| 00:01.17 | mlsmith9999 | tyman: I know that getting it through NAT can be a bit#$ |
| 00:18.41 | pabelanger | mlsmith9999: pb your sip.conf |
| 00:19.19 | mlsmith9999 | pabelanger: The whole hing? |
| 00:19.24 | mlsmith9999 | *thing? |
| 00:19.45 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 00:19.54 | pabelanger | your general section, plus settings for SIP/2468888 |
| 00:20.59 | pabelanger | mlsmith9999: your issue is you are getting SIP/2.0 403 Forbidden from your ITSP |
| 00:21.09 | mlsmith9999 | yup.. |
| 00:22.02 | pabelanger | 21.4.4 403 Forbidden |
| 00:22.02 | pabelanger | <PROTECTED> |
| 00:22.02 | pabelanger | <PROTECTED> |
| 00:22.19 | *** join/#asterisk Bartok (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 00:22.30 | Bartok | hey folks |
| 00:23.34 | pabelanger | mlsmith9999: you will have to contact your ITSP, and explain the problem |
| 00:23.43 | Bartok | I have an voice issue - once way SIP traffic, Asterisk 1.4.31, |
| 00:24.21 | genxweb | Bartok most likey you are behind a ant and need to open the ports and edit the sip nat file |
| 00:24.36 | mlsmith9999 | pabelanger: ok.. will do, that's what I needed.. |
| 00:24.45 | Bartok | when I call my PSTN number, my Asterisk box answers, but I hear no sound until the voicemail picks up |
| 00:25.33 | p3nguin | bartok: What happens when you do an echo test? |
| 00:26.04 | pabelanger | mlsmith9999: you may also need to setup a register statement. EG: register => user:password@your_itsp.com |
| 00:26.10 | Bartok | I guess I have not tried that yet |
| 00:26.14 | pabelanger | To register your asterisk box properly |
| 00:27.02 | Bartok | if I use it to dial out, it works fine - I get two-way SIP traffic then |
| 00:27.03 | mlsmith9999 | pabelanger: yeah been trying to do that.. can you tell me more on that? |
| 00:27.21 | Bartok | okay - this echo test - some reference somewhere?? |
| 00:27.33 | p3nguin | pabelanger: You think that ITSP requires registration before authentication will work? |
| 00:27.56 | pabelanger | mlsmith9999: http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/sip.conf.sample <- look for register => |
| 00:28.11 | mlsmith9999 | pabelanger: thanks. |
| 00:28.38 | pabelanger | p3nguin: Honestly no, hence the SIP 403 message (not SIP 401), but could also be an issue on the ITSP side |
| 00:28.43 | p3nguin | While uncommon for such a requirement to exist, it does show up from time to time. |
| 00:28.46 | pabelanger | p3nguin: don't hurt to try |
| 00:29.45 | genxweb | I have setup a custom context in my extensions.conf file to play a message on outbound calls. Then I setup a local trunk to point to that context and pass the number to be dialed with it. The issue I am having is even though the route uses the local one and then the secondary fall back (AKA the real trunk) it only tries the first trunk plays the message and fails with auto failthrough local |
| 00:29.45 | genxweb | unkown status and never fails to the actual trunk to call out. If I manually define a dial staement and defien the trunk it works. what do i need to do to gt it to just use teh default macro for the trunk. |
| 00:31.24 | p3nguin | ~freepbx |
| 00:31.24 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 00:31.25 | p3nguin | genxweb: ^^^^^^ |
| 00:32.29 | genxweb | yeah there are as helpful in there as um never mind wont say |
| 00:35.00 | mlsmith9999 | pabelanger: is this what your talking about: register=<omitted>@nsdaq.org:<omitted>:<omitted>@las-obproxy.commpartners.us/<omitted> |
| 00:35.29 | pabelanger | mlsmith9999: Ya, something along that lines. register => |
| 00:35.32 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 00:35.53 | mlsmith9999 | ok, yeah that's what they sent me to put into the sip.conf. |
| 00:36.29 | pabelanger | their you go |
| 00:37.19 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 00:39.54 | dev_ast | mlsmith999 make sure that you also define the authentication digest (page 103 in the Asterisk The Future of Telephony 2nd Edition) |
| 00:41.25 | dev_ast | otherwise your asterisk box will throw 403 error to your ITSP ( i believe) |
| 00:42.42 | mlsmith9999 | I actually have that book.. O'REILLY.. love em |
| 00:43.04 | mlsmith9999 | wrong edition though... first. |
| 00:43.37 | dev_ast | there are some minor errors however good book for to start with |
| 00:45.30 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:48.54 | *** join/#asterisk pepselap (~pepse@ip68-109-163-65.ph.ph.cox.net) |
| 00:49.57 | pepselap | any of you guys tried out this PBXInAFlash? |
| 00:50.45 | tyman | jitu6757 |
| 00:50.52 | p3nguin | pepselap: Several people have. |
| 00:51.24 | pepselap | it doesn't want to download the purple_load.tar.gz.. Network is configured and working.. I tried both the 64bit and 32bit isos.. |
| 00:51.38 | pepselap | Google doesn't show anyone else having this problem :) |
| 00:51.45 | pepselap | not sure if it's some mirror that's down right now or what |
| 00:51.56 | p3nguin | We have no way to know. |
| 00:52.17 | pepselap | yeah, there's no log of where it's trying to get it from |
| 00:52.25 | pepselap | and the 'piafdl' is a binary |
| 00:54.14 | p3nguin | pepselap: If you don't already have a lot of time devoted to it, dump it and get AsteriskNOW. |
| 00:54.32 | pepselap | i'm installing it in an esx vm, tho, so i'm not sure if that's screwing with anything. i wouldn't think so. |
| 00:54.32 | p3nguin | PiaF really sucks. |
| 00:54.37 | pepselap | does AsteriskNow come with 1.8? |
| 00:54.38 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 00:54.52 | p3nguin | Asterisk 1.8? I don't know. |
| 00:55.14 | p3nguin | I bet you could ask in the AsteriskNOW channel and they could tell you. |
| 00:55.19 | pepselap | I'd like to see the GV stuff in 1.8 in action (according to some blogs about PIAF) |
| 00:55.58 | p3nguin | PiaF needs to be renamed to PitA. |
| 00:56.24 | pepselap | I suppose I could just grab 1.8 source and see if it's easy to set up the GV stuff |
| 00:56.29 | p3nguin | It's really bad. It's like a box of Lincoln logs. |
| 00:56.47 | pepselap | yeah, really all the freepbx "distros" are |
| 00:56.59 | p3nguin | AsteriskNOW isn't. I had great success with it. |
| 00:57.41 | p3nguin | Asterisk from source is far less trouble than trying to figure out how to fix PiaF. |
| 00:59.05 | pepselap | it's hard to find example configs for GV in 1.8 with all the damn blog posts about how it's been released with GV and IPV6 support |
| 00:59.39 | p3nguin | I understand. https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 01:00.55 | pepselap | cool thanks, i'll mess with that. |
| 01:01.40 | x86 | cdr_odbc.c:147 odbc_log: Unable to retrieve database handle. CDR failed. |
| 01:01.45 | x86 | what's this mean? |
| 01:02.34 | x86 | I'm trying to setup CDR logging via ODBC... I've setup the system DSN with unixODBC / libmyodbc, and setup cdr_odbc.conf to talk to that DSN with the appropriate credentials / etc... |
| 01:02.50 | tyman | p3nguin: what is PiaF? |
| 01:03.12 | p3nguin | tyman: PBX in a Flash |
| 01:03.24 | tyman | oh, ok |
| 01:03.26 | p3nguin | tyman: And it sucks very much bad. |
| 01:03.32 | ChannelZ | or Pissing In A Funnel |
| 01:03.34 | tyman | i see that |
| 01:03.42 | x86 | ChannelZ++ |
| 01:03.43 | x86 | ;) |
| 01:04.29 | p3nguin | tyman: After you install it, you have to run all these stupid scripts that download files that should be included already and they configure things that you should have better tools to configure, etc. It's just crappy generally. |
| 01:04.59 | tyman | sounds like ass |
| 01:05.23 | p3nguin | AsteriskNOW is far superior. You can have it up and running inside of 20 minutes. |
| 01:06.30 | p3nguin | I think it took about 15 minutes from booting the CD to the first boot off the hard disk. |
| 01:07.43 | p3nguin | But, of course, if you don't need an out-of-the-box platform with FreePBX, go pure Asterisk and forget all the other crap. |
| 01:08.25 | tyman | I'm running ubuntu and using the distro pkgs |
| 01:08.41 | tyman | having major problems with my SoB ATT u-verse router/fw....shows ports opened but port scans from outside and a tcpdump watching on the host prove otherwise... combine this with being brand new to asterisk and you've got a worthless weekend |
| 01:09.11 | p3nguin | Remember that SIP and RTP ports are UDP. |
| 01:10.00 | tyman | i know...got udp 5000-5200 and 10000-2000 all open for sip/rtp |
| 01:10.28 | tyman | should respond to ports shown listening on my box via lsof -i |
| 01:10.54 | p3nguin | If Asterisk isn't configured to listen, lsof isn't going to show it. Paste your configs in the pastebin. |
| 01:10.57 | tyman | piece of crap web firewall |
| 01:11.26 | tyman | i'll repaste...have pasted in here twice now...repasting now |
| 01:11.44 | tyman | repost: I'm having problems making inbound calls from the pstn thru my sip provide. I'm just starting with asterisk and i'm almost certain that the problem should be right here in my gist https://gist.github.com/666849 |
| 01:12.03 | tyman | <mynum> and <omitted> had legit info removed for obvious reasons |
| 01:12.55 | *** join/#asterisk sbszulu (~dundubala@41.14.47.118) |
| 01:14.15 | p3nguin | Did you start asterisk? |
| 01:14.35 | tyman | i'm making calls out to the pstn, and between extensions |
| 01:14.42 | tyman | just not in |
| 01:14.59 | tyman | shows sip peering with provider in console |
| 01:15.04 | p3nguin | Your sip.conf has several issues. |
| 01:16.07 | p3nguin | Several of your general settings are listed in the ext-sip-account peer definition. |
| 01:18.40 | pabelanger | pepselap: GV and Asterisk 1.8 is pretty easy. Also works great |
| 01:18.50 | x86 | i'm using it right now |
| 01:19.01 | x86 | actually on a phone conversation with a polycom IP601 via GV |
| 01:19.01 | p3nguin | tyman: This is closer to what I would want to see on your system: http://pastebin.com/irAPfHZ7 |
| 01:19.02 | x86 | ;) |
| 01:19.02 | pepselap | nice, good to know. |
| 01:19.31 | tyman | p3nguin: checking it out...thx |
| 01:19.38 | pepselap | pabelanger: have you noticed if there's any more or less of a delay than with using something like gizmo or a cellphone? |
| 01:20.06 | pabelanger | pepselap: not that I have noticed. |
| 01:20.17 | pepselap | same dilly, then :) |
| 01:21.09 | pabelanger | My whole house is setup on Google Voice ATM. No different then any other ITSP. |
| 01:22.23 | p3nguin | than |
| 01:22.27 | p3nguin | No different THAN any other ITSP. |
| 01:22.42 | pabelanger | my bad |
| 01:23.24 | tyman | p3nguin: you want to override the general nat=yes for the ext-sip-account stanza with nat=no? |
| 01:24.17 | p3nguin | tyman: The general nat setting is for your system. The peer's nat setting is for the peer. If you are behing NAT, put nat=yes in [genera]. If the peer is not behind NAT, put nat=no in the peer's definition. |
| 01:25.00 | tyman | hmm...ok. let me check try this out now |
| 01:32.00 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 01:32.40 | tyman | p3nguin: made the changes you listed. Same results. Busy signal when dialing in from pstn to sip trunk number |
| 01:32.57 | tyman | extension to extension and inside to pstn works fine |
| 01:33.15 | p3nguin | tyman: I have to assume you don't have things configured appropriately. Show me a sip debug while making an inbound call that fails. |
| 01:33.17 | tyman | sounds like nat to me |
| 01:33.25 | p3nguin | not to me |
| 01:33.30 | tyman | ok |
| 01:33.48 | tyman | what do i type at the console |
| 01:33.50 | p3nguin | NAT usually causes one-way audio problems. SIP isn't affected by NAT -- RTP/SDP is. |
| 01:34.00 | p3nguin | sip set debug on |
| 01:34.12 | tyman | k |
| 01:34.15 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 01:36.10 | tyman | p3nguin: ok, it's big |
| 01:36.17 | p3nguin | I expect that. |
| 01:36.55 | tyman | there is private info in there. i'll have to pull that stuff out for a min before posting public |
| 01:38.11 | p3nguin | sigh |
| 01:38.19 | tyman | 2 min |
| 01:38.46 | p3nguin | Do you have any idea how irritating it is to try to debug a problem when people mutilate the debug info? |
| 01:38.59 | p3nguin | I assume you don't, since you're doing it yourself. |
| 01:39.30 | tyman | should i post my sip user/pwd on a public gist? |
| 01:40.02 | drmessano | Mask the password, but dont make the username |
| 01:40.14 | drmessano | Otherwise nobody can find simple errors you may have missed |
| 01:40.18 | drmessano | and yes, people do make errors |
| 01:40.27 | tyman | k |
| 01:42.52 | tyman | p3nguin: https://gist.github.com/667281 |
| 01:44.41 | p3nguin | Looking for 559403**** in default <-- bad |
| 01:47.25 | p3nguin | Calls coming in from that peer should be going to the from-voip-provider context, which contains the extension (your phone number). |
| 01:47.51 | tyman | [from-voip-provider] |
| 01:47.51 | tyman | exten => 1559403****,1,Dial(SIP/2000) |
| 01:47.59 | tyman | currently in extensions.conf |
| 01:48.56 | p3nguin | They aren't sending it to 1559....... anyway. |
| 01:49.16 | p3nguin | They are sending it to 559....... |
| 01:49.30 | p3nguin | But the problem is that the call is going to the wrong context, for whatever reason. |
| 01:49.51 | p3nguin | So we have to address that issue before the call will fail on the bad extension in the context where it should have gone. |
| 01:50.28 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 01:51.19 | tyman | ok....so, is assume my extensions.conf paste above with context [from-voip-provider] for the external to extension dial map is incorrect? |
| 01:52.28 | p3nguin | They are sending the call to your 10-digit phone number, so your extension for calls must be that of your 10-digit phone number. |
| 01:52.54 | p3nguin | But the problem is larger than that right now. |
| 01:53.14 | p3nguin | Change type back to friend and put back in the username. |
| 01:53.31 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 01:56.12 | *** join/#asterisk JuStIcIa_ (~justicia@190.52.236.133) |
| 01:57.05 | p3nguin | I don't quite understand why the debug says it matched the peer but didn't send the call to the right context. |
| 01:57.55 | p3nguin | I also don't understand why VoIP.ms does not let me change the type of subaccounts between IAX2 and SIP after creation. |
| 01:58.54 | tyman | readded username and changed type=friend, restarted *, same |
| 01:59.19 | tyman | also removed the 1 from 1559 |
| 01:59.30 | p3nguin | Did you look at the debug to see if it is still showing the same information about the peer match? |
| 01:59.31 | tyman | which seems bizarre |
| 02:00.13 | tyman | would you like me to repost (quick like)? |
| 02:00.17 | p3nguin | I'm guessing it has something to do with the IP address not matching the hostname you've used in the peer. |
| 02:00.30 | p3nguin | checks that |
| 02:01.24 | p3nguin | trunk1.freepbx.com has address 216.82.225.24 |
| 02:01.26 | p3nguin | Not the problem. :/ |
| 02:04.09 | tyman | should i have a +1559 in the sip number? |
| 02:04.58 | p3nguin | no |
| 02:05.24 | p3nguin | It won't matter what you have for the extension until after the peer matches and the call is sent into the correct context. |
| 02:06.23 | p3nguin | The only debug I saw was sending the call into the default context instead of the one you had configured in the paste of sip.conf I saw. |
| 02:06.25 | tyman | i get that...just don't know if that's best practice...certainly would think that at least having the country code is appropriate |
| 02:06.54 | tyman | i'll repaste a new one now |
| 02:07.23 | p3nguin | It really doesn't make any difference what they extension they send the call to, the call is not reaching that context. |
| 02:07.52 | p3nguin | You can often "tune" the extension they send calls to by your register statement. |
| 02:08.24 | p3nguin | But it still won't matter until the call reaches the proper context. |
| 02:10.20 | tyman | https://gist.github.com/4b9db410ff9cf14f1029 |
| 02:11.36 | *** join/#asterisk came0 (~nick@li181-40.members.linode.com) |
| 02:11.44 | p3nguin | Still not sending to the right context, despite the claimed peer match. |
| 02:11.57 | p3nguin | I don't know what causes that. Maybe someone else does and will help. |
| 02:12.04 | came0 | hey where do I setup the extention/secret in a polycom 335?? i cant find it in the menus |
| 02:13.56 | tyman | p3nguin: can you give me the line numbers of the last gist that most concern you? |
| 02:14.08 | tyman | a few of lines anyway... |
| 02:14.36 | p3nguin | tyman: I'm looking at 49 and 59. |
| 02:15.09 | p3nguin | tyman: It indicates a match for the peer, but then sends the call into context default rather than the one where the extension exists. |
| 02:15.32 | tyman | i see that |
| 02:17.59 | tyman | p3nguin:<--- Reliably Transmitting (no NAT) to 216.82.225.24:5060 ---> |
| 02:18.00 | tyman | SIP/2.0 503 Service Unavailable |
| 02:19.06 | tyman | there is also a sip/2.0 489 Bad event |
| 02:20.16 | p3nguin | That's way later... after the call was sent into a context without a matching extension. |
| 02:20.36 | p3nguin | My primary concern at this time is to get the call routed into the correct context. |
| 02:20.42 | tyman | k... |
| 02:21.47 | tyman | p3nguin: it shouldn't matter, but i put the [from-voip-provider] context above the default context just to test...same thing (as expected) |
| 02:22.53 | p3nguin | It can't make any difference because pbx_config arranges them the way it was coded to arrange them. |
| 02:23.37 | p3nguin | I can't believe no one else is around and willing to help you with this peer match issue. |
| 02:25.19 | ManxPower | p3nguin, what is the issue? |
| 02:25.56 | p3nguin | His sip debug indicates that the peer matches, but it sends the call to the default context instead of the context configured for the peer. |
| 02:26.50 | p3nguin | It usually sends to the default context when it doesn't match, but the debug says Found peer ... |
| 02:27.08 | ManxPower | is allowguest=no ? |
| 02:27.13 | tyman | ManxPower: any help would be greatly appreciated... |
| 02:27.48 | tyman | allowguest is not set |
| 02:27.56 | ManxPower | tyman, then the call is coming in as a guest. |
| 02:28.05 | ManxPower | set allowguest=no and I bet the call will be rejected |
| 02:28.38 | tyman | set this in which context ManxPower? |
| 02:28.40 | ManxPower | if that is the case, then the incoming call never *really* matched the peer you think it is matching. |
| 02:28.45 | ManxPower | tyman, in [general] |
| 02:29.17 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 02:30.28 | tyman | still get busy out...shall i repost a sip debug? |
| 02:31.05 | ManxPower | is the call coming into the wrong context still? |
| 02:31.09 | p3nguin | I can't understand why it says it matches for the peer if it really doesn't match. |
| 02:31.20 | ManxPower | p3nguin, because allowguest is not no |
| 02:31.31 | ManxPower | is the peer that it is matching the last peer defined? |
| 02:31.34 | p3nguin | But it explicitly says it matches the peer by name. |
| 02:32.03 | ManxPower | p3nguin, in my experience that message is a lie |
| 02:32.22 | tyman | Looking for 5594030000 in default (domain 192.168.1.65) |
| 02:32.25 | tyman | y |
| 02:32.30 | p3nguin | That sounds like a bug that needs addressed pretty quickly. |
| 02:32.35 | ManxPower | tyman, did you do a reload after you put in allowguest=no in sip.conf [general] |
| 02:32.44 | tyman | y |
| 02:32.54 | tyman | restarted * completely |
| 02:34.04 | p3nguin | Does it still say it matches the peer? |
| 02:34.12 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 02:34.18 | ManxPower | tyman, issue a "sip show settings" command in the CLI and pastebin the results |
| 02:34.28 | ManxPower | no~pastebin |
| 02:35.36 | tyman | https://gist.github.com/88c34a82a80b0bcd7af2 |
| 02:35.44 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 02:35.52 | ManxPower | if the call is not coming into the context you think it is, then it is ALWAYS not matching the incoming connection |
| 02:36.22 | ManxPower | tyman, repaste an incoming call without sip debug enabled |
| 02:36.32 | tyman | ok |
| 02:36.52 | tyman | general console debug level pref? |
| 02:37.16 | tyman | level 0 ok? |
| 02:37.56 | ManxPower | verbose 3, regular debug 0 |
| 02:37.57 | tyman | nothing on console without debug |
| 02:38.12 | ManxPower | then you have something else screwed up |
| 02:38.47 | ManxPower | tyman, are you running any kind of Asterisk GUI? |
| 02:38.54 | tyman | no |
| 02:39.12 | tyman | straight ubuntu with distro packages |
| 02:39.16 | tyman | ubuntu 10.10 |
| 02:39.19 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:39.39 | tyman | verbose 3 picked up stuff |
| 02:39.55 | p3nguin | I would hope so. |
| 02:40.04 | tyman | https://gist.github.com/2c24eacd33fc18c54b7a |
| 02:40.23 | tyman | wtf? |
| 02:41.12 | p3nguin | Is this an inbound call or an outbound call? |
| 02:41.15 | tyman | that's my provider (sipstation) |
| 02:41.19 | tyman | inbound |
| 02:41.36 | ManxPower | you should generally not expect to be able to call your own number via your service provider |
| 02:41.38 | tyman | from my cell to my global dn |
| 02:41.58 | tyman | from my cell? |
| 02:42.16 | p3nguin | The reason you get that info on the verbose is because of ... |
| 02:42.21 | ManxPower | Dial("SIP/ext-sip-account-00000008", "SIP/5594030000@ext-sip-account") in new stack |
| 02:42.22 | pabelanger | tyman: exten => _X.,1,Dial(SIP/ext-sip-account/${EXTEN}) |
| 02:42.24 | p3nguin | exten => _X.,1,Dial(SIP/${EXTEN}@ext-sip-account) |
| 02:42.25 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 02:42.35 | p3nguin | The call is incorrectly coming into [default] and matching that extension. |
| 02:42.39 | ManxPower | how about NO DIALS |
| 02:42.41 | p3nguin | Then it is sent right back out. |
| 02:42.55 | p3nguin | The problem is STILL that the call is not going to the appropriate context. |
| 02:43.20 | ManxPower | tyman, pastebin your sip.conf |
| 02:43.35 | tyman | ok |
| 02:43.39 | p3nguin | Other configuration problems, like putting useful extensions in [default] will still need to be addressed, though. |
| 02:43.42 | ManxPower | chnge ony passwords |
| 02:43.52 | tyman | k |
| 02:44.14 | ManxPower | p3nguin, using the [default] context is never needed and should be avoided for exactly these reasons |
| 02:44.28 | p3nguin | Like I said, it still has to be addressed. |
| 02:44.34 | p3nguin | But it's not causing this problem. |
| 02:45.17 | ManxPower | p3nguin, it is causing cli noise |
| 02:45.23 | p3nguin | agreed |
| 02:45.34 | phix | hey, how do I test if g729 codec is working correctly? |
| 02:45.38 | phix | I has licences |
| 02:45.58 | p3nguin | disallow=all allow=g729 make a call |
| 02:46.38 | phix | ok it is using alaw :/ |
| 02:46.49 | p3nguin | Not if you did what I said. |
| 02:47.00 | p3nguin | disallow=all makes sure it doesn't use alaw. |
| 02:47.04 | pabelanger | phix: g729 show licenses |
| 02:47.13 | pabelanger | phix: *CLI> g729 show licenses |
| 02:47.16 | pabelanger | when call is in progress |
| 02:48.46 | tyman | sip.conf: https://gist.github.com/8f0a86a079be3892803b |
| 02:49.11 | *** join/#asterisk Baylink (~jra@65.34.94.26) |
| 02:49.17 | phix | ok, my SIP provider doesn't support g729 :/ that could be why |
| 02:49.25 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-lbclezkstdzntdxe) |
| 02:49.35 | pabelanger | tyman: Ya, your sip.conf is messed up bad |
| 02:49.43 | phix | they support gsm but the quality sounds choppy |
| 02:49.51 | phix | what else would cause that desides bandwidth? |
| 02:49.55 | phix | echo filtering? |
| 02:50.13 | pabelanger | tyman: peer definitions and registration go after the [authentication] context |
| 02:50.14 | tyman | pabelanger: i hate these package sample configs |
| 02:50.26 | phix | s/desides/besides/ |
| 02:50.50 | drmessano | So don't use the sample configs |
| 02:51.05 | drmessano | That is why they are *samples* |
| 02:51.08 | pabelanger | tyman: move everything from line 99 to 125 to the bottom of sip.conf |
| 02:51.10 | phix | :) |
| 02:51.53 | tyman | drmessano: i'm not going to...there is a lot of cruft to go thru until i figure out what's going on... |
| 02:52.00 | pabelanger | tyman: And comment out unused settings. IE: you have 2 settings for context, line 96 and 129 |
| 02:52.02 | tyman | pabelanger: k |
| 02:52.15 | phix | No such command 'g729 show licenses' (type 'help g729 show licenses' for other possible commands) |
| 02:52.37 | ManxPower | tyman, paste that again, this time without the comments |
| 02:52.44 | pabelanger | phix: *CLI> module load codec_g729.so |
| 02:53.03 | tyman | k |
| 02:53.06 | phix | g729 debug is valid though |
| 02:53.18 | ManxPower | and use that the one without the comments to test with |
| 02:53.27 | phix | [Nov 8 13:53:38] WARNING[11293]: loader.c:655 load_resource: Module 'codec_g729.so' already exists. |
| 02:53.41 | ManxPower | phix, g729<tab> |
| 02:53.56 | phix | ManxPower: debug, that is all |
| 02:54.01 | pabelanger | phix: where did you get the g729 codec? |
| 02:54.27 | p3nguin | pabelanger: The register statements are required to go above the peer definitions. Moving line 99 to the bottom will break that. |
| 02:54.27 | phix | hmmmm good question |
| 02:55.03 | p3nguin | pabelanger: Wait, you're taking the peer definitions with it. Disregard. |
| 02:55.19 | ManxPower | this is why I want all the sip.conf noise ggone |
| 02:55.22 | phix | what is that free one called again? the one that violates the licence :) I think I might have an old install of that laying about |
| 02:55.40 | ManxPower | phix, that is the only topic forbidden by Digium on this channel |
| 02:55.47 | p3nguin | Now... why are we using a sample config? |
| 02:55.56 | ManxPower | p3nguin, ask tyman |
| 02:56.13 | x86 | ManxPower: Digium actually sponsors this channel? |
| 02:56.15 | dev_ast | g729 license can be purchased from digium at $10/channel |
| 02:56.17 | x86 | I did not know that... |
| 02:56.20 | phix | ManxPower: haha ok :) I have purchased licences though! |
| 02:56.21 | ManxPower | x86, Digium people have ops. |
| 02:56.23 | p3nguin | That's not the config he showed me earlier, and it's not the config I modified and sent back. |
| 02:56.30 | ManxPower | phix, then you can contact digium dfor support |
| 02:56.46 | phix | nice, what time zone are they in? |
| 02:56.46 | x86 | ManxPower: true... but I didn't know Digium "founded" the IRC channel or whatever |
| 02:56.58 | ManxPower | p3nguin, then I am done. I don't have time to deal with magically movnig target. and I have used too much of my time already. |
| 02:57.00 | x86 | interesting |
| 02:57.08 | ManxPower | x86, doesn't really matter if they did or they did not. |
| 02:57.09 | phix | Would some one be awake there now? |
| 02:57.11 | x86 | ManxPower: true |
| 02:57.17 | ManxPower | phix, not a chance. |
| 02:57.34 | phix | :( |
| 02:57.48 | phix | time zones are annoying |
| 02:57.56 | ManxPower | tyman, use a minimal config with allowguest=no and context=INVALID in sip.conf [general] |
| 02:58.26 | x86 | allowguest=maybe ;) |
| 02:58.35 | p3nguin | allowguest=eatme |
| 02:58.42 | x86 | allowguest=ifsheshot *evilgrin* |
| 02:58.55 | tyman | https://gist.github.com/ede567bc0e248604b9ef |
| 02:59.13 | tyman | ManxPower: this is the sample without comments |
| 02:59.24 | p3nguin | still broken |
| 02:59.34 | p3nguin | but at least it's clean this time. |
| 02:59.45 | WIMPy | x86: Give me your number and I'll get you some of those calls :-) |
| 03:00.16 | p3nguin | Time for The Walking Dead! Be back in an hour! |
| 03:00.16 | ManxPower | tyman, 1) remove the passwords 2) put it on a pastebin that allows me to edit it. |
| 03:00.26 | p3nguin | pastebin.com |
| 03:00.29 | phix | ok what is that debian command to search which package provides a certain file? |
| 03:00.30 | p3nguin | ~pb |
| 03:00.30 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 03:00.51 | p3nguin | phix: dpkg -L I think |
| 03:01.02 | phix | p3nguin: ok |
| 03:01.24 | phix | p3nguin: hmmm I need to know the name of the package, I guess I could try '*' |
| 03:01.30 | *** join/#asterisk sbszulu (~dundubala@41.14.139.5) |
| 03:01.47 | phix | :D -S |
| 03:02.02 | tyman | http://pastebin.com/n1YbvV6p |
| 03:02.26 | pabelanger | tyman: http://pastebin.com/gEg39ixX <- what a sip.conf _should_ look like. Not the difference |
| 03:02.28 | x86 | WIMPy: 312-725-4143 ;-) |
| 03:02.52 | pabelanger | s/Not/Note/ |
| 03:03.04 | x86 | can anyone try to URI dial me? |
| 03:04.27 | tyman | pabelanger: yep |
| 03:04.53 | ManxPower | tyman, try this; v |
| 03:05.20 | ManxPower | http://pastebin.com/6RsYvnq6 |
| 03:05.52 | x86 | 7800@x86lab.no-ip.org |
| 03:05.59 | x86 | can someone try calling me there? |
| 03:07.56 | tyman | ManxPower: trying now |
| 03:08.47 | tyman | Booyah! |
| 03:08.55 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 03:09.32 | tyman | ManxPower: wtf? |
| 03:09.54 | tyman | like an idiot, i pasted it in without the pwds and it's working |
| 03:10.24 | p3nguin | Incoming calls don't use passwords. |
| 03:10.44 | tyman | ah... |
| 03:11.01 | p3nguin | It matches the peer and there is no password, so the call proceeds. |
| 03:11.57 | tyman | p3nguin: i see |
| 03:12.08 | p3nguin | I think I worded the first sentence incorrectly. |
| 03:12.25 | tyman | ManxPower: what specifically was the problem |
| 03:12.43 | tyman | context=INVALID? |
| 03:13.09 | p3nguin | If your peer sends a call in to you and you don't challenge it for authentication, no password is needed because you matched the peer already. |
| 03:13.15 | phix | ok I am going to try install g729 again :/ |
| 03:13.24 | p3nguin | That's what I meant, but I said it wrong the first time. |
| 03:14.23 | p3nguin | phix: Delete your stolen/pirated codec first. |
| 03:14.34 | tyman | p3nguin: now, how do i get my internal extensions and outbound calls to work again |
| 03:14.50 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 03:14.51 | tyman | obviously thru modification of extensions.conf |
| 03:14.57 | p3nguin | Configure the extensions in extensions.conf |
| 03:15.18 | p3nguin | Put the relevant extensions in an appropriate context. |
| 03:15.31 | p3nguin | Also configure each peer with said context. |
| 03:16.59 | p3nguin | I configure all phone peers with context=phones |
| 03:17.28 | ManxPower | tyman, context=invalid make sure if you don't have a context defined for a peer the call fails |
| 03:17.32 | p3nguin | Then in [phones] I include contexts for outbound and internal. |
| 03:18.17 | p3nguin | Then in [outbound] I put my Dial() for my ITSP and in [internal] I put the Dial()s for my phones. |
| 03:18.35 | p3nguin | Your inbound context should never include your outbound context. |
| 03:20.05 | tyman | damn... |
| 03:20.09 | p3nguin | [default] should be empty, [invalid] should probably only include an extension for s at the most. |
| 03:21.03 | tyman | hmm...didn't think i changed anything but the secrets back in and it's broken in and out now... |
| 03:21.45 | p3nguin | If you added the secret, you may be challenging the peer to match your secret. |
| 03:22.12 | tyman | the peer being my provider? |
| 03:22.16 | p3nguin | yes |
| 03:22.36 | tyman | they gave it to me, why would they not be matching it i wonder? |
| 03:22.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 03:25.41 | ManxPower | i have never ever seen a provider provided asterisk configuration actually work. |
| 03:25.54 | p3nguin | Amazing, isn't it? |
| 03:25.58 | tyman | ManxPower: that's reassuring |
| 03:26.31 | p3nguin | It's like we expect an ITSP to know how to configure a phone system for peering or something. |
| 03:26.34 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 03:26.54 | tyman | damn... |
| 03:27.16 | tyman | shit doesn't work again...took out pwds again...this is embarassing |
| 03:27.32 | p3nguin | It's how we learn. |
| 03:27.47 | ManxPower | tyman, you are further than you were before. at least now you have a config that appears to work. |
| 03:27.57 | p3nguin | Did you read The Book? |
| 03:27.58 | tyman | y |
| 03:28.00 | p3nguin | ~book |
| 03:28.00 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 03:28.32 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 03:28.49 | tyman | p3nguin: I will indeed read this book asap. This is the best * book of all? |
| 03:29.14 | p3nguin | The next edition is in the works. I don't think it has been printed yet, though. |
| 03:29.27 | x86 | tyman: it is THE BOOK |
| 03:29.33 | x86 | tyman: it's the bible of Asterisk |
| 03:29.38 | tyman | i wonder if it's in rough cuts (beta) on safari |
| 03:29.41 | tyman | k |
| 03:30.32 | *** join/#asterisk OrNix (~ornix@l49-131-248.cn.ru) |
| 03:30.35 | ManxPower | the asterisk is the best book out there for asterisk. the asterisk book is also written for the asterisk version that is two asterisk versions behind current. |
| 03:30.53 | tyman | that was what concerned me |
| 03:31.13 | tyman | i have this book in print...never got to asterisk as planned (too busy on other stuff) |
| 03:31.26 | tyman | now i'm getting into this as of saturday |
| 03:32.35 | tyman | where is the best place to supplement the data gap between The Book and current version info |
| 03:33.24 | p3nguin | The sample configs and the documentation provided with each release of Asterisk should be pretty helpful. |
| 03:35.22 | tyman | ok...i was planning on running 1.8 from source on latest ubuntu |
| 03:35.46 | tyman | any caveats? |
| 03:35.50 | p3nguin | Documentation is included with the source. |
| 03:36.47 | ManxPower | documentation is not generally included in the packaged versions if asterisk |
| 03:39.05 | p3nguin | The docs are also online: http://svnview.digium.com/svn/asterisk/branches/1.8/doc/ |
| 03:39.07 | tyman | ok...i'll get cracking on this book asap. what just started as setting up a lab to start reading/experimenting became hours of time in tshooting for the inbound |
| 03:39.27 | tyman | i REALLY appreciate all this groups expertise and patience |
| 03:39.37 | tyman | thank you all very much |
| 03:44.21 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 03:46.07 | p3nguin | x86: Did you ever get your SIP call? |
| 03:52.43 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 03:55.04 | x86 | p3nguin: someone just tried it, it rang, but I was downstairs and didn't get to the phone in time |
| 03:55.14 | p3nguin | I did that. |
| 03:55.37 | x86 | you don't set caller ID on outbound calls to something useful? :P |
| 03:56.04 | x86 | try the call again, I want to verify two-way audio |
| 03:56.40 | p3nguin | I was just originating a call from the CLI. |
| 03:56.45 | x86 | ah ok |
| 03:56.52 | x86 | well cool, it rings in anyway |
| 03:56.57 | x86 | thanks :) |
| 03:57.00 | p3nguin | It probably said it was from asterisk. |
| 03:57.04 | x86 | yep |
| 03:57.22 | x86 | thanks again, off to bed for me |
| 03:57.24 | x86 | night * |
| 03:57.29 | x86 | (pun intended) |
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| 04:27.02 | dev_ast | CLI > core stop now (coz bed time) |
| 04:31.20 | phix | it is bed time?? |
| 04:32.37 | dev_ast | CST 10:30pm.. oh ya, it's bed time |
| 04:33.03 | phix | heh light weight :) |
| 04:33.17 | dev_ast | what's ur time zone |
| 04:33.21 | phix | you only need 4hrs of sleep :) |
| 04:33.37 | dev_ast | i need at least 8+ hrs |
| 04:33.37 | phix | +11 UTC atm due to daylight savings |
| 04:35.39 | phix | well that was a healthy lunch, half a can of pringles and a vitamin suppliment pill :P |
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| 06:38.35 | Supari | have a problem with a sip registration, knocks off. Says Request Sent... However i have 2 trunks 1 registeres perfectly other shows this until i reboot the entire system |
| 06:38.43 | Supari | running 1.4.36 |
| 06:39.57 | Supari | any ideas ? |
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| 07:18.11 | ManxPower | well isn't that special. freepbx does not support permit/deny |
| 07:20.36 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 07:35.48 | ChannelZ | toss that piece of shit in the crapper |
| 07:38.48 | ManxPower | ChannelZ, i would never use it if i had a choice. we are slowly writing replacements for more and more parts of it. it will be gone from our network as soon as it is feasible to do. |
| 07:40.41 | ManxPower | have about 80 percent of the user portal replaced, all of the provisioning system |
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| 07:57.12 | MCIML | hey guys i have a problem. I'm setting up an asterisk server with a polycom 501 and a aastra 6757i ct. the polycom works fine and can call the aastra, but the aastra cannot make any calls |
| 07:57.36 | MCIML | not sure where the problem lies |
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| 08:02.06 | MCIML | anyone around? |
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| 08:07.39 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 08:07.40 | schmidts | good morning |
| 08:08.00 | MCIML | hey |
| 08:08.20 | MCIML | i have a problem, can you help me? :D |
| 08:09.19 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 08:09.56 | schmidts | maybe ;) atleast i can try it |
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| 08:14.18 | MCIML | im setting up a server from source using the latest starfish guide for 1.8 |
| 08:15.13 | MCIML | i set up 2 phones, polycom 501 and aastra 6757i...the polycom can make all the calls in the dialplan and can dial the aastra phone, but the aastra phone cannot make any sort of outgoing calls |
| 08:17.59 | schmidts | <PROTECTED> |
| 08:19.01 | MCIML | asterisk log or phone log? |
| 08:19.26 | MCIML | aastra phone shows no service, but the polycom can dial the aastra phones extension and it rings... |
| 08:19.46 | schmidts | asterisk log ;) |
| 08:21.17 | MCIML | cli isnt showing any output from the aastra phone |
| 08:21.45 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 08:25.57 | schmidts | have you tried a sip debug? |
| 08:26.07 | schmidts | sip set debug peer "peerfromaastraphone" |
| 08:26.18 | schmidts | or you can also do a sip set debup ip "ipfromaastra" |
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| 08:33.51 | MCIML | nothing is coming up |
| 08:34.05 | MCIML | phone is showing no service |
| 08:34.05 | MCIML | yet it can be called... |
| 08:36.54 | schmidts | do you have configured a fix ip for this peer? if yes its no wonder that it can be called |
| 08:38.09 | MCIML | nope its dynamic |
| 08:43.03 | *** join/#asterisk X-Raimo (~asmpub@opensuse/member/raimoschmidt) |
| 08:44.12 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:44.17 | schmidts | strange ;) |
| 08:44.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:46.25 | X-Raimo | hello. We using Oracle+ODBC. DB is encoded in win cp1251. We have cyrillic CallerID field. And in func_obdc.conf we have: readsql=select convert(t.callerid, 'utf8') from sip_users t where t.name='${ARG1}' Asterisk receives Called ID in wrong way. |
| 08:47.14 | X-Raimo | everything is ok when we use non-cyrillic chars. |
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| 08:51.54 | Godfather_ | i'm having echo problems with a tdm410p, echo ratio with fxotune 0.179.. how i solve it? |
| 08:52.00 | Godfather_ | http://pastebin.com/wjFc6Nbu |
| 08:52.30 | *** join/#asterisk Burnz1984 (~Burnz1984@84.19.208.6) |
| 08:52.35 | Burnz1984 | good morning |
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| 08:54.34 | Burnz1984 | when i use musiconhold.conf to play mp3 files in asterisk 1.2 must i discripe which mp3-player in the musiconhold.conf shall play the files? |
| 08:54.37 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:55.49 | MCIML | schmidts any other suggestions? |
| 08:55.54 | Burnz1984 | like: [mp3] mode=files directory=xx/xx application=/usr/bin/xx? |
| 08:56.55 | schmidts | mciml you really see nothing incoming with sip debug turned on? if no check your proxy configuration in the aastra |
| 09:01.36 | tzafrir | Burnz1984, 'application' has no effect in 'mode=files' |
| 09:01.39 | *** join/#asterisk shimizu (~shimizu@li145-88.members.linode.com) |
| 09:02.09 | tzafrir | Godfather_, what EC do you use? |
| 09:02.28 | Godfather_ | tzafrir, oslec |
| 09:02.52 | tzafrir | Hmm... and it is actually loaded? |
| 09:03.03 | Godfather_ | echo ratio = 0.2996 (1365.1 / 4557.0) :-( |
| 09:03.04 | tzafrir | Do you see it set in the output of lsdahdi ? |
| 09:03.27 | Godfather_ | http://pastebin.com/HLEveD1A |
| 09:03.48 | Godfather_ | I assume yes |
| 09:03.54 | tzafrir | yes |
| 09:04.06 | tzafrir | And do you actually hear echo? |
| 09:05.00 | tzafrir | Burnz1984, if you want to play mp3 files with mode=files, you need to make Asterisk capable of playing mp3 files |
| 09:05.13 | tzafrir | use codec_mp3 from asterisk_addons |
| 09:05.13 | *** part/#asterisk X-Raimo (~asmpub@opensuse/member/raimoschmidt) |
| 09:05.21 | Godfather_ | tzafrir, yes... i modified rxgain and txgain rxgain=7.0 txgain=3.0 |
| 09:05.25 | Godfather_ | this could be a problem? |
| 09:05.44 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 09:05.59 | Godfather_ | tzafrir, i'm not pretty sure how to use fxotune to get better results |
| 09:06.15 | tzafrir | fxotune reduces the generation of echo |
| 09:06.25 | z4nD4R | <PROTECTED> |
| 09:06.30 | *** join/#asterisk coppice (~chatzilla@m121-203-236-211.smartone-vodafone.com) |
| 09:06.30 | tzafrir | the echo canceller tries to cancel the remaining echo |
| 09:06.38 | tzafrir | OSLEC is pretty good at that |
| 09:06.45 | Burnz1984 | how to amke it capable? |
| 09:07.08 | tzafrir | codec_mp3 from asterisk-addons . It's a separate package |
| 09:07.19 | Burnz1984 | ny extension.conf works in asterisk 1.4 an i have write it in 1.2 an there it don´t play mp3 |
| 09:07.28 | Burnz1984 | my* |
| 09:07.59 | tzafrir | http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addons-1.2.9.tar.gz |
| 09:08.09 | tzafrir | (for 1.2) |
| 09:08.46 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 09:10.11 | Godfather_ | tzafrir, http://pastebin.com/5wid1XsQ |
| 09:10.24 | Burnz1984 | our system plays mp3 but it don´t play the mp3 i want ^^ |
| 09:10.47 | Burnz1984 | i have make a directory like /var/lib/asterisk/busy |
| 09:12.01 | Burnz1984 | i this directory is my mp3 an i have make a link in the default class from moh.conf to play the musik with dial(sip/xx,10,m) the "m"-parameter shal play the musik in the default class or not? |
| 09:12.51 | *** part/#asterisk pabs3 (~pabs@203-59-93-176.perm.iinet.net.au) |
| 09:13.11 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 09:13.37 | tzafrir | Burnz1984, if you want the application= to have effect, don't use mode=files |
| 09:13.58 | tzafrir | But then again, consider converting them to wav files |
| 09:14.14 | Burnz1984 | i have changed it an it plays always the false mp3 |
| 09:14.14 | *** join/#asterisk coppice (~chatzilla@m121-203-236-211.smartone-vodafone.com) |
| 09:14.56 | Burnz1984 | the problem is, that it plays some mp3 but not the mp3 that is in my directory...with asterisk 1.4 it works |
| 09:16.21 | Burnz1984 | of course i have to restart the asterisk? i work with the "reload" command |
| 09:16.57 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 09:17.37 | Godfather_ | tzafrir, do you see anythinig rare on my chan_dahdi.conf? i dont know what exactly |
| 09:18.09 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 09:19.16 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 09:21.20 | z4nD4R | One Question: if i create certificate... what should by in Common Name? - if i put here my IP is this correct ? |
| 09:21.27 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 09:24.39 | *** join/#asterisk ketas-av (~ketas@fffu.vps.bitfolk.com) |
| 09:27.39 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 09:30.36 | z4nD4R | Hi, on my asterisk server 1.8.0 have 2 user succesfully registered with TLS... But if i want establish call i become this message http://pastebin.com/kwd1xHaM . On client show that "Is Ringing", but on second side is done nothing.... Any ideas? |
| 09:33.01 | Godfather_ | tzafrir, and this is a graphic made with fxotune... http://img17.imageshack.us/f/pantallazoix.png/ |
| 09:33.36 | AdvoWork | hi there, ive got a remote server at another site, that connects to the site here.. ive just logged into the asterisk server, done asterisk -rvvv and then done sip show peers. the phone thats NR shows as UNKNOWN. all others are ok, any suggestions please? Ive tried registering but no luck |
| 09:34.01 | tzafrir | Godfather_, frankly, I'm less interested by that |
| 09:34.28 | tzafrir | in asterisk, look at the output of: dahdi show channel 1 |
| 09:34.58 | tzafrir | How many "taps" does it have for the echo canceller? |
| 09:36.15 | AdvoWork | is that to me? |
| 09:42.39 | z4nD4R | One Question: if i create certificate... what should by in Common Name? - if i put here my IP is this correct ? |
| 09:44.31 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 09:46.40 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:52.05 | z4nD4R | Hi, on my asterisk server 1.8.0 have 2 user succesfully registered with TLS... But if i want establish call i become this message http://pastebin.com/kwd1xHaM . On client show that "Is Ringing", but on second side is done nothing.... Any ideas? |
| 10:05.36 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 10:08.50 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:15.02 | *** join/#asterisk OrNix (~ornix@l49-131-248.cn.ru) |
| 10:15.16 | Godfather_ | tzafrir, lol! |
| 10:15.25 | Godfather_ | 256 taps |
| 10:15.35 | Godfather_ | but it says "echo cancellaation off" |
| 10:15.44 | shamelessn00b | which card |
| 10:15.52 | Godfather_ | Echo Cancellation: |
| 10:15.52 | Godfather_ | 256 taps |
| 10:15.52 | Godfather_ | currently OFF |
| 10:15.53 | *** join/#asterisk SeTTleR (~bernd@p5DDEFD8F.dip.t-dialin.net) |
| 10:16.00 | tzafrir | Godfather_, currently off. |
| 10:16.19 | SeTTleR | hi |
| 10:16.19 | shamelessn00b | hello |
| 10:16.19 | Godfather_ | tzafrir, how can i enable it? |
| 10:16.19 | tzafrir | It should be on when there's actually a call |
| 10:16.22 | Godfather_ | ah |
| 10:16.24 | shamelessn00b | check during the call |
| 10:16.39 | Godfather_ | shamelessn00b, tdm410p |
| 10:16.41 | Godfather_ | ok |
| 10:17.02 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 10:18.15 | z4nD4R | Hi, on my asterisk server 1.8.0 have 2 user succesfully registered with TLS... But if i want establish call i become this message http://pastebin.com/kwd1xHaM . On client show that "Is Ringing", but on second side is done nothing.... Any ideas? |
| 10:18.37 | AdvoWork | can you remotely reregister a phone from the asterisk terminal? |
| 10:19.47 | z4nD4R | AdvoWork: for me? |
| 10:20.35 | AdvoWork | no.. |
| 10:20.43 | z4nD4R | ok |
| 10:21.08 | Godfather_ | during the call is enabled |
| 10:22.24 | Godfather_ | tzafrir, i noticied that i have echo with the snom m9 terminal, now i tried a spa504 enabled with no echo!, then the problem will be on the m9... |
| 10:23.13 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 10:23.40 | Godfather_ | i googled for it, and seems some people reporting bad echo with the snom |
| 10:33.45 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 10:37.49 | Burnz1984 | when i make some forwarding, i want to place a "F" to the calleridnum (Fxxxxxxx) that the person who get the forwarding see the calling is a forwarding... |
| 10:37.52 | Burnz1984 | i know that i can set the "F" in asterisk 1.4 with exten => s,x,set(CALLERID(num)=Z${CALLERID(num)}) an it works...but in asterisk 1.2 it doesn´t work |
| 10:39.40 | *** join/#asterisk sbszulu (~dundubala@41.14.164.151) |
| 10:40.54 | ectospasm | Burnz1984: Asterisk 1.2 is fast approaching its EOL date, don't use it if you can avoid it. |
| 10:41.35 | AdvoWork | can you register an extension via the console? |
| 10:42.04 | ectospasm | AdvoWork: I wouldn't think so, that device would need to contact Asterisk directly |
| 10:42.06 | tzafrir | AdvoWork, basically, yes. But it won't remain there after the next reload |
| 10:42.15 | Burnz1984 | is it possible that it works only with internal phones and not when a call comes from outside? |
| 10:42.38 | ectospasm | Burnz1984: that depends, does it route through the same dialplan? |
| 10:42.41 | z4nD4R | Someone to help with TLS? |
| 10:45.16 | *** join/#asterisk guilhermebr (~Guilherme@189.50.119.253) |
| 10:45.16 | AdvoWork | tzafrir, could you explain please? |
| 10:45.25 | AdvoWork | ive got 1 phone thats NR, all others are fine :S |
| 10:45.44 | tzafrir | AdvoWork, hmm... we were referring to different things as "extensions". Ignore what I wrote |
| 10:46.46 | AdvoWork | oh ok |
| 10:47.13 | Burnz1984 | i have found it...i must write ...CALLERID(number) not CALLERID(num) -.^ |
| 10:47.13 | AdvoWork | if i do sip show peers, i get: EXTENSION.. (Unspecified) D N 0 UNKNOWN |
| 10:49.49 | *** join/#asterisk Dovid[Laptop] (Dovid@213.8.121.90) |
| 10:50.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:52.48 | bn-7bc | 0,0,,0,0,00,0,00000000000000000000000000000000 |
| 10:53.15 | bn-7bc | upps,mi kebord fell down,sorry |
| 10:55.17 | z4nD4R | Hi. I'm trying to configure TLS support for SIP in asterisk ... my asterisk server 1.8.0 have 2 user succesfully registered with TLS... But if i want establish call i become this message http://pastebin.com/kwd1xHaM . On client show that "Is Ringing", but on second side is done nothing.... Any ideas? |
| 11:00.56 | AdvoWork | ectospasm, any idea why a few phones would work, monitored/unmonitored, but this one wont? |
| 11:01.24 | ectospasm | not off the top of my head, no |
| 11:01.58 | ectospasm | check the configuration of the nonworking phone against one that does... there's got to be something different. |
| 11:10.31 | AdvoWork | ectospasm, theres not, theyre all the same :/ checked |
| 11:10.57 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:11.05 | ectospasm | AdvoWork: what about each one's Asterisk config? |
| 11:11.18 | *** part/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 11:11.53 | *** join/#asterisk coppice (~chatzilla@m121-202-4-72.smartone-vodafone.com) |
| 11:12.05 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 11:21.28 | *** join/#asterisk centoslinux (~linuxcent@rhelbox.uio.no) |
| 11:23.06 | *** join/#asterisk OrNix (~ornix@host89-251-107-48.hnet.ru) |
| 11:50.49 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 12:02.44 | z4nD4R | Hi. I'm trying to configure TLS support for SIP in asterisk ... my asterisk server 1.8.0 have 2 user succesfully registered with TLS... But if i want establish call i become this message http://pastebin.com/kwd1xHaM . On client show that "Is Ringing", but on second side is done nothing.... Any ideas? |
| 12:07.21 | *** join/#asterisk iscario (~quassel@24.244.71-86.rev.gaoland.net) |
| 12:07.27 | *** join/#asterisk Tim_Toady (~moi@77.49.109.71.dsl.dyn.forthnet.gr) |
| 12:08.48 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
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| 12:25.49 | *** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net) |
| 12:28.20 | *** join/#asterisk Micha_Micha (~Administr@mail.splendor.net) |
| 12:29.13 | Micha_Micha | Hello guys...I have a question about asterisk process...I have 2 asterism processes running on my server.../usr/sbin/asterisk -f -vvvg -c and /bin/sh /usr/sbin/safe_asterisk |
| 12:29.36 | tzanger | Micha_Micha: safe_asterisk is a script that restarts asterisk if it crashes |
| 12:29.39 | Micha_Micha | Can you please help me to know the process /usr/sbin/asterisk -f -vvvg -c is for? |
| 12:30.07 | kaldemar | man asterisk will tell some more. |
| 12:30.12 | tzanger | that is the actual PBX, and depending on your system and version of pstools you might see dozens of those processes. Asterisk is heavily multi-threaded. |
| 12:30.32 | *** join/#asterisk sbszulu (~dundubala@41.14.68.203) |
| 12:30.32 | Micha_Micha | tzanger, is that mean if /usr/sbin/asterisk -f -vvvg -c has been restarted then whole asterisk has been restarted and calls has benn dropped |
| 12:30.36 | Micha_Micha | I'm right? |
| 12:31.50 | tzanger | correct, if asterisk restarts any calls would have been dropped. There is a chance that any calls where the RTP does not go through asterisk would stay up but I'm not 100% sure what happens to a call when the control disappears |
| 12:32.28 | Micha_Micha | tzanger, thanks a lot |
| 12:33.11 | tzanger | Micha_Micha: it's easy to test; establish a call and then kill -9 asterisk as root, see what happens to the audio |
| 12:33.44 | tzanger | (that's a very nasty thing to do to a process, but it'd simulate the kernel terminating the process with extreme prejudice (NULL pointer dereference, bad memory, etc.) |
| 12:34.43 | *** join/#asterisk srini (~deepak@219.91.201.74) |
| 12:36.11 | srini | hi everybody!!! |
| 12:36.42 | srini | can anyone help me in troubleshooting vicidial |
| 12:37.31 | srini | can anyone help me?!!! |
| 12:37.48 | srini | iam not able to make outbound calls in asterisk |
| 12:40.17 | wdoekes2 | ~ask |
| 12:40.18 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:40.29 | srini | hi wdoeskes2 |
| 12:41.12 | srini | iam not able to make outbound call through my asterisk server |
| 12:41.35 | srini | iam using VICIDIAL now |
| 12:42.25 | srini | wdowskes2 r u there? |
| 12:44.56 | srini | iam using VICIDIAL & iam not able to make outbound calls, can any one help regarding this? |
| 12:45.32 | pabelanger | srini: ~vicidial |
| 12:45.36 | pabelanger | err |
| 12:45.40 | pabelanger | ~vicidial |
| 12:45.40 | infobot | [vicidial] a predictive dialer available from http://astguiclient.sourceforge.net/vicidial.html . |
| 12:46.12 | z4nD4R | ~tls |
| 12:46.12 | infobot | from memory, tls is Transport Layer Security (ssl) but there could be thread-local storage or, in polish, TrzyLiterowy Skrot, what's mean ThreeLetter Acronym |
| 12:47.54 | srini | when the agent login is done it says you are currently the only one in this conference. |
| 12:48.55 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e7.bcn.adamo.es) |
| 12:49.05 | srini | when i select dial next it says "thats not a valid extension" |
| 12:49.19 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 12:49.28 | wdoekes2 | srini: pabelanger means you should be asking in a different place, see http://astguiclient.sourceforge.net/contact.html |
| 12:59.06 | *** join/#asterisk w0rk3r (~Senny@85.15.88.100) |
| 13:06.39 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:07.13 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 13:08.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:14.23 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 13:16.23 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 13:17.11 | adnc | i've hylafax running with iaxmodem on my asterisk. sometimes it simply doesnt pick up the call. restarting hylafax doesnt help. after restart of asterisk 1.6 most of the time it works. could someone point me to the right direction please, what could i do? |
| 13:18.22 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 13:19.15 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:23.53 | fauxalliance | <PROTECTED> |
| 13:24.31 | adnc | fauxalliance, yes, but suppose it is a asterisk problem |
| 13:24.37 | adnc | maybe of my configuration |
| 13:25.43 | adnc | but if that is your response i'm sure there will be no help even if it was a asterisk problem |
| 13:26.16 | fauxalliance | adnc, limited.... deduce IF it is an asterisk problem... show us evidence of the problem and we will help... |
| 13:26.22 | *** join/#asterisk AndyRomano (~Adium@195.34.154.20) |
| 13:26.44 | fauxalliance | coming in here spouting about hylafax and iaxmodem... neither of which are supported here. |
| 13:26.46 | adnc | fauxalliance, i don't know what i could show you? maybe the sip.conf? |
| 13:26.58 | adnc | fauxalliance, tell me the right way? |
| 13:27.09 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 13:27.24 | adnc | which way would be the most elegant, tell me and i will go for it |
| 13:28.08 | fauxalliance | start barking up the hylafax tree, perhaps even IAX modem... faxing over IP is not 100% reliable. |
| 13:29.38 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 13:34.18 | [TK]D-Fender | adnc: What does **SIP** have to do with **IAX**Modem? |
| 13:34.56 | Maliuta | [TK]D-Fender: he needs to SIP into something more comfortable? ;P |
| 13:34.58 | adnc | [TK]D-Fender, maybe the incomming call? |
| 13:35.10 | Maliuta | adnc: that would be IAX[2] |
| 13:35.16 | adnc | it is comming from a sip source |
| 13:35.16 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 13:35.34 | adnc | Maliuta, iax[2]? i didn't understand |
| 13:36.58 | Maliuta | IAX modem only takes IAX (which is actually IAX2 now). Your * box can terminate the SIP call and then connect the next leg via IAX2 ... but it takes the SIP out of any problem with IAX Modem |
| 13:37.21 | adnc | Maliuta, sorry, excuse me, sure it is iax2 |
| 13:37.43 | adnc | i just used freepopfax.com and the fax did arrive |
| 13:37.57 | adnc | maybe extension.conf? |
| 13:38.18 | adnc | i'm sure if i try it in an hour it won't answer the call anymore |
| 13:40.20 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:40.21 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 13:42.09 | atan | Anyone have luck with getting a Nokia E71 cell phone to do video calls? It supports SIP which is great =\ but it has h263 + a camera... but I can't for the life of me figure it out |
| 13:42.17 | [TK]D-Fender | adnc: SHOW US THE ACTUAL FAILURE |
| 13:43.10 | adnc | [TK]D-Fender, i can not see any failure. it just simply doesn't answer. but thank you very much |
| 13:43.25 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 13:43.46 | [TK]D-Fender | adnc: Of course you can't see it. If you coudl you'd have fixed it and it'd be working. Show US |
| 13:46.33 | adnc | http://pastebin.com/UyxRuzVJ this is what asterisk shows in the cli when the call comes in |
| 13:47.17 | adnc | if i can show anything else, please tell me |
| 13:47.58 | [TK]D-Fender | adnc: IAXModem creates a TTY I believe you should eb looking at. |
| 13:48.05 | [TK]D-Fender | adnc: And loko at the Hylafax side as well |
| 13:48.22 | adnc | <PROTECTED> |
| 13:48.44 | adnc | i've a lot of these entries in the iaxmodem logfile |
| 13:48.51 | adnc | but this is not asterisk anymore |
| 13:52.52 | [TK]D-Fender | adnc: Sounds like you should be checking out their mailing-lists, etc |
| 13:53.03 | adnc | thanks, i'm going through them |
| 13:53.23 | *** join/#asterisk Tim_Toady (~moi@77.49.109.71.dsl.dyn.forthnet.gr) |
| 13:54.59 | AdvoWork | in askerisk -rvvvv why would it show UNREACHABLE? |
| 13:57.24 | *** join/#asterisk moos3 (~rgenthner@cpe-72-224-105-166.maine.res.rr.com) |
| 13:58.15 | pabelanger | AdvoWork: your peer is UNREACHABLE |
| 13:58.16 | moos3 | how can have two pbx call extension seemless to each other ? |
| 14:00.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:00.06 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:04.21 | Katty | hi |
| 14:04.30 | *** join/#asterisk coppice (~chatzilla@m121-202-42-95.smartone-vodafone.com) |
| 14:05.26 | [TK]D-Fender | moos3: What does "seemless" mean in this case? |
| 14:05.41 | [TK]D-Fender | Katty: Mew. |
| 14:06.10 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 14:06.43 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:06.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:14.33 | atan | Can Asterisk convert video formats for video SIP calls? |
| 14:14.54 | atan | Like is one person is using h263, and the other caller is using uh, like, anything else, will Asterisk convert it? |
| 14:15.12 | atan | Or must both devices support the same codec? |
| 14:15.19 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 14:15.52 | AdvoWork | pabelanger, but why could they ring out, but me not ring in, its registering ok, but then going unreachabl |
| 14:16.15 | pabelanger | AdvoWork: I don't know. Show us the problem |
| 14:16.16 | moos3 | [TK]D-Fender, basically have my sip information living in ldap, just need to figure out if say i dial extn 2003 and i'm on pbx1 and say they are on pbx2 will it know enough to ring it on pbx2 and not look for it on pbx1 |
| 14:17.00 | [TK]D-Fender | moos3: What * does with what you dial is in your DIALPLAN. Go make it look up what should actually be called. |
| 14:17.16 | [TK]D-Fender | [09:14]<atan>Can Asterisk convert video formats for video SIP calls? <- No. |
| 14:17.21 | atan | ty |
| 14:17.59 | moos3 | [TK]D-Fender, so i have would have to have the dialplan be smart enough to know how to look up which pbx its registered to ? |
| 14:18.44 | [TK]D-Fender | moos3: Dialplan chooses what to do for every call. Yes it has to call out to the other PBX when appropriate. |
| 14:19.08 | moos3 | [TK]D-Fender, thanks, i'll have to play with it |
| 14:31.43 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 14:33.35 | *** join/#asterisk l0pht (~Excessive@88.251.39.124) |
| 14:34.08 | l0pht | hello |
| 14:34.22 | l0pht | I'm having a problem with AsteriskNow FreePBX system |
| 14:34.44 | l0pht | everything seems to work in the CentOS box, but I'm unable to connect to it |
| 14:34.50 | l0pht | via SIP, that is |
| 14:35.30 | wdoekes2 | ~freepbx |
| 14:35.30 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 14:35.48 | espiceland | Also |
| 14:35.50 | espiceland | ~asterisknow |
| 14:35.50 | infobot | asterisknow is probably based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 14:35.50 | l0pht | thanks |
| 14:39.21 | atan | Is h263 the most widely used video codec for all these random SIP video phones out there? I see Cisco supports it on their 7985, and Nokia supports it, and grandstream supports it... |
| 14:39.38 | atan | And, err, if anyone knows, is the GXV3000 built as cheaply as it looks? :S |
| 14:41.03 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:41.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:41.16 | *** join/#asterisk devmod (~devmod__@c-76-100-208-204.hsd1.md.comcast.net) |
| 14:43.03 | Baylink | Of course it is, it's from Grandstream. :-) |
| 14:43.31 | atan | =) figures =P how's nortel? |
| 14:44.41 | coppice | bankrupt |
| 14:49.21 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:49.28 | atan | coppice, so hardware is cheap now then, eh? |
| 14:50.41 | coppice | you can get some real giveaway offers on nortel phones |
| 14:51.08 | atan | coppice, are they going to be 100% useless in 2 years? |
| 14:52.13 | coppice | isn't everything going to be nearly 100% useless in 2 years? expect even less updates for nortel than you get for most other VoIP kit |
| 14:52.40 | [TK]D-Fender | [09:51]<atan>coppice, are they going to be 100% useless in 2 years? What do you mean IN 2 years? |
| 14:52.47 | [TK]D-Fender | looks right NOW |
| 14:53.04 | *** part/#asterisk AndyRomano (~Adium@195.34.154.20) |
| 14:53.24 | atan | =| okay okay, but I mean the phones are still for sale around town, they seem to support the codecs in use, and they're prices to sell... |
| 14:53.42 | atan | I'd just feel foolish to buy in only to find out it's completely useless to me |
| 14:53.54 | atan | But that being said 900mhz phones still work just fine in some places. =\ |
| 14:54.23 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-rbkksgttoycqrwfy) |
| 14:54.23 | coppice | they are SIP devices. if they work for you they are far from useless right now. if someone kinds a security issues, that might suddenly change |
| 14:54.59 | Chainsaw | (Unless the firmware source code gets released, which is extremely unlikely, the security issue can not be patched) |
| 14:55.25 | atan | I suspect most vulnerabilities should be able to be addressed on the server side of things, no? I mean unless there is an issue with the protocol itself, or their implementation of it... |
| 14:55.33 | atan | digs hole in sand to place head |
| 14:56.35 | *** join/#asterisk srini (~deepak@219.91.201.74) |
| 14:56.44 | srini | Hi all |
| 14:56.52 | Chainsaw | Good afternoon srini. |
| 14:57.02 | srini | Chainsaw Good Afternoon! |
| 14:57.23 | tzafrir_laptop | how usable are those Nortel phones with Asterisk? |
| 14:57.35 | tzafrir_laptop | (chan_unistim?) |
| 14:57.49 | coppice | the SIP versions are available cheap |
| 14:57.51 | srini | I need help in understanding one error : It reads - "Got SIP response 405 "Method Not Allowed" back from"... |
| 14:58.37 | tzafrir_laptop | We already had a customer or two who got an advantage for a cheaper price in replacing Nortel switches, as they could reuse the phones |
| 14:58.46 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 14:59.17 | srini | Also I get to read in the CLI - "No channel type registered for....." and "Unable to create channel of type..." |
| 14:59.42 | tzafrir_laptop | srini, what type? |
| 14:59.46 | srini | SIP |
| 14:59.50 | Baylink | Well, it depends on which Nortel gear you're talking about. If a Norstar is good enough for you, well, I have some of those that have been in-and-working for 20 years or more, and they still work jus' fine. :-) |
| 14:59.55 | srini | tzafrir_laptop: SIP |
| 15:00.00 | tzafrir_laptop | srini, is chan_sip.so loaded? |
| 15:00.01 | atan | tzafrir_laptop, you mean to say the Nortel video phones will be useless for me to attempt to use with Asterisk? |
| 15:00.11 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:00.16 | tzafrir_laptop | Or maybe you have an extra space or so somewhere? |
| 15:00.20 | srini | tzafrir_laptop: I am a newbie - how do I make sure? |
| 15:00.26 | tzafrir_laptop | atan, no idea |
| 15:00.43 | tzafrir_laptop | ~pb |
| 15:00.43 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:00.59 | tzafrir_laptop | srini, please provide a slightly more complete trace |
| 15:01.05 | srini | ok |
| 15:01.17 | *** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
| 15:01.27 | srini | tzafrir_laptop: should the pastebin of the complete CLI dump help? |
| 15:01.35 | tzafrir_laptop | yup |
| 15:03.30 | *** join/#asterisk ickmund_ (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 15:04.09 | srini | tzafrir_laptop: http://asterisk.pastey.net/142604 |
| 15:04.37 | [TK]D-Fender | srini: 'TataSIP:22620 <--- TataSIP is not a CHANNEL type |
| 15:04.57 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 15:04.57 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 15:05.02 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:05.20 | tzafrir_laptop | srini, Unable to create channel of type 'TataSIP:22620:04606@203.196.128.57:5060' (cause 66 - Channel not implemented) |
| 15:05.54 | srini | tzafrir_laptop, [TK]D-Fender: Does this mean there are issues with the trunk definition itself? |
| 15:06.02 | drift- | i have 2 office connections DSL and cable , my asterisk server is on linux router cable network and i'm trying to setup few phones on dsl line to connect to asterisk and some homeoffices also.... my problem is I have one phone working it dials out it recieives calls , but there is no volume... any ideas? asterisk 1.6.4 i think SIP polycom 501 phone centos |
| 15:06.18 | srini | tzafrir_laptop, [TK]D-Fender: SIP registration failure? |
| 15:06.24 | [TK]D-Fender | srini: No, that dial is crap |
| 15:06.34 | [TK]D-Fender | srini: Do you not even know why you put taht in there? |
| 15:06.46 | tzafrir_laptop | srini, what is that line in your dialplan? |
| 15:06.46 | [TK]D-Fender | srini: Dial(tech/resource/numbertodial |
| 15:07.04 | [TK]D-Fender | srini: So what is that junk you are putting instead of SIP/peer, etc? |
| 15:07.41 | srini | tzafrir_laptop, [TK]D-Fender: I will pasty the trunk definition |
| 15:07.45 | [TK]D-Fender | srini: NO |
| 15:07.57 | [TK]D-Fender | srini: Your dial is not starting with a valid CHANNEL TYPE |
| 15:08.10 | [TK]D-Fender | srini: You don;t seem to understand even how to format a DIAL COMMAND |
| 15:08.40 | atan | [TK]D-Fender, do you know of any Windows SIP Video clients I could use to play with for now? |
| 15:08.46 | [TK]D-Fender | srini: this has nothing to do with any other config file so far. That line is bad regardless of anything else |
| 15:08.48 | atan | Until I can find some decent hardware phone :P |
| 15:08.55 | [TK]D-Fender | atan: Ekiga, X-lite |
| 15:09.00 | tzafrir_laptop | tech = CHANNEL TYPE = 'SIP' |
| 15:09.16 | srini | tzafrir_laptop, [TK]D-Fender: http://asterisk.pastey.net/142605 |
| 15:09.34 | [TK]D-Fender | srini: YOU ARE CALLING DIAL WRONG |
| 15:09.39 | Baylink | http://www.gaarde.org/acronyms/?lookup=i |
| 15:09.48 | Baylink | Um, "oops; wrong window" |
| 15:09.55 | tzafrir_laptop | srini, TataSIP is the name of a trunk? |
| 15:10.03 | srini | tzafrir_laptop: Yes |
| 15:10.49 | [TK]D-Fender | srini: exten => _X.,2,Dial(SIP/TataSIP/${EXTEN:2},,tTor) |
| 15:10.51 | tzafrir_laptop | It should be something along the lines of: Dial(SIP/TataSIP/${EXTEN:2},,tTor) |
| 15:11.16 | tzafrir_laptop | It may also be simpler to use 'n' instead of an explicit '2' for the priority |
| 15:11.18 | [TK]D-Fender | srini: You were duplicating auth in yoru dial that exists in your peer anyways and is a bad security risk. AND you pasted all those IP's and passwords in PUBLIC |
| 15:11.24 | [TK]D-Fender | srini: Also not smart... |
| 15:12.15 | tzafrir_laptop | [TK]D-Fender, as in: duplicating it between the 'register =>' statement and the peer section? :-( |
| 15:12.49 | [TK]D-Fender | tzafrir_laptop: No, his previous Dial has al the auth as if to dial direct WITHOUT a peer at all as well as referencing the peer. Hodg-podge mess |
| 15:13.02 | tzafrir_laptop | yeah, I figured it out |
| 15:13.38 | tzafrir_laptop | OTOH, I figure that this account has already run out of credit by now, so I won't try to use it ;-) |
| 15:14.11 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:14.12 | srini | tzafrir_laptop, [TK]D-Fender: I am a newbie :) |
| 15:14.37 | srini | tzafrir_laptop, [TK]D-Fender: what does that 'tTor' stand for? |
| 15:14.46 | [TK]D-Fender | srini: "core show application dial" <- |
| 15:14.48 | [TK]D-Fender | ~book |
| 15:14.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 15:14.50 | [TK]D-Fender | ^^^^^^^ |
| 15:15.15 | BMJ | Also check out http://wiki.asterisk.org |
| 15:15.23 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-4d0baeb0.pool.mediaWays.net) |
| 15:15.35 | tzafrir_laptop | ~docs |
| 15:15.35 | infobot | Asterisk documentation is available at http://wiki.asterisk.org (Official Asterisk Documentation Wiki), the Voip-Info wiki at http://voip-info.org (~voip-info) or Asterisk: The Future of Telephony (~book) |
| 15:27.37 | *** join/#asterisk neurosys (~neurosys@173.200.195.81) |
| 15:28.14 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:28.21 | neurosys | :/ Did an update to 1.6.13, and now my transfer/pickup no longer works on polycom 650. anyone know about this? |
| 15:33.02 | ManxPower | neurosys, no, you didn't. |
| 15:33.08 | ManxPower | 1.6.13 does not exist |
| 15:33.22 | neurosys | Ugh. 1.6.2.13 |
| 15:33.36 | neurosys | I see you're in a better mood today ManxPower :P |
| 15:35.36 | Katty | maybe manx needs a hug |
| 15:35.39 | Katty | hugs ManxPower to see |
| 15:35.54 | Katty | ManxPower: i am sorry you are not feeling in good happy spirits. |
| 15:36.06 | Katty | ManxPower: if it was in my power, i'd go dispose of whatever is bothering you |
| 15:36.14 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:36.59 | neurosys | He'll probably say me :P |
| 15:37.25 | Katty | hugs glomps Naikrovek |
| 15:38.01 | Naikrovek | hugs Katty |
| 15:38.08 | Naikrovek | and SQUEEZES |
| 15:39.28 | ManxPower | Katty, kill the people that wrote fail2ban please |
| 15:40.05 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:40.20 | Chainsaw | ManxPower: It's not a dream to use, agreed. |
| 15:40.43 | p3nguin | I thought fail2ban was a pretty good tool. |
| 15:40.44 | Katty | ManxPower: mkay. weapon of choice? |
| 15:40.46 | ManxPower | Chainsaw, it gets the wrong YEAR when parsing my vsftpd logs |
| 15:40.58 | Katty | lights it on fire. |
| 15:41.05 | ManxPower | And since 365 days in the past is longer then the bantime..... |
| 15:44.23 | Baylink | Hey, Katty; you're gonna set off the smoke alarm in here... geez... |
| 15:45.23 | srini | tzafrir_laptop, [TK]D-Fender: Now, I am trying to dial some number say 555555 and it is going as 91555555! http://asterisk.pastey.net/142606 |
| 15:46.03 | srini | tzafrir_laptop, [TK]D-Fender: What am I doing wrong here! |
| 15:46.17 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 15:46.37 | Katty | Baylink: :P |
| 15:46.41 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:47.17 | *** join/#asterisk metiu (~chatzilla@85-18-228-185.ip.fastwebnet.it) |
| 15:47.26 | metiu | hi all |
| 15:48.23 | metiu | I'm trying to set up some global variables (e.g. priority) during a call, so that when a lower priority call comes in, it gets busy() |
| 15:48.45 | metiu | I was trying to use h as a catch-all extension so that I could reset the priority |
| 15:48.47 | p3nguin | srini: There are several problems with your dial plan. One is that you are using the default context. |
| 15:49.30 | p3nguin | metiu: 'h' is a special extension for "hangup" |
| 15:49.33 | *** part/#asterisk spditner (~simon@syria.uc.org) |
| 15:49.47 | metiu | however, since I'm using nested Dial() s to get to the final extension, it seems that * runs the h extension when it bridges the foremost and the last exensions |
| 15:50.08 | *** join/#asterisk netvient (~chatzilla@209.51.174.61) |
| 15:50.10 | metiu | basically, the call is set up correctly, but the h extension gets run |
| 15:50.12 | *** join/#asterisk ramih (~lokki@ppp-94-68-134-20.home.otenet.gr) |
| 15:50.13 | p3nguin | h is the hangup extension -- it is run when every call ends. |
| 15:50.35 | *** part/#asterisk ramih (~lokki@ppp-94-68-134-20.home.otenet.gr) |
| 15:50.54 | drift- | has anyone ever setup asterisk server to work from internal network to external? |
| 15:50.55 | metiu | well, the call is alive and kicking |
| 15:51.05 | drift- | like taking a phone to another internet connection plugging it in and making it work? |
| 15:51.15 | netvient | hi. Has anyone run into this message: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 ? |
| 15:51.19 | Kobaz | metiu: paste your console debug... with core set verbose 4 |
| 15:51.24 | p3nguin | drift-: That's what most people use Asterisk for. |
| 15:51.32 | drift- | p3nguin i'm in dilema heh |
| 15:51.32 | srini | tzafrir_laptop: I get to listen 'I am sorry thats not a valid extension' I am still with the trunk problems or something else! |
| 15:51.36 | drift- | i have 2 office connections DSL and cable , my asterisk server is on linux router cable network and i'm trying to setup few phones on dsl line to connect to asterisk and some homeoffices also.... my problem is I have one phone working it dials out it recieives calls , but there is no volume... any ideas? asterisk 1.6.4 i think SIP polycom 501 phone centos |
| 15:51.38 | metiu | ok |
| 15:52.01 | drift- | p3nguin: i get the call i can call but i hear nothing on otherside or when i check voicemail no volume :( |
| 15:52.14 | drift- | any idea how i can diagnose the problem? |
| 15:52.27 | neurosys | Anyone else have a problem with transfer/call pickup after updating to 1.6.2.13? |
| 15:53.04 | tzafrir_laptop | srini, you seem to have sent the call to the wrong context? |
| 15:53.50 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:54.20 | Naikrovek | drift-: the problem is not volume, the problem is that you're not receiving audio |
| 15:54.24 | Naikrovek | which means there's a NAT issue |
| 15:54.29 | tzafrir_laptop | srini, Starting Local/8600051@default-ce5d,1 at default,910016045320516,1 failed so falling back to exten 's' |
| 15:56.28 | srini | tzafrir_laptop: Where to look for the fix? extensions.conf? |
| 15:56.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:56.43 | metiu | Kobaz: dialplan excerpt is http://asterisk.pastebin.com/GCU9ZeUB |
| 15:56.52 | tzafrir_laptop | srini, I suppose so |
| 15:57.06 | tzafrir_laptop | What do you have in your default context now? |
| 15:57.12 | metiu | Kobaz: debug out is http://asterisk.pastebin.com/gXvYcgFk |
| 15:57.53 | drift- | yeah i think ur right with nat issue |
| 15:58.10 | drift- | my cable router i enabled DMZ to point to linux router which gives 5 machines internet and phone voip |
| 15:58.17 | drift- | those work flawlessly internal |
| 15:58.22 | metiu | Kobaz: and the starting exten is http://asterisk.pastebin.com/xb4rbFyY |
| 15:58.31 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 15:58.33 | drift- | 220/220 65.2.253.76 D N 49460 Unmonitored <<<<<< external phone |
| 15:58.35 | drift- | <PROTECTED> |
| 15:58.43 | p3nguin | DMZ should never be used for anything on a residential router device. |
| 15:58.44 | drift- | i tried hitting NAT = yes |
| 15:58.56 | srini | tzafrir_laptop: I am not specifying a context in the trunk definition though! |
| 15:59.05 | drift- | home busines i setup linux firewall arno's firewall |
| 15:59.09 | srini | tzafrir_laptop: 'default' should do? |
| 15:59.44 | p3nguin | srini: Context default should never be used for anything you have control over. |
| 15:59.49 | tzafrir_laptop | srini, the stuff under [default] in extensions.conf |
| 16:00.05 | tzafrir_laptop | or: dialplan show default |
| 16:00.12 | drift- | p3nguin what are some things i can test? or check? |
| 16:00.27 | drift- | i've been working on this for like 2 days running out of options n ideas :( |
| 16:00.33 | tzafrir_laptop | And yes, the Book should explain why it's generally not a good idea to use it |
| 16:00.54 | p3nguin | drift-: Configure the router without DMZ. Configure Asterisk and the remote phone appropriately. This is a NAT issue. |
| 16:01.14 | drift- | so forward each port from cable router to linux router manually? |
| 16:02.11 | p3nguin | Why do you want to use two routers both with NAT? |
| 16:03.29 | drift- | i dont want 2 routers |
| 16:03.29 | p3nguin | NAT is known to break RTP, so avoid it whenever possible. |
| 16:03.34 | drift- | its smc cable modem / router thingie |
| 16:03.42 | drift- | and i have linux box doing the router / asterisk |
| 16:03.57 | p3nguin | Can you set it to bridge mode so your Linux router can do its job more effectively? |
| 16:04.03 | atan | Hey guys, more on the Nortel IP1535 video phone, and it's h.263 video codec, is it truly this simple to get it running? According to http://nerdvittles.com/?p=703 they say "Using Asterisk with the Nortel 1535. We have a personal preference for Asterisk, and its a perfect fit with these phones. Just add these entries to sip_general_custom.conf in /etc/asterisk, and video support comes to life in all versions of PBX in a Flash once you restart Asterisk:" |
| 16:04.18 | *** join/#asterisk netvient (~chatzilla@209.51.174.61) |
| 16:04.25 | atan | err /s/ti's/its/ |
| 16:04.30 | atan | gives up at life |
| 16:06.11 | drift- | 1-to-1 Network Address Translation Add/Edit You can add or edit your NAT rules here. |
| 16:06.16 | drift- | public and private ip |
| 16:06.17 | *** join/#asterisk golikwid|mac (~chrislees@67.78.200.57) |
| 16:06.23 | drift- | thats on the cablemodem/router thing |
| 16:07.10 | p3nguin | 1-to-1 is certainly better than a typical home router. |
| 16:07.16 | drift- | ok |
| 16:07.21 | drift- | whats a public ip and private ip? |
| 16:07.33 | Nugget | public ip is one thats routable on the real internet. |
| 16:08.00 | Nugget | private ip is one that's covered by rfc1918 and not routable (10.x.x.x, 172.16.x.x, etc...) like you see behind nat |
| 16:08.25 | p3nguin | public is 174.48.1.164, private is probably 172.16.1.1 |
| 16:08.45 | drift- | Cant use 174.48.1.164 as WAN start IP |
| 16:08.54 | drift- | and my linux router is 172.16.0.3 |
| 16:09.09 | p3nguin | I took a guess at the private one. |
| 16:09.32 | drift- | yeah i understand |
| 16:09.38 | drift- | but it says that cant use ip wan start ip |
| 16:09.44 | p3nguin | Comcast won't want you to change the WAN IP address. |
| 16:10.01 | p3nguin | Probably won't allow it, even. |
| 16:10.07 | drmessano | Are you using an SMC? |
| 16:10.10 | drift- | yes |
| 16:10.10 | drift- | smc |
| 16:10.23 | drmessano | Business class router? |
| 16:10.32 | drift- | yes |
| 16:10.39 | drift- | Comcast Business Gateway |
| 16:10.49 | drift- | Vendor Name SMC Networks Hardware Version 1.01 |
| 16:10.51 | drmessano | Ok, and I see you set up DMZ to the router behind it, yes? |
| 16:11.05 | drift- | i have linux router thats ip of 172.16.0.3 |
| 16:11.09 | drift- | and then another nic in it feeds to switches |
| 16:11.11 | drift- | 172.16.1.1 |
| 16:11.54 | drmessano | Um ok |
| 16:12.05 | metiu | Noone has a clue on why my h(angup) extension is being called before the call ends? |
| 16:12.14 | p3nguin | Are the SMC devices known to break RTP? |
| 16:12.15 | drmessano | So the SMC network is 172.16.x.x? |
| 16:12.19 | drmessano | Nope |
| 16:12.24 | p3nguin | That's good! |
| 16:12.26 | drift- | smc 174.48.1.164 > runs into linux router 172.16.0.3 then nic#2 > 172.16.1.2 is gateway for internet here and phones |
| 16:12.44 | drift- | smc is on 172.16.0.1 |
| 16:12.53 | srini | tzafrir_laptop, [TK]D-Fender: Thanks for all the help! I will continue working on it offline |
| 16:12.53 | drift- | http://172.16.0.1/user/index.asp |
| 16:12.58 | drmessano | ok |
| 16:13.00 | drift- | heh i know u cant connect but thats my link |
| 16:13.01 | drift- | :) |
| 16:13.02 | srini | Thanks for all the help here! |
| 16:13.17 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 16:13.42 | drmessano | Sounds correct.. and you're using externip of 174.48.1.164 in Asterisk? |
| 16:13.54 | drift- | that is good question |
| 16:14.03 | drift- | <PROTECTED> |
| 16:14.04 | drmessano | Well, check the config |
| 16:14.17 | drmessano | get rid of nat=route |
| 16:14.18 | drmessano | nat=yes |
| 16:14.18 | drift- | my sipl.conf |
| 16:14.21 | drift- | ok |
| 16:14.32 | drift- | done |
| 16:14.35 | drift- | i dissabled dmz also |
| 16:14.43 | drmessano | disabled DMZ where? |
| 16:14.47 | drift- | on the smc |
| 16:14.53 | drmessano | no |
| 16:15.04 | drift- | ok turn it back on? |
| 16:15.05 | drmessano | You MUST enable DMZ on the SMC or this will NEVER work |
| 16:15.12 | drift- | got it |
| 16:15.33 | drift- | okay its on |
| 16:15.41 | drmessano | The router behind the SMC MUST be in the DMZ |
| 16:15.46 | drmessano | Ok, good |
| 16:15.47 | drift- | got it |
| 16:15.48 | drift- | :) |
| 16:16.00 | drift- | <PROTECTED> |
| 16:16.07 | p3nguin | Why not use 1-to-1 to the Linux router behind it? |
| 16:16.19 | drmessano | Because that won't work with the SMC |
| 16:16.47 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 16:16.56 | p3nguin | 1-to-1 has to be the next best thing to being able to bridge it, no? |
| 16:17.12 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:17.25 | metiu | what I see is that the original context's h(angup) extension gets called only when the call finishes, while the nested channel calls are hung up in some way, is that a consequence of * bridging the first and last calls? |
| 16:17.40 | *** part/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:18.27 | drmessano | No, in the SMC, DMZ is the best way to go |
| 16:18.33 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:18.49 | *** join/#asterisk yonahw (~yonahw@www.mcatrack.com) |
| 16:19.26 | drift- | so what do i do next? |
| 16:19.55 | drmessano | Next is your config in the secondary router |
| 16:20.03 | drmessano | Make sure you have ports forwarded, etc |
| 16:20.18 | drift- | well port 5060 is open |
| 16:20.19 | metiu | Kobaz: nevermind, I got it, I was missing the /n in the Dial() cmd, so the recursive call was becoming zombie etc etc... |
| 16:20.35 | drmessano | and you need the rtp ports open |
| 16:20.42 | drift- | what are those? |
| 16:21.26 | drift- | rtpstart=10000 rtpend=20000 |
| 16:21.35 | drmessano | yeah |
| 16:21.51 | drift- | so open 10000 and 20000 |
| 16:22.27 | drift- | ok give me min looking for ipchains command for centos |
| 16:23.05 | drift- | udp correct? |
| 16:23.07 | drift- | i mean iptables |
| 16:23.09 | yonahw | hi, I am trying to provision a polycom 601 from an ftp server. I have been trying to follow the esoteric docs from polycom yet the configuration doesn't take. I have pasted configs and logs to http://pastie.org/1281889. Phone has accepted new bootrom and sip.ld but it doesn't seem to be picking up the config files which are specified in 000000000000.cfg. Any ideas on what my next step should be? Should I just be making a macaddress.cfg? Supposedly ... |
| 16:23.15 | yonahw | ... I can just override in the 000000000000.cfg. |
| 16:23.46 | drift- | iptables --append INPUT --protocol ALL --source-port 10000 --jump ACCEPT |
| 16:23.49 | drift- | looks correct :) |
| 16:24.45 | WIMPy | You need all 10001 ports and only udp. |
| 16:25.28 | WIMPy | However if you use the proticol helper for sip, a RELATED should be enough. |
| 16:25.56 | p3nguin | drift-: 10000-20000 UDP, not 10000 and 20000. |
| 16:26.03 | drift- | ah got it |
| 16:26.08 | drift- | looking for iptables command now |
| 16:26.18 | drift- | iptables -A INPUT -p udp -m multiport --dports 10000-20000 -j ACCEPT |
| 16:26.40 | drift- | this is for inbound or outbound? |
| 16:26.42 | drift- | or both? |
| 16:26.45 | WIMPy | That looks better |
| 16:27.05 | WIMPy | Out could be anything. |
| 16:27.13 | *** join/#asterisk slackytude (~slacky@drms-4d000237.pool.mediaWays.net) |
| 16:27.24 | p3nguin | If your outbound default policy is to allow everything, then you don't have to worry about it. |
| 16:27.32 | drift- | [root@localhost asterisk]# iptables -A INPUT -p udp -m multiport --dports 10000-20000 -j ACCEPT iptables v1.3.5: invalid port/service `10000-20000' specified |
| 16:27.42 | p3nguin | 10000:20000 |
| 16:27.43 | drift- | ok |
| 16:27.46 | drift- | so i got it iptables -A INPUT -p udp -m multiport --dports 10000:20000 -j ACCEPT |
| 16:27.47 | drift- | yeah |
| 16:27.48 | drift- | :D |
| 16:27.52 | drift- | so retart phone and test it? |
| 16:29.09 | [TK]D-Fender | yonahw: 00000000.cg = worthless. it is applied ONCE ever until the phone reads its own. |
| 16:29.16 | [TK]D-Fender | yonahw: Same goes for the directory, etc. |
| 16:29.22 | [TK]D-Fender | yonahw: always make mac specific |
| 16:29.40 | drift- | so now that ports 10000 thru 20000 are open... |
| 16:29.45 | drift- | anytihng else i should do? |
| 16:30.06 | p3nguin | drift-: 5060 is for signaling, 10000-20000 is for audio. What more do you want? |
| 16:30.20 | drift- | not sure lets find out if it works restarting phone asterisk |
| 16:30.21 | Qwell | media* |
| 16:30.38 | p3nguin | s/audio/media/ |
| 16:31.42 | yonahw | [TK]D-Fender: thanks, seems confusing because I read in a few polycom docs to just use the 000000000000.cfg |
| 16:31.45 | drift- | well |
| 16:31.49 | drift- | no audio still |
| 16:31.50 | drift- | :( |
| 16:32.02 | drift- | 220/220 65.2.253.76 D N 49460 Unmonitored |
| 16:32.20 | AdvoWork | is there a way to get notified if a phone goes unregistered? |
| 16:34.15 | drift- | p3nguin is there anything else i'm missing? |
| 16:36.44 | p3nguin | I'm not familiar with the SMC configuration that drmessano was trying to achieve, so I don't know. |
| 16:37.18 | *** join/#asterisk troubled (~troubled@unaffiliated/troubled) |
| 16:37.32 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 16:39.31 | drift- | drmessano... any suggestions? |
| 16:40.13 | ruyo | drift-, do you have externip and localnet in your sip.conf? |
| 16:40.29 | drift- | externip=174.48.1.164 localnet=172.16.1.2/255.255.255.0 |
| 16:40.39 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 16:40.56 | drift- | should externalip be 172.16.0.3? |
| 16:41.04 | drift- | cause there are 2 nics in this router |
| 16:41.23 | drift- | linux router that is 172.16.0.3 which connects to the live ip with smc and then 172.16.1.2 which is the gateway |
| 16:41.30 | drift- | for machines and phones |
| 16:41.36 | ruyo | Your externip should be whatever IP says on http://www.whatismyip.com/ |
| 16:41.42 | drift- | got it |
| 16:41.44 | drift- | then its correct |
| 16:45.09 | ruyo | drift-, then your Asterisk configuration seems correct. Try catching the media packets to see if they are going to the correct IP or if they are being translated to something that's wrong. |
| 16:45.16 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 16:45.22 | drift- | how can i do that? |
| 16:45.39 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 16:45.53 | ruyo | Your call is placed, the only problem is the sound, is that right? |
| 16:46.03 | drift- | yeah i can recieve phonecalls |
| 16:46.06 | drift- | and i can make phonecalls |
| 16:46.11 | drift- | nobody can hear me i cant hear them |
| 16:46.17 | drift- | when i dial 10 for voicemail i dont hear it ask me for password |
| 16:46.19 | drift- | but call does happen |
| 16:46.32 | drift- | cause log shows that its waiting for password |
| 16:47.01 | ruyo | You're placing a call from outside Asterisk's lan then? |
| 16:47.05 | drift- | yes |
| 16:47.20 | drift- | on the DSL line |
| 16:47.33 | drift- | 220/220 65.2.253.76 D N 49460 Unmonitored |
| 16:47.44 | drift- | 211/211 172.16.1.162 D 5060 Unmonitored < internal |
| 16:48.08 | Kyosh | can you pastebin it? |
| 16:48.31 | ruyo | In that case, the problem seems to be in your NAT rules. |
| 16:48.46 | drift- | well i have SMC router doing DMZ |
| 16:49.02 | drift- | pointing to 172.16.0.3 which is nic #1 and nic#2 is 172.16.1.2 |
| 16:49.26 | drift- | i opened ports 10000 - 20000 |
| 16:49.27 | ruyo | Most DMZ suck. They usually don't actually do DMZ. |
| 16:49.29 | drift- | 5060 is open |
| 16:49.38 | drift- | this is smc business class smc |
| 16:50.26 | ruyo | Try using tcpdump to see if the Asterisk box actually gets any media packets. |
| 16:55.06 | drift- | ruyo how would i do that heh |
| 16:55.10 | yonahw | [TK]D-Fender: what needs to go into the mac_address.cfg? Is there a working sample config you can point me to please? My Google fu is weak this morning. |
| 16:55.14 | drift- | i tried tcpdump -t udp |
| 16:56.27 | [TK]D-Fender | yonahw: ADMIN GUIDE, not Google |
| 16:56.50 | [TK]D-Fender | yonahw: And the same sort of stuff you see in the sample 00000000.cfg |
| 16:56.50 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:57.27 | atan | yonahw, are you messing with a cisco phone by chance? |
| 16:57.36 | atan | I can't see/find your previous lines of chat |
| 16:57.59 | yonahw | [TK]D-Fender: Phone isn't picking up the additional config files. FTP transfer log shows no requests for them |
| 16:58.06 | yonahw | atan: Polycom 601 |
| 16:58.27 | atan | [TK]D-Fender, if one is pushing video over h264 should they bother picking up an expensive new HD webcam, or would the standard cheapo $10 webcam do the trick just fine? |
| 16:58.28 | [TK]D-Fender | yonahw: And I have no reason to believe you ever pointed it in the right direction in teh frist place |
| 16:59.01 | atan | yonahw, darn. I have not played with one of those yet so I must bow down. I did mess with my Cisco using tftp to load the configs so I thought I'd ask just for the heck of it =) best of luck!! =D |
| 17:00.31 | yonahw | [TK]D-Fender: what do you mean by that? It picks up the macaddress.cfg but the rest of the configs listed in CONFIG_FILES is ignored. My macaddress.cfg is pasted at http://pastie.org/1281994 |
| 17:01.13 | yonahw | atan: thanks, I have once gotten this to work many moons ago but I must be doing something really stupid this time around. Been playing with this for days and getting nowhere. |
| 17:01.16 | [TK]D-Fender | yonahw: CONFIG_FILES="phone10004f202c853.cfg, server.cfg, phone1_316.cfg sip_316.cfg" <-- inconsistent commas, and I don't SE your files. |
| 17:01.37 | [TK]D-Fender | yonahw: I have no proof of what exists |
| 17:02.08 | yonahw | [TK]D-Fender: thanks for pointing out the missing comma. I will edit that pastie in a moment with the rest of the configs once I fix that comma. |
| 17:02.09 | [TK]D-Fender | yonahw: Also no prrof of what file you even pasted there. |
| 17:02.12 | Katty | peeks in |
| 17:02.16 | Katty | HELLLLLLLLLLLLLOOOOOOOOOOOOOOOOOOOOOOOOOOO nurse. |
| 17:02.22 | Katty | disappears |
| 17:04.14 | drift- | ruyo |
| 17:04.20 | drift- | i have the tcpdump can i paste it here? |
| 17:04.21 | drift- | or pastebin? |
| 17:04.47 | drift- | i made the phone call |
| 17:04.48 | drift- | http://pastebin.com/PLfz3sJM |
| 17:05.29 | yonahw | [TK]D-Fender: I edited the pastebin with the rest of the configs. http://pastie.org/1281994 |
| 17:05.56 | yonahw | the phone1_316.cfg and sip_316.cfg are the defaults that came from polycom unedited |
| 17:06.41 | yonahw | [TK]D-Fender: thank you for spending time on this with me I really appreciate the assistance. |
| 17:07.20 | [TK]D-Fender | yonahw: I see no FTP access logs, etc... |
| 17:07.41 | drift- | ok how can i determine the nat issue ... ;( |
| 17:08.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 17:08.49 | Defraz | Does anyone have any experience an ADTRAN Total Access 924e. I am trying to have my PRI from my Telco connect to that then SIP any calls over to my Asterisk Server |
| 17:09.28 | yonahw | [TK]D-Fender: sorry updated the pastie again with ftp log and boot.log from phone |
| 17:09.32 | ruyo | drift-, try something like "tcpdump -i eth0 -n -s0 -w somefilename". |
| 17:12.05 | [TK]D-Fender | yonahw: 0101012510|copy |4|00|Server '192.168.0.189' said '2345-11605-001.sip_316.ld' is not present |
| 17:12.29 | yonahw | [TK]D-Fender: I noticed that but it then picks up the sip_316.ld and seems to load that fine |
| 17:12.39 | [TK]D-Fender | yonahw: And go look at the QAPP log <- |
| 17:12.42 | *** join/#asterisk angav (~angav@189.251.14.98) |
| 17:12.57 | yonahw | where can I find the QAPP log? I don't know what that is. |
| 17:13.01 | p3nguin | drift-: If it's a NAT issue, rtp debug will probably show the wrong IP address in on the packets. |
| 17:13.32 | yonahw | [TK]D-Fender: also the zip I downloaded from Polycom didn't have a model specific sip.ld for 3.1.6 which is the latest the 601 supports |
| 17:14.48 | angav | Hi!, CentOS 5.5 + AStersik 1.6.2 + Freepbx 2.7.0.6: Intended to install and configure squid to control networwk traffic and reserve calls bandwidht. Any known issues or advices? |
| 17:15.33 | *** join/#asterisk sbszulu (~dundubala@41.16.16.51) |
| 17:15.40 | [TK]D-Fender | yonahw: Go look at the other logs from the phone |
| 17:16.54 | yonahw | [TK]D-Fender: where can I find the other logs from the phone. The boot.log is the only file it uploaded to log/ |
| 17:17.35 | [TK]D-Fender | <APPLICATION APP_FILE_PATH="sip_316.ld" CONFIG_FILES="phone10004f202c853.cfg, server.cfg, phone1_316.cfg sip_316.cfg" <- fix yoru commas |
| 17:18.03 | [TK]D-Fender | yonahw: And dump the entire damn folder. I can't prove WHERE the files are, who owns them, etc |
| 17:18.26 | yonahw | [TK]D-Fender: I fixed it on the ftp server just didn't edit the pastie. I will dump an ls -la of the folder and log |
| 17:19.21 | atan | Is it possible to secure SCCP/skinny so it can be used securely over the internet kind of like SIP is? =\ |
| 17:19.44 | *** join/#asterisk wizhippo (~Adium@64.201.57.7) |
| 17:20.04 | p3nguin | atan: I think most people use a VPN for that. |
| 17:20.08 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 17:20.29 | atan | p3nguin, well that could be an option... hmm. |
| 17:20.32 | wizhippo | is there a way to make an agent of the call queue have to press pound before the call is connected? |
| 17:20.39 | yonahw | [TK]D-Fender: I have updated the same pastie. Fixed the comma in the pastie and on the bottom listed the directory contents |
| 17:21.04 | atan | p3nguin, thanks for the idea there. I think I could make that work easily for me using DD-WRT. It has built-in support for OpenVPN I do believe. |
| 17:21.15 | atan | I could just tunnel all traffic on that interface in to the server. |
| 17:21.38 | atan | p3nguin, you don't happen to know which ports it runs on? Like, for example, should I only limit port 2000 to vpn interface access only? |
| 17:21.47 | atan | Would that cover my behind for the most part? |
| 17:22.00 | [TK]D-Fender | yonahw: Your files are owned by ROOT. WTF <----- |
| 17:22.04 | [TK]D-Fender | yonahw: Fix this up |
| 17:22.18 | russellb | calm your nerves |
| 17:22.25 | netvient | "Bad address cast to IPv4" ? |
| 17:22.36 | Qwell | netvient: context |
| 17:22.59 | netvient | iax2 client registers, 2 Asterisk servers |
| 17:23.05 | netvient | both 1.8 |
| 17:23.07 | yonahw | [TK]D-Fender: I understand that isn't optimal but it does pick up the 0000000000.cfg and the macaddress.cfg which have the same ownership |
| 17:23.14 | russellb | that has been fixed in the 1.8 branch, it will be in 1.8.1 |
| 17:23.16 | Qwell | netvient: have the whole message? |
| 17:23.18 | netvient | in the -r console, this: |
| 17:23.21 | russellb | for now you can check out the 1.8 branch from svn |
| 17:23.23 | Qwell | well then! |
| 17:23.48 | netvient | russellb - is that for my prob your talking about ? |
| 17:23.53 | netvient | han_iax2.c:2304 peercnt_modify: Bad address cast to IPv4 |
| 17:24.10 | netvient | there is that message, but others too - from iax2_getpeername |
| 17:24.16 | netvient | and update_registry |
| 17:24.42 | russellb | yes |
| 17:24.56 | p3nguin | atan: If possible, configure the phone to send all its traffic across the VPN. |
| 17:24.56 | netvient | ok great, thank-you |
| 17:24.57 | russellb | $ svn co http://svn.digium.com/svn/asterisk/branches/1.8 |
| 17:25.38 | atan | p3nguin, well the router would deal with that part fine I'm just thinking about how skinny would be configured on asterisk. I would hate to leave a door wide open without knowing it. |
| 17:25.58 | atan | But if the VPN interface, maybe eth1, I suppose there is something within Asterisk that selects what to bind it to |
| 17:26.06 | atan | reads skinny conf |
| 17:26.23 | atan | Yes yes, here we go |
| 17:26.23 | atan | bindaddr=0.0.0.0; Address to bind to |
| 17:26.42 | drift- | hrm silent :( |
| 17:27.42 | yonahw | [TK]D-Fender: fixed the file ownership and repasted output. Phone is rebooting. Hoping it likes it this time. Any insight into why it's not uploading any other logs? Maybe I have the wrong sip.ld? I downloaded it from http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_6_legacy_release_sig_split.zip |
| 17:28.01 | yonahw | that link came from http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip601.html |
| 17:28.21 | drift- | http://pastebin.com/PLfz3sJM |
| 17:28.33 | drift- | how do i do rtp debut? |
| 17:28.34 | drift- | debug |
| 17:28.41 | drift- | let me google that one heh |
| 17:29.33 | p3nguin | "rtp debug" on the CLI |
| 17:29.53 | drift- | ok |
| 17:30.04 | drift- | its scrolling like crazy |
| 17:31.07 | p3nguin | You only need a couple lines to see what IP address is being used. |
| 17:31.13 | [TK]D-Fender | yonahw: reboot the phone |
| 17:32.05 | drift- | should i try calling? |
| 17:32.12 | drift- | and do this rtp thing |
| 17:32.24 | p3nguin | If rtp debug is scrolling, something is already on a call. |
| 17:32.35 | yonahw | [TK]D-Fender: it reboots as soon as the config failed. I just realized that I probably needed the combined packaged instead of the split. I am uploading the newly downloaded sip.ld's and will try again |
| 17:32.49 | [TK]D-Fender | yonahw: Shouldn't need the app at all |
| 17:33.02 | [TK]D-Fender | yonahw: probelm is not picking up the configs |
| 17:33.18 | drift- | Sent RTP P2P packet to 74.63.41.218:14836 (type 00, len 000160) Sent RTP P2P packet to 74.63.41.218:14836 (type 00, len 000160) Sent RTP P2P packet to 74.63.41.218:14836 (type 00, len 000160) Sent RTP P2P packet to 74.63.41.218:14836 (type 00, len 000160) Sent RTP P2P packet to 74.63.41.218:14836 (type 00, len 000160) |
| 17:33.19 | drift- | bam got it |
| 17:33.21 | drift- | i made that call |
| 17:33.56 | *** part/#asterisk [T]ank (~chwall@206.71.78.158) |
| 17:33.56 | p3nguin | I don't see a private address in the rtp, so that's good. |
| 17:34.45 | yonahw | [TK]D-Fender: phone just goes through a loop of downloading bootrom, app, master config and failing on config at which point it reboots. I would think that the problem must be in my configs but I just don't see what could be wrong. I would think there would be a SIP log but don't see anything. You mentioned before a QAPP log, what is that and where can I find it? |
| 17:35.16 | [TK]D-Fender | APP, not QAPP |
| 17:35.35 | drift- | p3nguin coo |
| 17:35.38 | [TK]D-Fender | yonahw: And it isn's saving. If you are looping then you have a corrupted config |
| 17:35.49 | p3nguin | Is your remote phone peer properly configured? I assume it is behind NAT since it is on DSL... but did you configure nat=yes and canreinvite=no for the peer in sip.conf? |
| 17:35.54 | [TK]D-Fender | yonahw: reverse out your changes till it starts working. |
| 17:36.47 | yonahw | [TK]D-Fender: I never provisioned this phone correctly. I updated the bootrom and the app and dropped these initial configs. It hasn't once accepted the configs |
| 17:36.55 | atan | Can you set your caller id / name on outbound calls using Google's PTSN service? |
| 17:37.20 | [TK]D-Fender | yonahw: put in STOCK configs and do minimal chanegs. You have split off multiple override files, etc. |
| 17:37.28 | drift- | p3nguin i did type nat=yes |
| 17:37.34 | drift- | let me check careinvintei |
| 17:37.58 | yonahw | [TK]D-Fender: Thanks, will do |
| 17:38.04 | drift- | <PROTECTED> |
| 17:38.07 | drift- | type=peer |
| 17:38.09 | drift- | host=dynamic |
| 17:38.15 | drift- | secret=password |
| 17:38.45 | drift- | 220/220 65.2.253.76 D N 49460 Unmonitored |
| 17:38.50 | drift- | it shows NAT no tho :( |
| 17:39.06 | p3nguin | drift-: The N under NAT meant it is set to use NAT. |
| 17:39.08 | jermudgeon | got his redfone working with dahdi-dynamic-ethmf, and is happy |
| 17:39.12 | drift- | ah ok |
| 17:39.19 | drift- | then thats good then :) |
| 17:39.39 | russellb | yes, and it's painfully confusing |
| 17:39.49 | russellb | I hate that it's "N" or blank, heh |
| 17:40.56 | atan | Is there a config option for asterisk that tells how many lines/calls a SIP peer can have? |
| 17:41.07 | p3nguin | call-limit |
| 17:41.24 | atan | So I could set call-limit:3; would prevent them from having more than 3 calls at any given time? |
| 17:41.27 | atan | Would ring busy, per se? |
| 17:41.51 | drift- | p3nguin: any other ideas? |
| 17:42.00 | drift- | heh we opened ports 10000 - 20000 |
| 17:42.01 | p3nguin | call-limit=3 would. |
| 17:42.36 | p3nguin | drift-: How about the canreinvite setting? You didn't show it above. |
| 17:42.39 | drift- | p3nguin check this out! |
| 17:42.46 | drift- | i dunno what this is phone 201 |
| 17:42.51 | drift- | <PROTECTED> |
| 17:42.55 | drift- | but its using outside ip and port 5061 |
| 17:42.57 | drift- | how the hell |
| 17:43.15 | drift- | i'm trying to setup 220/220 65.2.253.76 D N 49460 Unmonitored < and its using 49460 ? |
| 17:45.12 | drift- | p3nguin: i put careinvite=no |
| 17:45.42 | p3nguin | Make sure you always save the changes and run sip reload after changing something in sip.conf. |
| 17:46.10 | p3nguin | Hopefully you didn't typo it in the config. |
| 17:46.38 | p3nguin | canreinvite = can re-invite .... care invite = fail |
| 17:47.53 | drift- | [220] type=peer host=dynamic secret=xxxx context=users4 nat=yes careinvite=no |
| 17:48.35 | p3nguin | fail |
| 17:48.44 | p3nguin | CAN RE INVITE |
| 17:48.47 | drift- | ok |
| 17:48.48 | drift- | i just did it |
| 17:48.50 | drift- | restarted asterisk |
| 17:48.52 | drift- | test agian? |
| 17:48.54 | p3nguin | not CARE INVITE |
| 17:49.08 | drift- | <PROTECTED> |
| 17:49.12 | p3nguin | okay |
| 17:49.18 | p3nguin | save, run sip reload |
| 17:49.24 | p3nguin | test again. |
| 17:49.29 | drift- | i dont have to do restart now? |
| 17:49.32 | p3nguin | no |
| 17:49.33 | p3nguin | sip reload |
| 17:49.36 | drift- | no shit sip reload lol this entire time i been booting people off phones |
| 17:49.38 | drift- | LOL |
| 17:49.53 | drift- | learn something new everyday ;) |
| 17:49.57 | drift- | testing... |
| 17:50.31 | drift- | negative |
| 17:50.33 | drift- | no audio still :( |
| 17:52.54 | Katty | hai |
| 17:53.20 | Katty | what's the good word, yos |
| 17:54.17 | drift- | p3nguin no go :( |
| 17:54.38 | p3nguin | pewp |
| 17:54.54 | robl^laptop | katty: a good word!? I have always been fond of "supercalifragilisticexpialidocious". It's a very good word. |
| 17:55.07 | Katty | kay |
| 17:55.18 | Katty | infobot: bird |
| 17:55.18 | infobot | i heard bird is the word. |
| 17:55.38 | Katty | infobot: forget bird |
| 17:55.39 | infobot | Katty: i forgot bird |
| 17:55.45 | Katty | infobot: bird is the word. it's got groove. it's got meaning. |
| 17:55.45 | infobot | okay, Katty |
| 17:55.52 | tzafrir_laptop | OT: http://laforge.gnumonks.org/weblog/2010/11/07/#20101107-all_your_baseband_are_belong_to_us |
| 17:57.12 | wizhippo | anyone know how to create a macro in pbx_lua? i want call a lua routine from dial using M option |
| 17:57.12 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 18:05.09 | p3nguin | katty: Did you feel any shaking over there last night? |
| 18:07.09 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 18:07.17 | Katty | p3nguin: nope, i was passed out |
| 18:07.22 | Katty | p3nguin: rumor has it there was an earthquake tho |
| 18:07.46 | p3nguin | That was sort of the reason I asked. |
| 18:08.34 | Katty | checks |
| 18:09.04 | Katty | found it |
| 18:09.07 | Katty | http://earthquake.usgs.gov/earthquakes/recenteqsus/Quakes/nm110810a.php |
| 18:10.01 | Katty | apparently it was just a little 2.8 |
| 18:15.47 | *** join/#asterisk DennisG (~DennisG@541E88D0.cm-5-7c.dynamic.ziggo.nl) |
| 18:17.25 | *** join/#asterisk shapr (~shapr@nat/digium/x-vgqxjonmjkwjczna) |
| 18:18.02 | shapr | Does Asterisk support the SIMPLE SIP extensions? At least to the point of passing through SIP MESSAGE method calls to other SIP endpoints? |
| 18:19.41 | [TK]D-Fender | shapr: No. * is not a messaging platform |
| 18:20.06 | *** join/#asterisk acxty (~acxty@200.107.239.55) |
| 18:20.20 | acxty | HI guys, does someone knows of a open source webphone |
| 18:22.54 | drift- | p3nguin: any other things i can try? with this nat issue? |
| 18:25.07 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:26.29 | shapr | [TK]D-Fender: Have you ever tried sending a SIP MESSAGE method call during an active call? |
| 18:28.00 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 18:30.30 | *** part/#asterisk wizhippo (~Adium@64.201.57.7) |
| 18:31.34 | p3nguin | drift-: No clue. Maybe we can look at a sip debug of a call with no audio? |
| 18:31.49 | drift- | how can check if these ports opened up 10000-20000 |
| 18:31.55 | drift- | [root@localhost asterisk]# iptables -A INPUT -p udp -m multiport --dports 10000:20000 -j ACCEPT [root@localhost asterisk]# |
| 18:32.03 | shapr | [TK]D-Fender: Looking at channels/chan_sip.c line 12519 (Asterisk 1.8.0), it appears that sending SIP MESSAGE with text is supported. |
| 18:32.24 | p3nguin | drift-: iptables -L INPUT -nv will show what rules you have in the chain. |
| 18:33.02 | [TK]D-Fender | shapr: While ni a CALL perhaps, but not as far as passing a message from phone to phone without being on a call. |
| 18:33.08 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 18:33.24 | drift- | <PROTECTED> |
| 18:33.46 | p3nguin | I've never used multiport, so I don't know how it affects things. |
| 18:34.22 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:34.56 | p3nguin | Asterisk is running on the host where you are issuing the iptables commands, right? |
| 18:36.46 | ManxPower | does it work if you remove uptables? |
| 18:36.47 | p3nguin | Err, I meant I would never use multiport to specify a single port or port range. |
| 18:37.02 | drift- | hrm |
| 18:37.06 | drift- | how would you do it then? |
| 18:38.46 | p3nguin | Well, you aren't listing multiple ports (or port ranges), so just run iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT |
| 18:39.14 | p3nguin | I doubt your multiport specification had any negative effect on the rule, though. |
| 18:39.24 | drift- | ok done |
| 18:39.25 | p3nguin | I can't see how it would break anything. |
| 18:39.43 | p3nguin | I was just saying I don't use it for a single port or range, so I don't know how it affects anything. |
| 18:40.06 | p3nguin | (1234.56) <p3nguin> Asterisk is running on the host where you are issuing the iptables commands, right? |
| 18:40.10 | Katty | what's the name of the room where zeeek has his friday thing |
| 18:40.14 | p3nguin | Still wanting an answer to that question. |
| 18:40.24 | p3nguin | katty: #vuc |
| 18:40.29 | Katty | oh yeah. thanks |
| 18:40.35 | p3nguin | VoIP Users' Conference |
| 18:41.05 | drift- | p3nguin yes |
| 18:41.12 | p3nguin | Good. |
| 18:41.12 | drift- | linux router / asterisk server same centos machine |
| 18:42.48 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 18:43.18 | ManxPower | does it work without the firewall enabled? |
| 18:43.24 | p3nguin | When you call to and from the problem phone, does it make other phones ring and does it ring? |
| 18:43.56 | drift- | yeah |
| 18:44.06 | drift- | if call from internal network to extension it rings i can pick up no audio |
| 18:44.13 | drift- | i can call out to my cell phone no audio |
| 18:44.15 | ManxPower | so it does not work then |
| 18:44.20 | drift- | it does work |
| 18:44.22 | drift- | just no audio |
| 18:44.26 | drift- | nobody can hear me i cant hear them |
| 18:44.28 | ManxPower | no audio == not work |
| 18:44.52 | ManxPower | put disallow=all and allow=ulaw and canreinvite=no in sip.conf [general] |
| 18:45.13 | drift- | [general] port=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw |
| 18:45.15 | drift- | allrdy ahead of you |
| 18:45.22 | ManxPower | and the canreinvite? |
| 18:45.31 | ManxPower | you don't want a port and bindaddr |
| 18:45.32 | drift- | canreinvite is setup to no on extension 220 |
| 18:45.52 | drift- | so take port out? |
| 18:46.03 | drift- | or bindaddr? |
| 18:46.05 | ManxPower | neither option does anything useful during your testing |
| 18:46.36 | ManxPower | but don't bother with iptables until you can get the phones actually talking to each other. |
| 18:46.55 | drift- | well what do you mean |
| 18:47.01 | drift- | i have DSL connection and CABLE connection |
| 18:47.05 | drift- | dsl is extension 220 |
| 18:47.09 | drift- | cable is main asterisk with 10 other phones |
| 18:47.18 | ManxPower | You did not say that. |
| 18:47.32 | ManxPower | does it work between phones on the same server? |
| 18:47.45 | drift- | if i put it on cable network yes |
| 18:47.48 | drift- | if i put it on my dsl no |
| 18:47.58 | ManxPower | Why don't you try getting the local phones to work before trying something this complex? |
| 18:48.03 | drift- | they do all work |
| 18:48.13 | drift- | i want this phone to be working at home when i bring it home |
| 18:48.28 | ManxPower | enable sip debug, reproduce the issue and pastebin the output |
| 18:48.35 | drift- | ok |
| 18:49.12 | ManxPower | I hope you are not using some Asterisk GUI |
| 18:49.27 | drift- | nope |
| 18:49.43 | ManxPower | so your dialplan should be nice and simple and easy to read in the debug |
| 18:49.49 | drift- | http://pastebin.com/hfLM3Q5L |
| 18:50.54 | ManxPower | looks like nat to me <--- Transmitting (NAT) to 65.2.253.76:49460 ---> |
| 18:51.34 | drift- | so how can i fix this? |
| 18:52.03 | ManxPower | is the asterisk server directly on the internet with a public IP address? |
| 18:52.12 | drift- | smc router/cable modem |
| 18:52.19 | drift- | does DMZ to linuxrouter/asterisk |
| 18:52.35 | ManxPower | then your setup is too complicated for me to spend time on. Here is some nat pointers |
| 18:52.37 | ManxPower | ~sipnat |
| 18:52.37 | infobot | extra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:53.10 | ManxPower | BTW, DMZ means NAT |
| 18:53.19 | drift- | 174.48.1.164 < smc cablemodem goes to > 172.16.0.3 nic#1 > 172.16.1.2 > rest of phones and computers |
| 18:53.56 | drift- | and then i have phone on DSL 220/220 65.2.253.76 D N 49460 Unmonitored |
| 18:54.08 | drift- | <PROTECTED> |
| 18:54.11 | ManxPower | then you need to have the externip= localnet= set |
| 18:54.35 | drift- | externip=174.48.1.164 localnet=172.16.1.2/255.255.255.0 |
| 18:54.57 | p3nguin | That localnet looks wrong. |
| 18:55.14 | p3nguin | localnet=172.16.0.0/255.255.0.0 |
| 18:55.30 | drift- | hrm |
| 18:55.30 | p3nguin | You have things on both 172.16.0 and 172.16.1 |
| 18:55.34 | drift- | yeah |
| 18:56.01 | drift- | ok changed it |
| 18:56.06 | drift- | going to test |
| 18:56.07 | p3nguin | save, sip reload |
| 18:56.42 | *** join/#asterisk devmod (~devmod__@c-76-100-208-204.hsd1.md.comcast.net) |
| 18:57.13 | drift- | no go :( |
| 18:59.09 | devmod | Where cn I find the branch mentioned on this video? SIP/RTMP http://www.youtube.com/watch?v=3h6-PSpD-Oc ? |
| 19:02.15 | drift- | p3nguin? heh |
| 19:02.27 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 19:06.26 | p3nguin | I use a custom Linux-based router firewall and my remote devices work fine. One time, I installed a Cisco 831 in place of the Linux machine because I was trying to take up less space and use less power... it broke at least one remote SIP device. |
| 19:06.41 | *** join/#asterisk DoDaT69 (~DoDaT69@173.160.86.155) |
| 19:06.49 | drift- | heh |
| 19:07.04 | p3nguin | The solution: take the Cisco back out and put the Linux box back in place. |
| 19:07.44 | p3nguin | No one here was able to (or maybe not willing to) help me fix the problem and keep using the Cisco device. |
| 19:08.04 | Katty | peeks in |
| 19:08.19 | Katty | if anyone is interested in participating in the asterisk christmas card exchange, please /query me for details! |
| 19:08.35 | p3nguin | Now I can't use the Cisco anyway because it only has a 10 Mbit WAN port and I have far more bandwidth than that. |
| 19:08.56 | Katty | Cutoff time for the list is December 15th! |
| 19:09.35 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 19:09.50 | p3nguin | I had more than 10 Mbit back then, too, and I couldn't figure out why things slowed down. It took a while, but I finally figured out that the router was the problem. |
| 19:10.07 | Baylink | p3nguin: PPS? |
| 19:10.13 | fullstop | I'm trying to get iaxmodem / hylafax up and running. I have it working, but sometimes the faxes show up all screwy. |
| 19:10.14 | p3nguin | baylink: what? |
| 19:10.22 | Baylink | VoIP is famous for trashing devices that can't handle high PPS counts. |
| 19:10.39 | fullstop | packets per second |
| 19:11.31 | drift- | p3nguin so what do you suggest i try at this point? |
| 19:11.42 | fullstop | anyway, is there anything I can find in the logs which would show an incomplete fax or the reasoning behind the screwy fax? |
| 19:12.04 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 19:12.32 | fullstop | I'm using the latest spandsp and iaxmodem and hylafax. My outgoing trunk is a digium TDM T1 card, and the digium fax channel works fine. |
| 19:12.34 | p3nguin | The problem I had with the Cisco router was that all my rtp packets from a remote device behind NAT had the device's private IP address on them, so rtp had no idea how to deal with them. |
| 19:13.24 | p3nguin | Replace the Cisco with the Linux box and all the rtp packets then show the device's public IP from outside the NAT on the remote LAN. Problem solved. |
| 19:18.31 | yonahw | [TK]D-Fender: thanks for all of your assistance earlier. I finally got it working. |
| 19:19.40 | yonahw | I still don't understand what the problem was really. I managed to get the phone to dump its app log to the server. Saw that it was failing to download the config files. Tried the credentials it had logged in the browser, which worked. Next reboot the phone could magically download the files without any other changes. I'm not sure if vsftpd is messed up or what. |
| 19:25.25 | yonahw | exit |
| 19:25.39 | yonahw | oops wrong terminal |
| 19:27.19 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 19:27.26 | wcselby | o/ |
| 19:28.43 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@adsl-dyn210.78-98-252.t-com.sk) |
| 19:31.57 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 19:32.59 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 19:39.00 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 19:43.54 | *** join/#asterisk niekie (quasselcor@CAcert/Assurer/niekie) |
| 19:50.19 | *** join/#asterisk jermudgeon (~jhaustin@216-67-61-242.static.acsalaska.net) |
| 19:55.27 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 19:57.42 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 19:58.59 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.90) |
| 20:17.48 | wcselby | wow |
| 20:17.50 | wcselby | busy in here |
| 20:27.52 | Baylink | On and off |
| 20:28.03 | drift- | yep |
| 20:28.13 | drift- | i been pulling my hair out trying to make phone work outside my internal network :( |
| 20:28.35 | ChannelZ | Think of how much money you'll save on shampoo |
| 20:28.46 | thehar | lol |
| 20:29.46 | drift- | ha |
| 20:30.44 | WIMPy | Ah, so that's the way you save money by using voip. |
| 20:31.28 | wcselby | drift- - what's not working? |
| 20:31.55 | drift- | SMC 174.48.1.164 < cablemodem > DMZ to 172.16.0.3 < asterisk/linuxrouter > internal network 172.16.1.2 < gatway from linux router |
| 20:32.08 | drift- | so smc feeds .03 and .03 feeds 1.2 |
| 20:32.12 | drift- | that feeds computers and phones |
| 20:32.15 | drift- | internal phones work fine... |
| 20:32.20 | drift- | but i have phone on DSL line |
| 20:32.30 | [TK]D-Fender | drift-: This has been very sad to wath |
| 20:32.41 | drift- | ;( |
| 20:32.45 | [TK]D-Fender | drift-: Just running around in circles spouting off this and that. |
| 20:32.49 | [TK]D-Fender | drift-: And never REALLY looking |
| 20:33.08 | wcselby | ~sipnat |
| 20:33.09 | infobot | i guess sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:33.13 | drift- | i have looked |
| 20:33.17 | wcselby | drift- - have you reveiwed these links? |
| 20:33.30 | drift- | yea... wcselby the phone can dial 10 for voicemail gets no audio |
| 20:33.38 | drift- | i can call the phone from my cell phone it receives call no audio |
| 20:33.40 | drift- | it can dial out no audio |
| 20:33.40 | [TK]D-Fender | drift-: Not one pastebin with SIP DEBUG. |
| 20:33.41 | wcselby | do you have nat setup properly? |
| 20:33.48 | [TK]D-Fender | wcselby: Of course not |
| 20:33.57 | drift- | wcselby i guess i dont , i dont know |
| 20:34.02 | wcselby | [TK]D-Fender - i'm just trying to prod him into the right direction |
| 20:34.12 | [TK]D-Fender | ~wmmfpb |
| 20:34.12 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 20:34.16 | [TK]D-Fender | ^^^^^^^^^ |
| 20:34.18 | [TK]D-Fender | There |
| 20:34.21 | drift- | lol okway pastebin what? |
| 20:34.24 | drift- | sip debug |
| 20:34.27 | wcselby | drift- - if you review the first link infobot gave above, it will show you how to setup your NAT statements in sip.conf |
| 20:34.27 | [TK]D-Fender | LOOK AT THE FUCKING CALL. |
| 20:34.42 | [TK]D-Fender | reaches for his rusty-nail upgraded ClueBat(tm) |
| 20:34.49 | wcselby | uh-oh |
| 20:35.01 | [TK]D-Fender | wcselby: 1st link is dead till I fix my routing |
| 20:35.07 | wcselby | lol oops |
| 20:36.01 | drift- | right now phone is not showing its on network for some reason |
| 20:36.14 | drift- | 220/220 (Unspecified) D N 0 Unmonitored |
| 20:36.54 | [TK]D-Fender | drift-: It has never refgistered |
| 20:37.12 | drift- | Scheduling destruction of SIP dialog '78aec0fa2a30b60865970d090ae3bc15@127.0.0.1' in 32000 ms (Method: REGISTER) [Nov 8 15:32:12] NOTICE[20691]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for 74.63.41.218 is 120 sec (Scheduling reregistration in 105 s) |
| 20:37.15 | drift- | yeah waqs |
| 20:37.21 | drift- | cause i could recieve calls and make calls and it called out |
| 20:37.50 | [TK]D-Fender | drift-: that isn't ASIP DEBUG. |
| 20:37.53 | [TK]D-Fender | drift-: TRY AGAIN |
| 20:38.38 | wcselby | drift- - please pastebin your sip.conf [general] sectino plus the peer entry for the remote phone, and also pastebin the CLI with verbosity at least 6 with a sip debug enabled for the peer, the cli from when you attempt to make a call to the phone or from the phone. |
| 20:40.20 | drift- | wcelby 1 sec |
| 20:40.24 | drift- | inet went down i believe on dsl |
| 20:40.26 | drift- | i had to restart |
| 20:40.51 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 20:41.17 | drift- | http://pastebin.com/eTt7XRzC < my sip.conf |
| 20:41.21 | nny | hmmph. What's the easiest way to tell asterisk-addons not to compile h323 support? It is throwing errors and I don't need it for this build |
| 20:41.43 | [TK]D-Fender | make menuconfig <- |
| 20:42.02 | *** join/#asterisk SparFux (~raoul@e182023183.adsl.alicedsl.de) |
| 20:42.10 | WIMPy | make menuselect |
| 20:42.35 | SparFux | Hi all. When trying to load dahdi zaphfc - it worked before without any change - I get no IRQ and dahdi_dummy is attempted to be loaded. |
| 20:42.46 | nny | [TK]D-Fender: in the asterisk-addons directory or asterisk directory? |
| 20:43.05 | SparFux | "No hardware timing source found in /proc/dahdi, loading dahdi_dummy" |
| 20:43.07 | [TK]D-Fender | addons clearly |
| 20:43.32 | nny | [TK]D-Fender: yeah nm, i need to install ncurses heh |
| 20:43.39 | WIMPy | SparFux: Any info in desg? |
| 20:43.49 | wcselby | SparFux - do you have a timing source other than dahdi_dummy? |
| 20:43.54 | nny | mm nm ncurses is installed. CentOS issue now, thanks |
| 20:44.56 | SparFux | vzaphfc: card 0: no irq! |
| 20:45.21 | SparFux | wcselby: I think the HFC pci card is a timing source. it worked without dahdi_dummy. |
| 20:45.22 | WIMPy | Looks like it can't find the hardware. What has changed? |
| 20:45.34 | drift- | this is wierd |
| 20:45.39 | drift- | now my phone wont connect to asterisk server |
| 20:45.44 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:45.46 | drift- | from outside network |
| 20:46.10 | moos3 | so can i use fax over a sip trunk ? |
| 20:46.56 | SparFux | perhaps I crapped the card. I already rebooted. I will try cold start, perhaps. |
| 20:52.04 | [TK]D-Fender | moos3: with T.38, yes |
| 20:53.59 | moos3 | [TK]D-Fender, so i sip provider as no way of preventing it do they ? |
| 20:54.52 | [TK]D-Fender | moos3: Please reword that so it looks like a real sentence... |
| 20:55.04 | Khratos | This is probably one of the most non-related question, to this channel, but I had no option but to try to get an answer: what is a 'closer' campaign? (on vicidial) |
| 20:55.43 | Khratos | My native language is not english, and no information seems to take me to a real understanding of that term (in that context) |
| 20:55.49 | *** join/#asterisk siouxes (~sfreddio@88-149-210-58.dynamic.ngi.it) |
| 20:56.25 | siouxes | hi all |
| 20:57.15 | atan | What's the command to view loaded modules? I want to see if chan-sccp is in there |
| 20:58.32 | [TK]D-Fender | Khratos: Vicidial is not supported in this channel. |
| 20:59.00 | atan | [TK]D-Fender, my goodness you're still here giving out advice. |
| 20:59.07 | atan | [TK]D-Fender, my goodness you're dedicated :P |
| 20:59.20 | yonahw | atan: module show for all or module show like keyword for specific |
| 20:59.35 | atan | can I grep? |
| 20:59.42 | atan | like module show | grep sccp |
| 20:59.51 | atan | Or anything similar? If I don't know the exact name? |
| 20:59.58 | p3nguin | Only if you use a shell and not the asterisk CLI. |
| 21:00.21 | atan | My lord I wish they would just put everything in alphabetical order. :( |
| 21:00.58 | yonahw | atan: like will look for the string anywhere in the name |
| 21:01.23 | siouxes | hi guys, i have a strange problem regarding dtmf and fax detection... can i post here? |
| 21:01.25 | atan | yonahw, WIN! |
| 21:01.25 | atan | Thanks |
| 21:01.33 | yonahw | np |
| 21:01.40 | *** join/#asterisk SparFux (~raoul@e182023183.adsl.alicedsl.de) |
| 21:02.23 | siouxes | i try... |
| 21:02.49 | siouxes | i have an asterisk box with 2 E1... one is connected to public carrier... the other one is connected to customer pbx |
| 21:02.59 | atan | A tutorial I am reading says ..."chan_sccp is a newer version of the skinny support for asterisk, it replaces the original chan_skinny" but, uh. Is chan_sccp superior in some way? Or is the default one included in Asterisk just fine? |
| 21:02.59 | SparFux | Re. dahdi still says the hfc pci card had no irq, but lspci -vvv shows "Interrupt: pin A routed to IRQ 19" |
| 21:03.00 | siouxes | so the asterisk box is in the middle |
| 21:03.44 | WIMPy | SparFux: What changed since it worked? |
| 21:04.19 | Khratos | I was not asking for 'support', but for a definition of a word in certain context |
| 21:05.24 | SparFux | WIMPy: I just dropped two #define DEBUG statements in two files to get no more debugging output. |
| 21:05.31 | siouxes | in a first time i have problem with dtmf detection, so i have enabled hardware dtmf detection on the PRI card and dtmf goes working well... in this situation if i make a fax call from customer pbx (that transit in asterisk box and directly exit to public carrier) the PRI card send a 'f' digit that will be routed (as a dtmf) to the public carrier... |
| 21:06.14 | siouxes | ...public carrier, that not recognize the 'f' dtmf, close the call. |
| 21:06.21 | SparFux | WIMPy: trying with the #define DEBUG again, I know exactly where I'd placed them. |
| 21:06.39 | WIMPy | siouxes: There is not f digit. |
| 21:07.41 | WIMPy | siouxes: Do you have fax detection enabled? |
| 21:07.57 | SparFux | WIMPy: even with DEBUG on again, it's wrecked. |
| 21:08.10 | drift- | ok i see the phone trying to connect |
| 21:08.19 | drift- | 16:02:38.881706 IP adsl-2-253-76.mia.bellsouth.net.49154 > 172.16.0.3.5070: UDP, length 507 16:02:42.881354 IP adsl-2-253-76.mia.bellsouth.net.49154 > 172.16.0.3.5070: UDP, length 507 |
| 21:08.22 | drift- | my tcpdump |
| 21:08.27 | drift- | but its not letting it connect |
| 21:08.44 | siouxes | WINPy: i have disable faxdetection into chan_dahdi |
| 21:08.46 | drift- | my external phone on dsl trying to connect to internal |
| 21:09.01 | WIMPy | SparFux: Restore backup? |
| 21:09.20 | SparFux | WIMPy: I have done that. still get no irq. The card must have broke right away :-( |
| 21:09.22 | WIMPy | SparFux: Or just try to load a kernel driver and see if that finds the card. |
| 21:09.58 | siouxes | WINPy: but the dahdi driver still sending |
| 21:10.03 | siouxes | WINPy: DEBUG[18902] chan_dahdi.c: Detected digit 'f' |
| 21:10.13 | SparFux | WIMPy: ah, good idea. |
| 21:10.23 | *** join/#asterisk Bartok (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 21:10.48 | SparFux | WIMPy: yes, that works :-) puh! |
| 21:10.56 | WIMPy | siouxes: Haven't tried it, but I guess that it what fax detection spews out. So I guess that's active. |
| 21:11.40 | Bartok | Hey folks - still have my one-way voice issue - |
| 21:11.54 | Bartok | Asterisk 1.4.31 built by root @ xxxxxx on a x86_64 running Linux on 2010-06-10 14:32:34 UTC |
| 21:12.12 | SparFux | what have I changed???? |
| 21:12.31 | Bartok | this is what is weird - the Echo test seems to work well - almost no lag (well, I am sure if I ran wireshark I would see) |
| 21:12.52 | siouxes | WINPy: if i try 'dahdi show channel 1' , i have faxhandled: no |
| 21:12.56 | Bartok | dialing in, no sound on the receiving end |
| 21:13.01 | WIMPy | SparFux: None of us know. |
| 21:13.16 | Bartok | dialing out- appears to work fine - no issue |
| 21:13.24 | Bartok | any words of wisdom |
| 21:13.28 | Bartok | ? |
| 21:13.31 | SparFux | WIMPy: me IDIOT! |
| 21:13.39 | SparFux | WIMPy: I changed the kernel source :-( |
| 21:13.47 | siouxes | WIMPy: but the problem is not really related to faxdetection... the real problem is that the 'f' digit is sent over public connection |
| 21:13.48 | *** join/#asterisk DennisG (~DennisG@541E88D0.cm-5-7c.dynamic.ziggo.nl) |
| 21:13.49 | SparFux | kicks his own ass. |
| 21:13.59 | WIMPy | siouxes: Maybe you need to disable it in the driver module? |
| 21:14.10 | siouxes | i have no option to do this... |
| 21:14.15 | WIMPy | siouxes: s.a. |
| 21:14.38 | siouxes | excuse me, what is 's.a.' ? |
| 21:14.46 | WIMPy | see above |
| 21:14.50 | siouxes | ops |
| 21:15.13 | WIMPy | siouxes: There is no f digit. |
| 21:16.20 | wcselby | drift- - do you have an iptables rule for sip traffic on your server? |
| 21:16.48 | siouxes | WIMPy: i know that an f digit is not valid. this IS the issue. Asterisk consider it as valid and sent it as information packet over outbound isdn connection... |
| 21:17.12 | WIMPy | siouxes: How would it do so? |
| 21:17.36 | siouxes | i have the pri debug showing information packet with digit 'f' (66 hexdecimal) |
| 21:17.41 | WIMPy | I'm pretty sure that's the fax detections way to tell it found a fax CNG. |
| 21:17.51 | WIMPy | Urgs |
| 21:18.03 | WIMPy | I'd like to see that. |
| 21:18.09 | siouxes | if you want i can paste it into this channel |
| 21:18.32 | WIMPy | Surely sounds interesting. |
| 21:18.33 | siouxes | ok |
| 21:18.50 | siouxes | 1 of 3:Message type: INFORMATION (123) |
| 21:18.59 | siouxes | 2 of 3:[70 02 80 66] |
| 21:19.08 | siouxes | 3 of 3:Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'f' ] |
| 21:19.23 | siouxes | this is my issue! very very strange |
| 21:19.42 | WIMPy | Oh. That's even more interesting. |
| 21:19.42 | siouxes | obviusly public carrier didn't understund this and hangup the call |
| 21:20.03 | WIMPy | Are you dialling using 'information'? |
| 21:20.07 | siouxes | yes |
| 21:20.28 | WIMPy | Yes, I can see that they won't like it. |
| 21:20.41 | WIMPy | That looks like a bug. |
| 21:20.52 | siouxes | yes i think too |
| 21:21.22 | siouxes | do you think that i can do any workaround? |
| 21:21.41 | WIMPy | But how fast is that happening? I'd expect that you should have received a proceeding bedore any fax would be detected. |
| 21:21.56 | WIMPy | Yes, you need to disable fax detection. |
| 21:21.59 | *** part/#asterisk SparFux (~raoul@e182023183.adsl.alicedsl.de) |
| 21:22.13 | siouxes | yes... i am a newbe in irc... and i don't won't to paste all my trace... |
| 21:22.23 | wcselby | siouxes - use pastebin |
| 21:22.24 | wcselby | ~pb |
| 21:22.24 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 21:22.41 | siouxes | by the way, i have a normal cal setup.... setup, setup ack, call proceding... |
| 21:22.52 | siouxes | ah ok |
| 21:23.04 | citywok | Is anybody interested in a phonebook for the Aastra phones that reads from the users outlook contact list? |
| 21:23.25 | WIMPy | Hmm. After a proceeding no more digits should be sent anyway. |
| 21:23.53 | WIMPy | Sounds like two bugs at the moment. |
| 21:26.08 | siouxes | excuse me again WIMPy, i am at home and i saw just now that i haven't the complete trace, just the information packet (that i have used to google around)... tomorrow morning i return to customer site and i can get the complete debug file.... |
| 21:26.41 | WIMPy | bad luck. |
| 21:27.05 | siouxes | there is a way, in your mind, to just filter out wich dtmf digit is allowed to be sent ? |
| 21:27.13 | WIMPy | But you might have to take a look at the module parameters if you have hardware DSP. |
| 21:27.44 | siouxes | i am using the wct4xxp |
| 21:27.56 | WIMPy | No, I didn't dig that deep, yet. |
| 21:28.15 | siouxes | that has no info regarding fax, only regarding dtmf hardware detection... |
| 21:29.01 | siouxes | if i set it to off, i haven't the 'f' issue but dtmf detection goes very bad... no detection or double/triple detection... |
| 21:29.03 | WIMPy | Maybe you can only disable both. If in doubt ask Digiums support. |
| 21:29.34 | siouxes | to ask to digium, i have to pay? |
| 21:30.05 | WIMPy | If you bought their hardware, there should be support. |
| 21:30.13 | WIMPy | Take a look at their website. |
| 21:31.09 | russellb | yep |
| 21:31.13 | siouxes | i'll take a look... for now thank's for all... tomorrow i'll came back with complete debug file... |
| 21:31.14 | russellb | http://www.digium.com/en/supportcenter/ |
| 21:31.26 | siouxes | ok thank you guys |
| 21:36.36 | drift- | <PROTECTED> |
| 21:36.53 | p3nguin | It means: turn off comfort noise. |
| 21:37.04 | drift- | where on phone? |
| 21:37.10 | p3nguin | That's what it says. |
| 21:37.18 | drfreeze | I have two servers using BLF. Both are identically setup with polycom phones |
| 21:37.30 | drift- | btw p3nguin the 220 extension doesnt show in "sip show peers" :( |
| 21:37.32 | drift- | the dsl phone |
| 21:37.38 | p3nguin | Silense suppression, voice activity detection... there's a setting for it somewhere. |
| 21:37.50 | p3nguin | If it doesn't show up, then it isn't registered. |
| 21:37.53 | drfreeze | As of last Friday, BLF was working on both systems, today, one of the system reports that all the phones are idle all the time, even when in use |
| 21:37.58 | *** join/#asterisk segv` (~segv@eon.segv.net) |
| 21:37.59 | drift- | yeah how can i figure out why? |
| 21:38.01 | drift- | or how? |
| 21:38.09 | drfreeze | I resetarted asterisk, and the phones, but get the same result. |
| 21:38.10 | p3nguin | How about the sip debug? |
| 21:38.15 | drift- | i dont see anything |
| 21:38.28 | drift- | thats relivant |
| 21:38.39 | p3nguin | Then the phone either is not trying to register or you have blocked it with some type of networking situation. |
| 21:38.56 | WIMPy | drfreeze: Ask the phone. |
| 21:39.13 | p3nguin | During testing, I like to set my register time on devices very low. |
| 21:39.27 | drift- | how do you do that/ |
| 21:39.32 | drift- | this is a polycom 501 sip phone |
| 21:39.34 | p3nguin | If it is high, you have to wait a long time in between attempts. |
| 21:39.57 | drift- | Scheduling destruction of SIP dialog '054a9e8b6d05791f7ba1268177bd3c54@127.0.0.1' in 32000 ms (Method: REGISTER) [Nov 8 16:34:38] NOTICE[21250]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for 74.63.41.218 is 120 sec (Scheduling reregistration in 105 s) |
| 21:40.03 | drfreeze | The watchers list is correct |
| 21:40.38 | drift- | p3nguin i know its trying... cause i have tcpdump on the ip address |
| 21:40.53 | p3nguin | Okay, then you've managed to block it. |
| 21:41.15 | drift- | 16:36:24.684318 IP adsl-2-253-76.mia.bellsouth.net.49154 > 172.16.0.3.5070: UDP, length 508 16:36:25.199709 IP adsl-2-253-76.mia.bellsouth.net.49154 > 172.16.0.3.5070: UDP, length 508 16:36:25.699221 IP adsl-2-253-76.mia.bellsouth.net.49154 > 172.16.0.3.5070: UDP, length 508 16:36:26.699536 IP adsl-2-253-76.mia.bellsouth.net.49154 > 172.16.0.3.5070: UDP, length 508 |
| 21:41.18 | drift- | 4 attempts at a time |
| 21:41.23 | JoseBravo | Is possible change the codec of active SIP channel? |
| 21:42.12 | drfreeze | WIMPy: phone reports: CallingPres : Presentation Allowed, Not Screened |
| 21:42.50 | citywok | russellb: i ended up finding a couple issues with my exchange setup, unfortunately having fixed them * still segfaults the same way. lol |
| 21:43.03 | russellb | :-/ |
| 21:43.15 | russellb | doc/backtrace.txt |
| 21:43.57 | Qwell | russellb: *thwap* |
| 21:43.59 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace :D |
| 21:44.10 | russellb | yes. taht. |
| 21:44.15 | russellb | s/taht/that |
| 21:44.19 | *** join/#asterisk Preytell (~jerry.win@68-188-27-90.static.stls.mo.charter.com) |
| 21:44.22 | russellb | s/// fail. |
| 21:44.24 | citywok | lolol. Yes, it's already all on the issue tracker :P |
| 21:44.41 | russellb | oh? then why haven't you fixed it yet |
| 21:44.41 | Preytell | can polycom transfer keys be mapped to do "##" transfer? |
| 21:44.56 | citywok | b/c it's over my head / C programming abilities |
| 21:45.29 | citywok | i think it may be the neon library that asterisk is relying on, but i'm not really sure. and * shouldn't seg, it should handle it gracefully. |
| 21:45.54 | russellb | issue #? |
| 21:46.01 | russellb | and i was just kidding |
| 21:46.11 | citywok | i kinda figured, lol 18220 |
| 21:47.03 | russellb | oh, the calendar thing |
| 21:47.07 | russellb | i thought you were talking about SIP |
| 21:47.14 | citywok | oh, no hah |
| 21:47.23 | citywok | i don't have any issues with SIP anymore, no seg's therel ol |
| 21:47.48 | russellb | pitel on there wrote it, hopefully he can look at it ... |
| 21:49.51 | citywok | yea, hopefully. if not i am able to replicate the features in PHP and could ghetto-fix it for the time being. |
| 21:50.28 | citywok | the res_calendar thing needs a couple new features too, but i can't code them so i can't submit it on the tracker lol |
| 21:50.47 | russellb | heh |
| 21:50.55 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 21:51.05 | russellb | you could email twilson@digium.com (and CC russell@digium.com) with your feature ideas |
| 21:51.20 | russellb | i'll throw them on the heap of ideas |
| 21:51.35 | citywok | kk, they're pretty simple |
| 21:51.56 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:54.52 | p3nguin | drift-: If you don't see it in sip debug, then it isn't reaching Asterisk. If you see it in a tcpdump on the host, then it is at least reaching the host. Find out what you changed to prevent it from reaching Asterisk. |
| 21:57.28 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:00.01 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:01.40 | drift- | p3nguin heh i'm taking break on this asterisk stuff i'll be back in office tomorrow ;( |
| 22:01.46 | drift- | got headache |
| 22:01.52 | drift- | thanks so much for your help |
| 22:02.10 | drift- | you been very kind :) |
| 22:02.24 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 22:02.24 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 22:02.35 | carrar | There are no breaks |
| 22:02.40 | carrar | GIT R DONE |
| 22:08.36 | citywok | alright russell, i sent it to you guys. Looking at the code in res_cal_ews it looks like it'd be simple to add the XML to the requests, the question being how to implement the options in calendar.conf -- of course, this assumes it works at all and * doesn't segfault lol. |
| 22:09.36 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 22:13.46 | Katty | if anyone is interested in participating in the asterisk christmas card exchange, please /query me for details! Cut off for sign-ups is December 15th! |
| 22:15.32 | p3nguin | suggests a timer |
| 22:16.11 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 22:18.42 | atan | Okay I'm having almost no luck getting my little dd-wrt router to work as an OpenVPN client. Pfft. |
| 22:19.00 | atan | Are there any other methods one could use to somewhat securely use sccp? |
| 22:21.46 | *** join/#asterisk [canniballllera] (~cannibale@200-138-252-150.fnsce703.dsl.brasiltelecom.net.br) |
| 22:39.36 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:39.37 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 22:39.44 | IsUp | hello everyone |
| 22:40.12 | IsUp | i have a problem with Asterisk startup. When i start asterisk, it takes long time to initialize SIP |
| 22:40.22 | *** join/#asterisk CattyRayheart (~quassel@pool-173-66-190-190.washdc.fios.verizon.net) |
| 22:40.24 | IsUp | meanwhile, i am unable to do run any 'sip show peers' or any SIP commands |
| 22:40.37 | IsUp | i have 3 outbound providers, 1 sip user for X-Lite |
| 22:40.40 | IsUp | is that normal? |
| 22:40.57 | IsUp | same as 'sip reload', when i do 'sip reload' it takes about 2 mins to get SIP up |
| 22:47.25 | *** part/#asterisk jplank (~G_Bove@208-104-67-26.dyn.fttp.comporium.net) |
| 22:53.38 | IsUp | anybody alive? :) |
| 22:54.17 | *** join/#asterisk sbszulu (~dundubala@41.14.136.204) |
| 22:54.18 | WIMPy | DNS issues? |
| 22:54.27 | IsUp | well, its just IP based |
| 22:54.30 | IsUp | never doing a host lookup |
| 22:54.38 | IsUp | i suspect of "qualify" |
| 22:54.47 | IsUp | it makes a lag or something as i believe |
| 22:54.54 | WIMPy | That could be an expanation |
| 22:55.01 | IsUp | is that normal? |
| 22:55.10 | WIMPy | Can it find it's own identity? |
| 22:55.14 | p3nguin | its own |
| 22:55.27 | WIMPy | yes |
| 22:55.37 | p3nguin | "it is own" wouldn't make any sense. |
| 22:55.57 | IsUp | well basicly, i am running Asterisk on VMWare, i have X-Lite and i have 3 providers for outbound calls |
| 22:55.59 | WIMPy | all your base are belong to us! |
| 22:56.06 | IsUp | so its just for personal use i mean |
| 22:56.11 | IsUp | i am not sure best configuration in sip.conf |
| 22:56.20 | IsUp | can anyone help me if u put to pastebin? |
| 22:57.20 | *** join/#asterisk DelphiWorld (~VoIpMan@41.200.3.11) |
| 22:57.22 | DelphiWorld | hi |
| 22:57.29 | DelphiWorld | how to do early media in asterisk? |
| 22:57.30 | WIMPy | I'm not sure what Asterisk, or chan_sip exactely do, but it might try to enumerate your interfaces and their host names. |
| 22:57.53 | IsUp | i am using OpenDNS in Linux |
| 22:57.59 | WIMPy | So if that's not possible, that might be a common issue. |
| 22:58.19 | IsUp | DelphiWorld: try Progress and then Playback |
| 22:58.29 | IsUp | which channel technology you are using? |
| 22:58.32 | DelphiWorld | iscario: thank you |
| 22:58.47 | DelphiWorld | iscario: what about if i want to let my client to pass there early media? |
| 22:58.52 | p3nguin | delphiworld: Use Playback() and the noanswer option for it. |
| 22:59.02 | *** join/#asterisk rrb3942 (~rbullock@67.242.215.62) |
| 22:59.02 | *** part/#asterisk segv` (~segv@eon.segv.net) |
| 22:59.10 | DelphiWorld | p3nguin: thanks and IsUp also |
| 22:59.20 | iscario | DelphiWorld: you're welcom :D but that was not me ^^ |
| 22:59.29 | IsUp | DelphiWorld: try pass to call without answering it, i think |
| 22:59.44 | DelphiWorld | IsUp: ;) |
| 22:59.47 | IsUp | there must be a "rogressinband" option for SIP |
| 22:59.54 | IsUp | *progressinband i mean |
| 22:59.54 | DelphiWorld | iscario: lol you are lucky always thinked hehehe |
| 23:00.32 | IsUp | which channel technology you are using? |
| 23:02.15 | IsUp | and folks, does "qualify" necessary for outbound calls? |
| 23:03.13 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 23:05.09 | DelphiWorld | bye guys |
| 23:05.14 | DelphiWorld | thank for all that helped me |
| 23:05.15 | *** part/#asterisk DelphiWorld (~VoIpMan@41.200.3.11) |
| 23:05.16 | IsUp | goodbye |
| 23:06.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:11.14 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 23:15.56 | *** join/#asterisk fofware (~Fabian@host127.200-82-50.telecom.net.ar) |
| 23:16.31 | *** join/#asterisk guilhermebr (~Guilherme@189.63.33.163) |
| 23:22.56 | IsUp | well |
| 23:22.59 | IsUp | my problem is solved |
| 23:23.05 | IsUp | by switching to Comodo DNS :) |
| 23:23.10 | IsUp | thanks to everyone |
| 23:24.04 | p3nguin | (1702.14) <IsUp> and folks, does "qualify" necessary for outbound calls? <-- no. qualify is used for NAT keepalive and to remove portable devices when they have gone "away" |
| 23:24.24 | IsUp | understand |
| 23:24.42 | IsUp | and whatis "insecure" |
| 23:24.48 | IsUp | does it necessary for every peer? |
| 23:25.00 | IsUp | and one last question, whats the difference between "friend" and "peer" :) |
| 23:25.50 | p3nguin | I can't think how to describe insecure, but friend is both a peer and a user. |
| 23:25.52 | *** part/#asterisk [canniballllera] (~cannibale@200-138-252-150.fnsce703.dsl.brasiltelecom.net.br) |
| 23:26.24 | *** join/#asterisk tinkerghost (~ghost@pool-72-70-245-49.spfdma.east.verizon.net) |
| 23:26.26 | IsUp | okay thank you |
| 23:26.32 | IsUp | i'll take a look to voip-info for insecure |
| 23:27.55 | tinkerghost | I am having an intermittant issue with calls being rejected - 2 calls for the same phone number will have 1 routed correctly & the other rejected as not being able to find the extension |
| 23:29.53 | IsUp | can you explain more? |
| 23:31.48 | tinkerghost | I have a SIP connection to my DID provider. The extension routes it directly to a single context - no included contexts. |
| 23:31.56 | [TK]D-Fender | ~pb |
| 23:31.57 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:31.58 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 23:32.04 | [TK]D-Fender | Explain less, show more. |
| 23:32.11 | IsUp | exactly. |
| 23:32.20 | IsUp | ^^ |
| 23:34.02 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 23:35.29 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 23:39.48 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 23:42.18 | tinkerghost | sorry had cop/paste issues w/ remote server. http://pastebin.com/0cYCa1Ug |
| 23:45.39 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 23:47.27 | tinkerghost | I just updated it with the console report on 2 calls - 1 succeeded, the other failed |
| 23:48.21 | IsUp | tinkerghost: i didnt understand your problem actually |
| 23:48.32 | IsUp | u can PM me if u want |
| 23:48.37 | IsUp | we can try |
| 23:48.52 | [TK]D-Fender | tinkerghost: useless... pastebin the FAILED CALL |
| 23:56.41 | *** join/#asterisk sbszulu (~dundubala@41.16.127.251) |
| 23:57.56 | *** join/#asterisk genxweb (~genxweb@c-174-59-109-188.hsd1.pa.comcast.net) |
| 23:58.13 | genxweb | is there anyone here that can elp me with phpagi issue |
| 23:58.27 | IsUp | whats up? |
| 23:58.36 | genxweb | let me put it in a pastebin to show you |
| 23:58.39 | IsUp | sure |
