1234567890,150,r,t) |
| 21:16.47 | saint_ | when I read the documentation, I can see that some Asterisk version are "cert". What does it mean ? If I want to use DPMA , do I need a "cert" version ? Finally, how do I install it with svn if I need it ? |
| 21:17.09 | iztech | not working for me it is only ringing the extension |
| 21:18.22 | WIMPy | iztech: *123... doesn't look like a valid number. And you have one parameter too much. |
| 21:19.04 | iztech | 123-456-7890 |
| 21:19.19 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 21:19.27 | iztech | which parameter? |
| 21:19.41 | WIMPy | Dial only has 3. |
| 21:22.10 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 21:23.02 | iztech | thx WIMPy |
| 21:23.04 | iztech | fixed |
| 21:23.13 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 21:24.32 | *** join/#asterisk Rumbles (~jstocker@212.183.128.249) |
| 21:25.13 | Rumbles | thnks for the help earlier WIMPy, managed to get dahdi working by recompiling asterisk... just now cepstural isn't working :) |
| 21:25.56 | WIMPy | Is that the name of some provider? |
| 21:26.33 | Rumbles | sorry cepstral, text-2-voice :) |
| 21:27.35 | Rumbles | I'm sure it's a minor issue I can figure out when I have slept :) |
| 21:31.11 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:32.36 | mathi | to get inbound sip calls, I used "register => ..." and created a new context in sip.conf, there I specify the context to enter in extensions.conf |
| 21:32.53 | mathi | but what should be the name of the first extension in that context ? |
| 21:37.15 | igcewieling1 | mathi: perhaps you should read the Asterisk Book. |
| 21:37.26 | mathi | ive read |
| 21:42.41 | ChannelZ | It depends on your provider. |
| 21:43.22 | ChannelZ | Probably they send to an extension matching your DID. They might let you specify it yourself when you register. |
| 21:43.28 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:43.28 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:48.44 | mathi | thanks i'll ty |
| 21:54.01 | *** join/#asterisk schultza (~schultza@rc1.rcherbals.com) |
| 21:54.13 | schultza | how do i rebuild (from source) asterisk after it's already rebuilt/installed? |
| 21:55.40 | schultza | nevermind |
| 22:00.05 | ChannelZ | yes exactly |
| 22:01.08 | schultza | no ... i forgot that it was running... it didnt like reconfiguring for some reason |
| 22:08.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:10.52 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:13.52 | saint_ | I had a version not CERT from asterisk. When I added a rpm (like jabber), i would just need to run make menuselect and make install.. menuselect would find my stuff and enable it by defaut. |
| 22:14.12 | saint_ | With the CERT version that I just installed, I am realizing that all the stuff that were enabled by default before have now to be enabled by hand. |
| 22:14.36 | saint_ | Is there any way to keep it automatic ? As per : Asterisk sees that a package is here, so it enables it automatically in menuselect ? |
| 22:16.39 | *** join/#asterisk mathi (3eebd673@gateway/web/freenode/ip.62.235.214.115) |
| 22:25.36 | mathi | I try to get SIP calls but I got error when someone tries to call me: |
| 22:25.41 | mathi | http://pastebin.com/raw.php?i=6kkQ0YKD |
| 22:25.47 | mathi | can anyone have a look ? |
| 22:27.06 | [TK]D-Fender | MatShow complete SIP debug and CLI for the call |
| 22:27.34 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 22:29.36 | *** join/#asterisk amadmin (~chatzilla@41.58.6.205) |
| 22:31.45 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 22:32.38 | mathi | http://pastebin.com/raw.php?i=R1cQDfZm |
| 22:32.44 | mathi | [TK]D-Fender: ^ |
| 22:33.08 | mathi | somewhere there is "username mismatch, have <VoIPProviderInbound>, digest has <anonymous>" |
| 22:34.11 | amadmin | Good Day people |
| 22:34.37 | amadmin | please any ideas how i can get HD video conferencing in Asterisk |
| 22:34.39 | amadmin | ?? |
| 22:35.00 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 22:35.39 | *** join/#asterisk amachefe (293a06cd@gateway/web/freenode/ip.41.58.6.205) |
| 22:35.49 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 22:37.08 | navaismo | confbridge can make videoconference but no HD |
| 22:38.25 | [TK]D-Fender | mathi: Sending to 91.121.129.20:5060 (NAT) <- first, providers almost NEVER behind NAT so go fix your peer first and maks sure you don't have multiple witht he same IP/host |
| 22:40.49 | amadmin | yeah @navaismo |
| 22:41.15 | amadmin | i need to provide HD, any other suggestions. Diastar seems not to support HD too |
| 22:44.56 | mathi | [TK]D-Fender: how do I do this ? |
| 22:49.13 | jeev | has anyone encountered a call coming in over PRI or SIP, within 14 seconds, it drops the call and comes back, the caller never notices that the PBX has dropped the call, call comes back in and goes through the ring group in full this time. |
| 22:50.27 | [TK]D-Fender | mathi: .... nat=no |
| 22:52.57 | mathi | [TK]D-Fender: I have put nat=no but I still have "Sending to 91.121.129.20:5060 (NAT)" |
| 22:54.56 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:55.40 | mathi | [TK]D-Fender: I changed friend to peer and now I have: Sending to 91.121.129.20:5060 (no NAT) |
| 22:55.41 | [TK]D-Fender | mathi: And I don't see where you put things or know that you've applied changes, etc |
| 22:55.44 | mathi | is this better ? |
| 22:55.58 | [TK]D-Fender | Already, yes |
| 22:56.01 | [TK]D-Fender | test |
| 22:57.21 | mathi | [TK]D-Fender: I still have that same error: username mismatch, have <VoIPProviderInbound>, digest has <anonymous> ... |
| 22:58.52 | talntid | i love how when a voip provider has an issue... i get calls from all sorts of asterisk people, saying their system stopped working, so they made <insert change here>, and it still isn't working... |
| 22:59.16 | talntid | instead of thinking "hmm, I havn't changed anything in 2 years, and it stopped working." ... might be the carrier? :P |
| 23:00.14 | mathi | [TK]D-Fender: it still does one time (NAT) then it does (no NAT), strange: http://pastebin.com/raw.php?i=3qn7vpks |
| 23:01.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.42 | [TK]D-Fender | mathi: If you're already authing by IP/host perhaps your provider isn't passing standard auth with the cal. "isecure=port,invite" <- |
| 23:02.23 | mathi | [TK]D-Fender: I want to receive any number, what is this auth thing about ? |
| 23:03.04 | [TK]D-Fender | * doesn't trust them as being your provider. They are failing to authenticate |
| 23:03.13 | [TK]D-Fender | mathi: Many don't send authed calls at all |
| 23:06.19 | mathi | [TK]D-Fender: WOW it works with insecure=port,invite. What about this (NAT) and (no NAT) issue, should I care ? |
| 23:07.12 | [TK]D-Fender | mathi: Because it should auth immediately it certainly shouldn't |
| 23:07.23 | *** part/#asterisk igcewieling1 (~igcewieli@user-24-214-153-32.knology.net) |
| 23:07.37 | [TK]D-Fender | The first response was a challenge before and because they weren't accepted yet the peer's rules didn't apply to the challenge sent to them |
| 23:09.33 | mathi | [TK]D-Fender: dude I'm sorry it's like I am reading Chinese |
| 23:16.07 | mathi | [TK]D-Fender: do you know what this means ? Agent policy for SIP/VoIPProviderInbound-00000000 is 'never'. CC not possible |
| 23:16.19 | mathi | I have this message several times during the call |
| 23:16.36 | [TK]D-Fender | MatDon't actually know that one. |
| 23:20.08 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
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| 23:22.21 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-myraolntzgqzpjkw) |
| 23:27.58 | jeev | http://pastebin.ca/2306125 any idea what's going on there? there are notes there starting line 158. thanks. |
| 23:28.06 | *** join/#asterisk amadmin (~chatzilla@41.58.11.17) |
| 23:28.35 | *** join/#asterisk minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 23:31.42 | amadmin | hello |
| 23:32.05 | amadmin | any suggestion of a product that can do tht |
| 23:32.28 | *** join/#asterisk amachefe (293a0b11@gateway/web/freenode/ip.41.58.11.17) |
| 23:33.13 | ChannelZ | tht? |
| 23:33.25 | [TK]D-Fender | jeev: probably due to the fact you aren't answering the call and your hunting around while not getting an answer is causing TIMEOUTS to the caller's end |
| 23:34.23 | jeev | fender, it's 14 seconds though, the caller is my cell, no? what's the phone i'm using have to do with it, it's 14 seconds into the call |
| 23:34.51 | amadmin | @channel, i need to add HD video conference to asterisk |
| 23:35.07 | jeev | so that pastebin was someone answer it, if they didn't, it would've gone all the way through to the voicemail after the hunt was completed. |
| 23:35.16 | amadmin | i have tried Diastar, which doesnt seen to support it |
| 23:36.38 | [TK]D-Fender | amathere isno split-screen in * |
| 23:36.46 | [TK]D-Fender | amadmin: there isno split-screen in * |
| 23:37.35 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 23:38.22 | amadmin | D-Fender: I am aware if that, Diastar does provide that. But there is no HD video provided |
| 23:42.42 | [TK]D-Fender | amadmin: * also doesn't transcode. The best you can get is "follow the speaker as long as everyone is using the same codec otherwise DOA |
| 23:42.57 | [TK]D-Fender | adaAnd the best * supports is H.264 in passthrough |
| 23:45.30 | *** join/#asterisk amadmin (~chatzilla@197.242.108.215) |
| 23:45.39 | *** part/#asterisk amadmin (~chatzilla@197.242.108.215) |
| 23:45.41 | *** join/#asterisk amadmin (~chatzilla@197.242.108.215) |
| 23:45.58 | [TK]D-Fender | heads out for a few hours |
| 23:46.49 | amadmin | thanks D-Fender.. |
| 23:47.50 | amadmin | but i will need a mixture of codec |
| 23:52.50 | navaismo | not with asterisk |
| 23:52.56 | navaismo | you mnay check polyvom solutions |
| 23:53.13 | navaismo | may check polycom solutions* |
| 23:59.48 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 23:59.56 | jeev | navaismo, you have any idea what my issue could be ? http://pastebin.ca/2306125 |
