| 00:02.50 | singler | my issue is that I am filling my first bug, and I want to do it correctly :) |
| 00:14.00 | Precognist | p3nguin: can you tell me why my iphone cant make calls to the computers, but the computers can call the iphone? |
| 00:14.24 | Precognist | p3nguin: i mean, it can call, but there is not audio. |
| 00:14.26 | p3nguin | Not without more evidence, no. |
| 00:14.46 | Precognist | know of anyway to troubleshoot? |
| 00:14.47 | p3nguin | One-way audio is most typically caused by NAT. |
| 00:14.58 | Precognist | thats router 2? |
| 00:15.15 | Precognist | ahhh |
| 00:15.21 | p3nguin | Do you have a NAT between Asterisk and the iPhone? |
| 00:15.48 | Precognist | a router, thats it |
| 00:16.04 | p3nguin | So the iPhone is outside the NAT and Asterisk is inside? |
| 00:16.35 | Precognist | no, all inside on the network. via wifi |
| 00:17.13 | p3nguin | I'd have to see a sip debug and maybe even an rtp debug of a call with one-way audio to make a guess. |
| 00:18.05 | Precognist | ok, wow. how do i do that? |
| 00:18.12 | Precognist | in asterisk or the sip |
| 00:18.18 | Precognist | i know how on the sip client |
| 00:18.28 | p3nguin | in asterisk |
| 00:18.39 | p3nguin | Asterisk CLI, sip set debug on |
| 00:18.42 | Precognist | Shi...um.. shins. |
| 00:18.51 | Precognist | ok |
| 00:19.12 | Precognist | enabled |
| 00:19.29 | p3nguin | If the peer is working correctly, you might be able to filter by peer name to reduce some of the debug stuff. sip set debug peer <phone's peer name> |
| 00:20.27 | p3nguin | With debug enabled, make a call which has one-way audio. |
| 00:20.39 | Precognist | what if its set to friend not peer |
| 00:20.53 | p3nguin | Capture the entire call from the time you dial the number, to the time you have no audio, to the time you hang up. |
| 00:21.09 | p3nguin | A peer's type being set to friend isn't relevant at this point. |
| 00:21.15 | Precognist | ahhh |
| 00:22.45 | p3nguin | The peer's type mainly dictates how peer matching is performed. |
| 00:24.12 | Precognist | ok, enabled |
| 00:24.22 | Precognist | (had to re-connect sip on phone) |
| 00:25.57 | singler | at issue creation time can files be uploaded? |
| 00:27.22 | Precognist | ok, i think i did it. http://pastebin.com/nmY7LaKi |
| 00:34.05 | p3nguin | Do you have any firewall rules loaded on the Ubunturd system? |
| 00:34.47 | p3nguin | iptables -L |
| 00:35.29 | Precognist | Chain INPUT (policy ACCEPT) |
| 00:35.30 | Precognist | target prot opt source destination |
| 00:35.30 | Precognist | Chain FORWARD (policy ACCEPT) |
| 00:35.30 | Precognist | target prot opt source destination |
| 00:35.30 | Precognist | Chain OUTPUT (policy ACCEPT) |
| 00:35.31 | Precognist | target prot opt source destination |
| 00:35.33 | Precognist | sorry |
| 00:38.24 | ChannelZ | everyone seems to be on the same LAN |
| 00:39.33 | *** part/#asterisk Precognist (~yeshualoo@adsl-75-15-226-185.dsl.bkfd14.sbcglobal.net) |
| 00:41.34 | p3nguin | Yes, they do, but the INPUT chain blocks or allows things to the host, regardless of their proximity. I was hoping to see something stupid in the firewall blocking those RTP ports, or other goofiness. |
| 00:42.16 | *** join/#asterisk precognist (~precognis@adsl-75-15-226-185.dsl.bkfd14.sbcglobal.net) |
| 00:42.21 | p3nguin | Since the debug looks reasonable and there are no blocking firewall rules, I'd probably have to look at the client next. |
| 00:42.52 | precognist | I left for a second. Did I miss anything? |
| 00:42.59 | p3nguin | nothing important. |
| 00:43.06 | ChannelZ | Your softphones are broken |
| 00:43.20 | ChannelZ | :) or firewalls running on those machines are mucking up the traffic possibly |
| 00:43.36 | p3nguin | I thought the softphones were working but the iPhone wasn't. |
| 00:44.15 | ChannelZ | He said from iPhoney to his Mac |
| 00:44.19 | ChannelZ | I think |
| 00:44.22 | precognist | <PROTECTED> |
| 00:44.40 | p3nguin | I couldn't follow the long story, so it's possible that you're right. |
| 00:45.22 | precognist | I think it's the connection from the iPhone |
| 00:45.23 | ChannelZ | or actually it's only one way? Mac-calls-iPhone works but iPhone-calls-Mac doesn't |
| 00:46.40 | *** join/#asterisk agnogenic (agnogenic@c-67-176-218-28.hsd1.il.comcast.net) |
| 00:47.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:47.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:50.32 | *** join/#asterisk precognist_ (~precognis@166.205.138.211) |
| 00:50.45 | ChannelZ | You seem to have connectivity problems in general |
| 00:57.02 | precognist | No. Switched 2 iPhone |
| 01:00.35 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 01:00.44 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:05.20 | *** join/#asterisk smeet2002 (~smeet2002@173.248.230.237) |
| 01:05.56 | smeet2002 | hi everybody |
| 01:06.10 | smeet2002 | any alive persons here? |
| 01:06.10 | WIMPy | lo single one |
| 01:06.27 | WIMPy | ~ask |
| 01:06.27 | 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. |
| 01:07.25 | smeet2002 | I am trying to troubleshoot connection with my provider...it seems it doesn't receive my packets...or may be I can't receive his.. |
| 01:07.46 | smeet2002 | it's always in UNRECHABLE state |
| 01:07.54 | smeet2002 | but I can ping it well |
| 01:08.19 | smeet2002 | who can suggest anything where to dig? |
| 01:08.20 | WIMPy | Or they just don't like OPTIONS packets. |
| 01:08.27 | smeet2002 | I turned on sip debugging |
| 01:08.46 | smeet2002 | by the way I have a lot of OPTIONS packet written in my log |
| 01:09.07 | smeet2002 | re-transmitting all the time...even if I don't make calls..WTF? |
| 01:09.08 | singler | smeet2002: try using qualify=no |
| 01:09.19 | smeet2002 | btu ti was working fine before |
| 01:09.25 | smeet2002 | but it |
| 01:09.59 | smeet2002 | I can receive calls from it but I can't place calls |
| 01:10.33 | WIMPy | You shot yourself in the foot. |
| 01:10.45 | smeet2002 | I am using configuration that they provided to me... |
| 01:10.57 | smeet2002 | it voicenetwork.ca |
| 01:11.02 | WIMPy | Do as singler said. |
| 01:11.51 | smeet2002 | Ok...they want "qualify=yes" for incoming peer..why I need to turn it off? |
| 01:12.10 | smeet2002 | I am little bit retarded...sorry for asking too much questions |
| 01:12.26 | WIMPy | Because it obviousely doesn't work. |
| 01:12.39 | *** join/#asterisk BuenGenio (~BuenGenio@059148208218.ctinets.com) |
| 01:12.39 | WIMPy | And that's why you can't call out. |
| 01:13.03 | smeet2002 | interesting...where is the logic? ...I will try it right now anyway |
| 01:13.24 | smeet2002 | but it worked before...worked fine... |
| 01:13.58 | WIMPy | The logic is not even to try calling a peer that is known to be unreachable. |
| 01:16.09 | smeet2002 | that makes sense WIMPy... |
| 01:16.39 | smeet2002 | I put it "unmonitored" state and it doesn't work :-(( |
| 01:17.22 | *** join/#asterisk james_zhu (~Administr@183.16.215.92) |
| 01:17.55 | smeet2002 | could anybody give me any advice how to dig it? I tried tcpdump, I can't see any packets coming from their side... |
| 01:18.11 | smeet2002 | but I can ping them...that's weird... |
| 01:24.10 | p3nguin | What's so weird about that? ping is ICMP, VoIP isn't. |
| 01:31.24 | smeet2002 | yea..you're right p3nguin...I told you, I am slightly retarded...nevertheless...what can I do apart of turning sip debug on and writing everything into the log file? |
| 01:32.03 | p3nguin | What ITSP are you using? |
| 01:34.17 | smeet2002 | voicenetwork.ca |
| 01:34.30 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:34.30 | p3nguin | Are you using a register statement? |
| 01:35.37 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 01:36.44 | smeet2002 | apparently no...but they don't want it...they give the whole config |
| 01:36.47 | smeet2002 | they want |
| 01:37.17 | p3nguin | So you have configured IP auth in the user portal? |
| 01:37.56 | p3nguin | Do your have a static IP address for your Asterisk system? |
| 01:38.04 | smeet2002 | yes I have static ip |
| 01:38.26 | smeet2002 | and I have this "secret=.." |
| 01:38.27 | p3nguin | That was part B. Don't skip part A. |
| 01:39.01 | smeet2002 | sorry...I thought "secret=." is an authorization |
| 01:39.13 | p3nguin | secret is the secret, aka password. |
| 01:39.36 | p3nguin | But in the user portal on the ITSP, you either have to use IP auth or SIP registrations. |
| 01:40.13 | smeet2002 | how can I see it on my side what I am using? |
| 01:40.37 | p3nguin | I don't use voicenetwork.ca, so I don't know where it is in the user portal. |
| 01:40.54 | smeet2002 | isn't this "secret=.." refer to sip authorization? |
| 01:41.10 | p3nguin | It's related to it, but it is not what I am asking you about. |
| 01:41.46 | smeet2002 | you mean it should be smth in web interface to input my ip ? |
| 01:41.54 | smeet2002 | and authjorize it? |
| 01:42.06 | p3nguin | I don't know what smith is, but I'm talking about in the user portal of the ITSP. |
| 01:42.42 | smeet2002 | yes I have portal acces...with all statistic and all this crap |
| 01:43.17 | p3nguin | Okay, now find out if you are supposed to use IP auth or SIP registration. |
| 01:43.41 | p3nguin | If you don't know, I'm going to assume it is SIP registration. And if so, that's probably why things aren't working for you. |
| 01:44.35 | smeet2002 | hmm...what do you mean? wrong password? |
| 01:45.35 | p3nguin | Did I say ANYTHING about a friggin' password? |
| 01:46.15 | smeet2002 | probably not... |
| 01:46.28 | p3nguin | Just read the words I'm typing. |
| 01:46.37 | smeet2002 | I am trying |
| 01:46.41 | smeet2002 | doing my best |
| 01:46.45 | p3nguin | Stop trying to guess at an alternate meaning. I mean what I'm saying. |
| 01:47.40 | smeet2002 | but what really freaks me, it all worked before... |
| 01:47.46 | smeet2002 | now it doesn't:-(( |
| 01:48.01 | p3nguin | I guess you'd better figure out what you changed, then. |
| 01:49.43 | *** join/#asterisk agnogenic (agnogenic@c-67-176-218-28.hsd1.il.comcast.net) |
| 01:49.48 | agnogenic | I'm looking for provider like sipgate(They aren't accepting new registrations atm) who has free numbers for inbound calling. Any recommendations? |
| 01:49.56 | smeet2002 | I set up firewall...but now if even I turn it off, it still doesn't work..so I asume it's not the issue... |
| 01:50.49 | p3nguin | agnogenic: ipkall, ipcomms |
| 01:51.18 | smeet2002 | OK..thanks anyway...I will contact them, send them my logs and they will probably find out |
| 01:52.01 | p3nguin | It could be the firewall. |
| 01:59.58 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 02:02.35 | agnogenic | ty p3nguin |
| 02:04.29 | smeet2002 | what could prevent provider to answer my packets? whether he can't receive mine, whether I can't receive his...is there any possibility that my Internet provider cuts Voip packets? they have their own Voip telephones...may be they pushing people to use theirs? |
| 02:05.03 | p3nguin | It's very possible they could block standard VoIP ports. |
| 02:05.32 | p3nguin | Many providers offer non-standard ports to bypass those blockages. |
| 02:08.07 | smeet2002 | Probably the best way is just to ask VOip provider to see if they get my packets...and then we will see |
| 02:08.31 | *** join/#asterisk Precognist (~yeshualoo@adsl-75-15-226-185.dsl.bkfd14.sbcglobal.net) |
| 02:08.49 | *** join/#asterisk moy_ (~moy@69.157.46.221) |
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| 02:18.30 | *** join/#asterisk diijiib (~nobodysho@bas10-kitchener06-1279411209.dsl.bell.ca) |
| 02:18.54 | diijiib | anybody in here thats free to lend a hand? |
| 02:19.05 | p3nguin | ~ask |
| 02:19.05 | 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. |
| 02:20.42 | diijiib | ok has my openwrt 10.03 asterisk16 box setup fresh today. had calls comming from voip provider, and able to call internal extensions. |
| 02:20.51 | diijiib | i started playing around with automixmon |
| 02:21.08 | diijiib | and specifically a symbolic link bug in 1.6 |
| 02:21.20 | diijiib | /var/lib/asterisk/sound |
| 02:21.22 | diijiib | was missing |
| 02:21.58 | diijiib | anyways... ill try and make it short.. here is my dialplan debug |
| 02:21.59 | diijiib | http://pastebin.com/un8Sy5Ke |
| 02:22.12 | diijiib | and these are my sip & extension |
| 02:22.12 | diijiib | http://pastebin.com/QPYxhrmA |
| 02:22.15 | diijiib | what do? |
| 02:22.52 | p3nguin | |
| 02:23.50 | diijiib | the link was made from /usr/lib/asterisk to /var/lib/asterisk |
| 02:26.03 | diijiib | am i in the wrong channel? |
| 02:26.54 | WIMPy | Did I miss the question? |
| 02:27.17 | p3nguin | I don't think so. |
| 02:27.20 | diijiib | did you see the 11 lines previous? |
| 02:27.37 | WIMPy | yes |
| 02:27.45 | diijiib | was that a 'i dont think so tim' |
| 02:27.52 | diijiib | so the question was. what do? |
| 02:28.29 | diijiib | do my configs look right, why is debug giving me errors and why have the phones stopped working |
| 02:28.38 | diijiib | ? |
| 02:29.03 | WIMPy | Oh, you've got errors and issues with your phones? |
| 02:29.19 | WIMPy | I didn't see any mention of that before. |
| 02:30.01 | diijiib | ok well then yes thats the case sit |
| 02:30.02 | diijiib | sir |
| 02:32.28 | diijiib | so.... anything wrong in those configs? |
| 02:35.54 | diijiib | or not interested |
| 02:35.57 | diijiib | ??? |
| 02:36.42 | WIMPy | Did you not that part about being specific? |
| 02:37.54 | WIMPy | Mu magical glass sphere is currently away for repair. |
| 02:37.57 | diijiib | ok specifically there is an issue which is unknown to me in my configs |
| 02:38.12 | diijiib | http://pastebin.com/un8Sy5Ke |
| 02:38.31 | WIMPy | What is that? |
| 02:38.40 | diijiib | its a pastebin of my configs. |
| 02:39.04 | WIMPy | That's not a kind of config, I've seen before. |
| 02:39.25 | diijiib | its sip.conf & extensions.conf on that site to make it east to see |
| 02:39.35 | diijiib | or would you like me to spam them in here? |
| 02:39.48 | diijiib | easy not east |
| 02:40.15 | *** join/#asterisk precognist_ (~precognis@75.15.226.185) |
| 02:40.30 | p3nguin | You'd be flooding if you pasted it here. |
| 02:40.39 | p3nguin | Spam is what I get in my email every day. |
| 02:40.41 | diijiib | hey Precognist is in, maybe he has the magical glass sphere |
| 02:41.06 | diijiib | see dude im not an irc guru, im just some guy with an asterisk problem |
| 02:41.16 | diijiib | i mean you no offence |
| 02:41.16 | p3nguin | And just so you know, that paste is neither sip.conf nor extensions.conf. |
| 02:41.24 | diijiib | its not? |
| 02:41.29 | p3nguin | No, it's not. |
| 02:41.53 | diijiib | grabbed the wrong one, that one is the dialplan debug |
| 02:41.58 | diijiib | this is the confs |
| 02:41.59 | diijiib | http://pastebin.com/QPYxhrmA |
| 02:42.04 | p3nguin | I have no flippin' clue what that other paste was. |
| 02:42.12 | WIMPy | And you couldn't be much more vague about what you're trying to fix. |
| 02:42.16 | Precognist | glass sphere? |
| 02:42.24 | diijiib | in asterisk console when you, 'dialplan debug' gives you that |
| 02:43.01 | diijiib | Precognist, WIMPy was was making fun of me earlier cuz im new or something |
| 02:43.28 | p3nguin | And what was the problem again? I see the confs now. |
| 02:43.33 | Precognist | ahhh |
| 02:43.41 | diijiib | nothing works. all are extension not found. |
| 02:44.03 | p3nguin | Where is the call coming from which fails? |
| 02:44.52 | diijiib | either 100 -> 200 fails, 200 -> 100 fails, 100or200 -> voipms fails, voipms -> all internal fail |
| 02:45.07 | diijiib | cant reach voicemailmain |
| 02:45.10 | p3nguin | So basically you have nothing working. Nothing at all. |
| 02:45.37 | diijiib | was working 100 percent until i think that symbolic link |
| 02:45.57 | diijiib | do the configs look ok context and syntax wise? |
| 02:46.02 | p3nguin | mostly |
| 02:46.33 | diijiib | sip show peers is good for 100 & 200 |
| 02:46.40 | p3nguin | Your contexts are all jacked up, but the syntax seems okay. |
| 02:46.52 | diijiib | jacked up how? |
| 02:47.00 | p3nguin | There should be one context for incoming calls, and it's not going to be the same as the phones use. |
| 02:47.32 | p3nguin | But you have everything set to a context called "mycontext," which does not exist. |
| 02:47.41 | diijiib | so incoming outgoing internal |
| 02:47.59 | p3nguin | Do you want me to rewrite it correctly? |
| 02:48.02 | diijiib | mycontext doesnt exist? |
| 02:48.20 | p3nguin | It's not in the paste you showed me. |
| 02:48.23 | diijiib | no just tell me what config (sip.conf ?) |
| 02:48.38 | diijiib | k let me check that, thanks p3nguin |
| 02:48.41 | p3nguin | both sip.conf and extensions.conf need help. |
| 02:49.40 | p3nguin | Inbound calls should never have access to the outbound context. |
| 02:49.42 | diijiib | lol |
| 02:49.58 | p3nguin | Phones need not include the inbound context, because they will never call from outside. |
| 02:49.58 | diijiib | see this is my first attempt at asterisk today. |
| 02:50.01 | diijiib | so im learning |
| 02:50.26 | diijiib | looked like ['my'context] my was missing from extensions |
| 02:50.41 | p3nguin | I'm going to show you how it should look. |
| 02:52.16 | diijiib | your rewriting both configs? |
| 02:52.29 | diijiib | where can i cand you paypal moneys? |
| 02:56.54 | diijiib | pretty involved eh? |
| 02:57.03 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 02:57.31 | WIMPy | No, because you make us guess what your're on about. But you need to tell us, if you expect help. |
| 02:57.36 | WIMPy | Sorry for the delay. |
| 02:58.08 | diijiib | hey man, fixing that conte of the system back. |
| 02:58.18 | diijiib | now im back to the symbolic link issue. |
| 02:58.31 | diijiib | aside from my poor formatting and config building skills |
| 02:58.34 | p3nguin | http://pastebin.com/0aQH9eat |
| 03:00.53 | diijiib | p3nguin, did you only change extensions or both? |
| 03:01.28 | p3nguin | I changed sip.conf and extensions.conf. |
| 03:02.05 | diijiib | good to set dtmfmode? |
| 03:02.10 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 03:02.11 | diijiib | or i had auto? |
| 03:02.17 | p3nguin | You had auto. |
| 03:04.02 | p3nguin | What I have changed will be closer to how it should be. You had a lot of nonsense going on. |
| 03:04.27 | diijiib | :D |
| 03:04.28 | diijiib | lol |
| 03:04.34 | diijiib | you changed everything eh |
| 03:04.40 | diijiib | contexts |
| 03:04.43 | p3nguin | Pretty much. |
| 03:04.55 | diijiib | syntax with my 1,2,3,n actions |
| 03:04.59 | diijiib | yikes thanks man |
| 03:05.48 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:05.59 | p3nguin | Like I said, lots of nonsense. :) |
| 03:11.29 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 03:15.14 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:15.19 | diijiib | thanks p3nguin that was great. * is back up to 110% |
| 03:16.28 | p3nguin | That sounds like a good thing. |
| 03:16.54 | diijiib | if all my context nonsense is fixed YOU ARE THE MAN |
| 03:17.19 | diijiib | so now AutoMixMon, what do? |
| 03:17.30 | p3nguin | I haven't even heard of that before. |
| 03:18.17 | diijiib | its a more advance 'automon' |
| 03:18.32 | diijiib | mixes the two files into one apparently |
| 03:18.43 | diijiib | like outgoing sound/incoming |
| 03:18.46 | p3nguin | I take it that it's an automon version of MixMonitor(). |
| 03:19.23 | diijiib | i would presume you would be right with my limited background |
| 03:19.59 | p3nguin | automon is to Monitor() as automixmon is to MixMonitor(). |
| 03:20.02 | p3nguin | :) |
| 03:20.07 | diijiib | apparently in your sytax you use ,x instead of ,w you would use for incall recording with automon |
| 03:20.31 | diijiib | would i need to put that? |
| 03:20.41 | diijiib | Dial(SIP/XXX,x) |
| 03:20.47 | diijiib | MixMonitor() |
| 03:20.48 | p3nguin | Is that a patch or is that someone available in newer Asterisk versions? |
| 03:21.02 | diijiib | its in 16 |
| 03:21.04 | p3nguin | MixMonitor() is the app that you would use in dial plan. |
| 03:21.29 | p3nguin | If there is an automixmon, that's a feature; see features.conf to configure it. |
| 03:24.21 | diijiib | ;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call! |
| 03:24.52 | p3nguin | nice. Uncomment it to use it. |
| 03:25.15 | p3nguin | Then set x or X, depending on which side of the call you want to be able to start/stop the recording. |
| 03:25.31 | diijiib | so if caller as ,X and if callee plan use ,x? |
| 03:26.04 | p3nguin | Where do you want the option to go? I would put it in my internals and use x. |
| 03:26.46 | p3nguin | In the outbound plan, I would use X. |
| 03:27.16 | diijiib | ok so in voipms-outbound use X in voipms-inbound use x |
| 03:27.16 | p3nguin | That way if someone calls you on 100, you can turn on recording; if you call outbound, you can turn on recording. |
| 03:27.46 | diijiib | ok ok i hear you as, 100 is getting hit anyways. |
| 03:28.18 | p3nguin | In the inbound, if you are doing Dial(SIP/100,20,x), then 100 would be able to enable recording when someone called in on the 877 number. |
| 03:28.44 | diijiib | inbound or internal? |
| 03:28.55 | p3nguin | Or you could change the Dial() to a Goto(internal,100,1). |
| 03:28.59 | diijiib | ok nvmd inbound |
| 03:29.11 | diijiib | goto does what? |
| 03:29.19 | diijiib | any quicker? |
| 03:29.42 | diijiib | and whats the 1 you have behind 100? |
| 03:29.51 | p3nguin | It moves the dial plan execution to another place. It's not going to be quicker, but it keeps the phones' Dials in one place instead of all over the place. |
| 03:29.58 | p3nguin | priority 1 |
| 03:31.12 | diijiib | so ? exten => 877XXXXXXX,1,Goto(internal,100,1,x) |
| 03:31.20 | p3nguin | no |
| 03:31.30 | p3nguin | Goto(internal,100,1) |
| 03:31.31 | diijiib | i fail |
| 03:31.41 | diijiib | where would the x option come in? |
| 03:31.47 | p3nguin | x is a Dial() option, not a Goto() option. |
| 03:31.57 | p3nguin | In the internal context, where the phone is being dialed, of course. |
| 03:32.29 | diijiib | how long have you been using asterisk? |
| 03:32.47 | p3nguin | Exactly? I don't know... |
| 03:32.51 | p3nguin | Approximately? A while. |
| 03:32.53 | diijiib | lol |
| 03:33.04 | diijiib | i love it. very cool system |
| 03:33.12 | james_zhu | :) |
| 03:33.27 | james_zhu | yes, asterisk is a new world |
| 03:33.29 | diijiib | so just use goto in internal |
| 03:33.41 | p3nguin | That's certainly one way to do it. |
| 03:33.49 | p3nguin | Then you don't have Dial()s all over. |
| 03:38.15 | *** join/#asterisk bmg505 (~leon@196-209-44-142.dynamic.isadsl.co.za) |
| 03:38.19 | diijiib | hows this look? |
| 03:38.20 | diijiib | http://pastebin.com/cw9nvBFc |
| 03:38.48 | p3nguin | screwed up |
| 03:39.15 | p3nguin | It looks like you completely guessed at how to use Goto() and Dial(). |
| 03:39.42 | p3nguin | Goto(somecontext,someextension,somepriority) |
| 03:40.37 | diijiib | i dont get it then |
| 03:40.42 | diijiib | not tonight anyways |
| 03:40.49 | *** join/#asterisk nix8n82 (~nate@24.143.28.16) |
| 03:42.49 | p3nguin | http://pastebin.com/VEnuwuiV |
| 03:43.22 | p3nguin | crap, error... |
| 03:43.33 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 03:43.55 | p3nguin | fixed, reload. |
| 03:44.23 | p3nguin | Forgot the double comma in the outgoing, but I fixed it. |
| 03:45.04 | diijiib | can i have multiple extensions in the goto command. before my troubleshooting i has it Dial(SIP/100&SIP/200,20) |
| 03:45.15 | p3nguin | Those aren't extensions. |
| 03:45.26 | diijiib | 100 & 200 are though |
| 03:45.33 | p3nguin | no, they aren't. |
| 03:45.53 | diijiib | oh asterisk day one |
| 03:46.15 | p3nguin | Extensions start with exten in extensions.conf. |
| 03:46.22 | p3nguin | SIP/100 is a device. |
| 03:46.35 | diijiib | why double commas? |
| 03:46.41 | p3nguin | (or a phone, in your case) |
| 03:47.11 | diijiib | behind a pap2t |
| 03:47.12 | p3nguin | In the Dial, you have the tech, the peer, the extension, the timeout, then the options. |
| 03:47.42 | p3nguin | SIP/voipms/${EXTEN},120,X) for example |
| 03:47.54 | p3nguin | But you aren't using a timeout, so you leave out the number 120. |
| 03:48.00 | p3nguin | SIP/voipms/${EXTEN},,X) |
| 03:48.53 | p3nguin | And that's where I first made an error... I didn't leave the blank space for the timeout. |
| 03:50.16 | diijiib | so you need blank spaces where any ommited part is |
| 03:50.20 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 03:50.27 | *** part/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 03:50.50 | p3nguin | If you didn't leave the blank space for the timeout, it would parse X as the timeout value, and not parse any options. |
| 03:50.57 | p3nguin | But we want no timeout and X as the option. |
| 03:51.35 | diijiib | by that how can this be right ? exten => 100,1,Dial(SIP/100,15,x) |
| 03:51.40 | p3nguin | If you would have loaded it with the missing comma, you should have seen an error that basically said the same thing that I just said. |
| 03:52.04 | diijiib | your the man, i bet im getting on your nerves by now eh |
| 03:52.23 | diijiib | can i do goto(internal,100&200,1) |
| 03:52.34 | p3nguin | That line says: extension 100 will dial a device called 100 using the SIP channel driver, let it ring for 15 seconds before timing out, and it'll use x as the only option. |
| 03:52.40 | p3nguin | No, you cannot. |
| 03:52.58 | diijiib | so i need to use dial() |
| 03:53.03 | p3nguin | But you can create another extension in internal and have it Dial() both devices... then goto that extensions. |
| 03:53.09 | p3nguin | extension, I mean. |
| 03:53.31 | p3nguin | or just use Dial() in the extension directly rather than the Goto(). |
| 03:53.35 | p3nguin | either way will be fine. |
| 03:54.04 | diijiib | k |
| 03:54.09 | diijiib | thanks so much man |
| 03:54.15 | diijiib | i really appreciate this |
| 03:58.25 | diijiib | any idea where automixmon would output to? |
| 03:58.39 | p3nguin | /var/spool/asterisk/monitor/ |
| 03:58.44 | diijiib | nein |
| 03:58.51 | p3nguin | That's the typical place. |
| 03:58.52 | diijiib | nothing there, but asterisk/voicemail |
| 03:59.08 | diijiib | i can hear the option entered, and call continues |
| 03:59.12 | diijiib | but no output |
| 03:59.13 | p3nguin | I'm sure it's different in a dd-wrt build. Check asterisk.conf for the paths. |
| 04:00.00 | diijiib | that location is defined |
| 04:00.06 | diijiib | as astspooldir |
| 04:00.18 | p3nguin | You can watch core verbose output to make sure the monitor gets started. |
| 04:00.35 | diijiib | core verbose ? |
| 04:00.42 | p3nguin | core set verbose 4 |
| 04:01.01 | diijiib | kk |
| 04:01.06 | p3nguin | Then make a call and press *3 after the other side has answered. |
| 04:05.13 | diijiib | no errors on caller side, had an error on callee but that becuase of no option set i suppose |
| 04:05.29 | p3nguin | What error? |
| 04:05.32 | diijiib | features.c:1115 builtin_automixmonitor: Cannot record the call. The mixmonitor application is disabled. |
| 04:05.46 | diijiib | thats when i did *3 from callee |
| 04:05.53 | p3nguin | oh |
| 04:05.59 | diijiib | but no error on caller when i did the *3 |
| 04:06.02 | diijiib | no new files |
| 04:06.11 | p3nguin | Are you calling out from the SIP phone? |
| 04:06.22 | diijiib | no only internal |
| 04:06.34 | p3nguin | You're calling from internal to internal? |
| 04:06.36 | diijiib | 200 -> 100 |
| 04:06.40 | diijiib | yah |
| 04:06.49 | diijiib | or that should work |
| 04:06.50 | diijiib | ? |
| 04:06.53 | p3nguin | Then only 100 can start the recording. |
| 04:07.21 | p3nguin | That explains why there was no error when 200 pressed *3. |
| 04:07.36 | p3nguin | it was ignored. |
| 04:09.18 | diijiib | i just got this trying to get a call from outside |
| 04:09.19 | diijiib | NOTICE[556]: chan_sip.c:20059 handle_request_invite: Call from '128869' to extension 's' rejected because extension not found |
| 04:09.45 | diijiib | need to defins 's'? |
| 04:09.54 | p3nguin | no, just wait a second. |
| 04:10.36 | diijiib | in voipms-inbound instead of 877XXXXXXX but 128869? |
| 04:10.40 | diijiib | put |
| 04:10.56 | p3nguin | No, stop guessing. |
| 04:11.02 | ChannelZ | Maybe try 389472138 |
| 04:11.16 | diijiib | i can dial out. |
| 04:11.17 | p3nguin | Might as well, since it's just as random. |
| 04:12.12 | p3nguin | Okay, you shouldn't need fromuser in the voipms entry in sip.conf, so take it out. |
| 04:12.58 | diijiib | line is gone |
| 04:13.06 | p3nguin | save, then sip reload |
| 04:14.52 | diijiib | did, still |
| 04:14.56 | diijiib | same thing |
| 04:14.57 | p3nguin | I'm curious how they managed to send a call to s. |
| 04:15.15 | p3nguin | They've never sent calls to s before that I've known. |
| 04:15.27 | diijiib | voipms? |
| 04:15.54 | p3nguin | yes |
| 04:16.03 | diijiib | its been sending calls to s since i set it up using the voipms config examples |
| 04:16.10 | diijiib | wanna see those? |
| 04:16.15 | p3nguin | http://pastebin.com/fJgNLGLM |
| 04:16.34 | p3nguin | ITSPs don't know how to configure Asterisk for end users. It's silly. |
| 04:17.14 | diijiib | http://pastebin.com/aGZAy6L9 |
| 04:17.22 | diijiib | thats the basic config |
| 04:17.52 | diijiib | whos's sip.conf is that? |
| 04:17.58 | p3nguin | whos's? |
| 04:18.05 | diijiib | who's |
| 04:18.07 | p3nguin | whose |
| 04:18.12 | diijiib | sure |
| 04:18.23 | p3nguin | I guess it's mine, since I wrote it. |
| 04:18.41 | diijiib | should have omited that crazy password no? |
| 04:18.44 | p3nguin | It's the one I give to everyone who has trouble configuring a peer for VoIP.ms. |
| 04:18.45 | diijiib | omitted |
| 04:18.52 | diijiib | cool |
| 04:18.55 | p3nguin | It's not real. |
| 04:19.12 | diijiib | should the permit & deny lines be there? |
| 04:19.18 | p3nguin | Yep. |
| 04:19.20 | diijiib | i have a dynamic addy |
| 04:19.26 | p3nguin | But they don't. |
| 04:19.30 | diijiib | ok |
| 04:19.34 | diijiib | just checking what side that is |
| 04:19.54 | p3nguin | If you're not going to use chicago, you'll have to change the address in the permit. |
| 04:20.58 | p3nguin | I'd imagine you'll use montreal or toronto. |
| 04:21.08 | diijiib | toronto2 |
| 04:21.19 | diijiib | figured it would be less busy |
| 04:21.38 | p3nguin | I don't know why, but toronto2 is the same as toronto. |
| 04:21.58 | p3nguin | 174.137.63.206 |
| 04:22.41 | p3nguin | Make sure you use that IP address in the permit line if you use toronto{,2}. |
| 04:23.20 | p3nguin | I'm still puzzled as to what would make them send to extension s. They've NEVER done that. |
| 04:23.54 | diijiib | same submet on that permit? |
| 04:24.10 | diijiib | did you see the last pastbin i sent of there sample configs? |
| 04:24.17 | diijiib | in that would have bearing |
| 04:24.21 | p3nguin | yes, 255.255.255.255 means only the address listed rather than a range in a subnet. |
| 04:24.56 | p3nguin | I looked at it, but I didn't understand it. They don't know how to configure your Asterisk. |
| 04:25.16 | p3nguin | I have yet to find an ITSP that gives a good sample config to an end user. |
| 04:25.25 | p3nguin | And I've used a bunch. |
| 04:26.10 | p3nguin | They either don't work at all, or they have so much nonsensical crap that your system just accepts everything thrown at it. |
| 04:27.01 | diijiib | it definitly didnt work at all with the pap2t-na |
| 04:27.37 | *** join/#asterisk cerienjean (~iper@ALamentin-106-1-12-213.w90-43.abo.wanadoo.fr) |
| 04:27.56 | p3nguin | I don't understand why certain industries have companies operating within that industry and have no clue about the technologies they involve. |
| 04:28.11 | p3nguin | I don't get it. |
| 04:28.21 | p3nguin | I don't know how they can do it. |
| 04:28.38 | p3nguin | Call the cable company to fix a problem with the cable services... they have no clue what to do. |
| 04:28.42 | WIMPy | Technical knowledge isn't neccessary. Every manager knows that. |
| 04:28.42 | diijiib | you would think the market would reflect that |
| 04:29.01 | diijiib | so what about 's' |
| 04:29.08 | diijiib | should i go back to my |
| 04:29.10 | p3nguin | Use my example. |
| 04:29.31 | diijiib | k brb let me grab phones.. im outside smoking |
| 04:29.32 | p3nguin | Change your username, password, host, and permit. |
| 04:30.29 | diijiib | nope |
| 04:30.31 | diijiib | <PROTECTED> |
| 04:30.44 | p3nguin | Now that doesn't make any sense. |
| 04:31.19 | diijiib | im thinking exten => s,1,Dial(SIP/100&SIP/200,15,x) |
| 04:31.31 | diijiib | here let me bound my * box |
| 04:31.32 | diijiib | ? |
| 04:31.41 | p3nguin | While that will probably work, that doesn't fix the problem of them sending to extension s. |
| 04:31.48 | diijiib | no ill just restart the daemon. |
| 04:31.49 | diijiib | lol |
| 04:32.11 | p3nguin | You're just providing a workaround for a problem that I don't understand. |
| 04:32.11 | diijiib | they should be sending to my 877XXXXXXX |
| 04:32.12 | diijiib | ? |
| 04:32.19 | p3nguin | correct |
| 04:32.24 | diijiib | ok i follow you |
| 04:33.02 | p3nguin | I've set up a lot of systems with voipms, and never once do I remember any of them having calls sent to s. |
| 04:33.41 | p3nguin | I also do not remember any of them having call from <voipms username>. |
| 04:35.17 | diijiib | weird |
| 04:35.43 | diijiib | ive definined exten => s in internal and it still not working |
| 04:35.58 | p3nguin | What's it saying now? |
| 04:35.58 | diijiib | i think im missing something context wise...? |
| 04:36.01 | diijiib | same |
| 04:36.04 | p3nguin | Yes, you are. |
| 04:36.13 | ChannelZ | is that what your sip peer's context is set to? internal? |
| 04:36.15 | p3nguin | The calls from voipms don't go into internal. |
| 04:36.35 | p3nguin | They go to the voipms inbound context, as configured on the voipms peer in sip.conf. |
| 04:37.01 | p3nguin | But this still does not explain where the exten s came into play. |
| 04:37.17 | diijiib | k let me fix that. |
| 04:37.23 | ChannelZ | have to see a sip debug |
| 04:37.24 | p3nguin | I'm tempted to switch over to SIP and test a call from them. |
| 04:37.37 | *** join/#asterisk cerienjean (~iper@ALamentin-106-1-12-213.w90-43.abo.wanadoo.fr) |
| 04:37.48 | diijiib | your ousing AIX? |
| 04:37.50 | diijiib | using |
| 04:37.58 | p3nguin | IAX2, yes |
| 04:38.06 | p3nguin | AIX is a Unix. |
| 04:38.25 | diijiib | ya. sry |
| 04:38.26 | diijiib | lol |
| 04:38.45 | diijiib | less bandwidth using IAX2 |
| 04:38.47 | diijiib | i hear |
| 04:39.09 | p3nguin | Yeah, I use the trunking feature. |
| 04:39.42 | diijiib | ok that worked. voipms-inbound with s instead of 877xxxxxxx |
| 04:39.53 | diijiib | i dont even know. |
| 04:42.00 | ChannelZ | You know, I'm re-reading sip.conf.sample's 'Naming devices' section and its explanation of type=xx doesn't seem right |
| 04:43.03 | diijiib | k back to working. |
| 04:43.07 | diijiib | than again p3 |
| 04:43.10 | diijiib | p3nguin, |
| 04:43.28 | p3nguin | Well wtf... I can't figure out how to change from IAX2 to SIP on my DIDs. |
| 04:43.48 | diijiib | dont u just use the other configuration. |
| 04:44.07 | p3nguin | DID routing has to be set on the portal. It has to use either IAX2 or SIP. |
| 04:44.53 | p3nguin | I see a setting for the main account, but I use a sub account. I don't see any way to change the sub account. |
| 04:45.41 | diijiib | it would be DID Numbers > Manage DID(s) > edit > sip/iax dropdown... but i think it was a setup option |
| 04:46.04 | diijiib | use a sub-account? |
| 04:48.45 | wasanzy | some one suggested last time I can only use digium card for ss7 |
| 04:48.45 | diijiib | how can there be 190 users in here and only 3 active |
| 04:49.01 | wasanzy | <PROTECTED> |
| 04:49.27 | ChannelZ | wasanzy: WHY DO YOU WANT TO USE SS7 |
| 04:49.40 | ChannelZ | The questions you've been asking suggest you really don't know why |
| 04:50.24 | WIMPy | wasanzy: You can use any of the PRI cards. |
| 04:50.42 | wasanzy | ok |
| 04:51.02 | WIMPy | At least any that end up using dahdi. |
| 04:52.05 | wasanzy | but the sangoma A200 analog, also useses dahdi but it is said not to support ss7 |
| 04:52.18 | p3nguin | I guess I can't change a sub account from IAX2 to SIP or SIP to IAX2. It looks like only the main account can switch back and forth. |
| 04:52.44 | p3nguin | So I'll have to create another sub account for SIP. Silly, but I'm going to do it so I can see what they are doing. |
| 04:52.53 | wasanzy | am trying to but some information together, that is why I came back with the questions |
| 04:53.08 | WIMPy | Unfortunately I don't have a log, but I told you several times, you need a digital connection, a PRI as the absolute minimum. |
| 04:53.27 | wasanzy | ok |
| 04:56.10 | diijiib | lol @ p3nguin |
| 04:58.10 | p3nguin | Okay, it's working as it should. The call goes to the exten matching my DID. |
| 04:58.23 | p3nguin | not to 's' |
| 04:58.24 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 04:58.59 | diijiib | then whats with mine? |
| 04:59.21 | p3nguin | My only guess is that you didn't use my config from the pastebin. |
| 04:59.22 | diijiib | should i try doing it through the sub account? |
| 04:59.27 | p3nguin | no |
| 04:59.28 | diijiib | i so did |
| 04:59.37 | p3nguin | I have no other explanation. |
| 04:59.40 | diijiib | ill show you if you want |
| 05:00.02 | p3nguin | If you want to paste your entire sip.conf, that would be okay. Hide only your passwords. |
| 05:00.14 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:02.57 | diijiib | http://pastebin.com/3KBCY0Qr |
| 05:03.00 | diijiib | at your command |
| 05:03.10 | diijiib | callerid is ok to define now eh? |
| 05:03.38 | p3nguin | no |
| 05:03.50 | p3nguin | Well, yeah, you can if you want. |
| 05:03.57 | p3nguin | But I typically do it in dial plan. |
| 05:04.04 | p3nguin | So that's why I leave it out of sip.conf. |
| 05:04.15 | diijiib | where in dialplan do i put it? |
| 05:04.33 | p3nguin | I set it before the outbound Dial(). |
| 05:04.49 | p3nguin | I set it based on the phone and "line" on the phone. |
| 05:05.10 | p3nguin | That's why I don't hard-wire it in sip.conf. |
| 05:05.22 | p3nguin | If you define it in the peer entry, you can't override it in dial plan. |
| 05:05.37 | diijiib | how would you do it here? |
| 05:05.42 | diijiib | exten => _1NXXNXXXXXX,1,Dial(SIP/voipms/${EXTEN}, ,X) |
| 05:06.03 | p3nguin | You have an extraneous space. |
| 05:06.20 | p3nguin | But it would go before that line, and you'd have to change that line's priority from 1 to n. |
| 05:06.35 | diijiib | before it how? |
| 05:07.19 | p3nguin | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=whateveryourphonenumberis) |
| 05:07.31 | p3nguin | exten => _1NXXNXXXXXX,n,Dial(SIP/voipms/${EXTEN},,X) |
| 05:08.23 | p3nguin | To prevent having to renumber the priority 1 all the time, I usually use a NoOp() on line 1 which never gets changed, then all others use n. |
| 05:08.29 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:08.39 | p3nguin | exten => _1NXXNXXXXXX,1,NoOp() |
| 05:08.44 | p3nguin | exten => _1NXXNXXXXXX,n,Set(CALLERID(num)=whateveryourphonenumberis) |
| 05:08.46 | p3nguin | exten => _1NXXNXXXXXX,n,Dial(SIP/voipms/${EXTEN},,X) |
| 05:09.05 | p3nguin | I feel like complexity is building. |
| 05:09.22 | diijiib | lol yeh my brain is about to explode |
| 05:09.39 | p3nguin | Get out the plastic to hang on the walls. |
| 05:10.21 | diijiib | and have the ammonium at the ready |
| 05:11.41 | diijiib | http://www.asteriskguru.com/tutorials/calleridname_function_image272425.jpg |
| 05:12.46 | p3nguin | Same concept, but not current. |
| 05:13.29 | p3nguin | And also goofy: the Answer() is not needed since Playback() is the first app and it performs an answer. |
| 05:17.03 | diijiib | k laddies.. im off for the evening. p3nguin thank you for all your efforts.. ill be back tomorrow i presume since this is a bravenew world i have entered and i need to gain some XP and MANA |
| 05:17.09 | diijiib | thanks again |
| 05:17.28 | p3nguin | Good luck. See you the next time around. |
| 05:26.12 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:28.10 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279411209.dsl.bell.ca) |
| 05:28.30 | dijib | hey p3nguin did u ever figure out that s thing? |
| 05:28.46 | dijib | looking at my sip.conf? |
| 05:31.27 | p3nguin | In the past eleven minutes? No. |
| 05:31.57 | p3nguin | You showed me that you changed your config to what I gave you, and what I gave you works correctly on my system. |
| 05:32.09 | p3nguin | Did you run sip reload after putting in my config? |
| 05:49.42 | ChannelZ | or show a sip debug? if it's coming in with no extension, it's coming in with no extension. Unless you told them to in your register line... |
| 05:49.51 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:50.22 | p3nguin | I never send an extension in the register statement, and they always send calls to the extension which is the DID. |
| 05:52.37 | p3nguin | When I only had one DID with them, they still sent the call to it without specifying it in the register. |
| 05:53.58 | dijib | yes i did reload |
| 05:54.59 | ChannelZ | Hmph. |
| 05:56.22 | WIMPy | BTW: Does anyone know how the story ended with the guy not getting sync on the ports of the 2nd card? |
| 05:56.47 | ChannelZ | not me |
| 05:57.23 | p3nguin | I've configured many a system in the same way, and the calls always go to the DID as the extension. This is the first time I have ever seen it go to s from voipms. |
| 05:57.56 | p3nguin | This makes me think that maybe it could be a user-configurable option in the portal. |
| 05:58.16 | p3nguin | I have no idea what it would be called, but that's the only thing I can come up with. |
| 05:59.26 | p3nguin | His call was also coming from his username on their side. Calls to me are from my phone number (as a username) to my phone number (as an extension). |
| 05:59.26 | ChannelZ | well again if it pays attention to the exten given when you register, he could have messed that up |
| 05:59.57 | p3nguin | He says he's using the config I provided, which does not include an extension. |
| 06:00.18 | ChannelZ | people say things |
| 06:00.26 | p3nguin | Heh, I know. |
| 06:01.00 | p3nguin | I don't know how he can really prove it to me. He showed me the config, edited with his username and pop... |
| 06:01.15 | p3nguin | I can only assume he saved it and loaded it. |
| 06:01.15 | ChannelZ | Was he using a different regional hostname or something I thought? |
| 06:01.54 | p3nguin | Yeah, using the toronto pop; I use the chicago one currently, but I've used the toronto one before and it didn't send to s. |
| 06:02.24 | ChannelZ | shrugs |
| 06:02.34 | ChannelZ | who knows |
| 06:03.12 | *** join/#asterisk james_zhu (~Administr@183.16.215.92) |
| 06:04.21 | p3nguin | I sure don't, and I'm not going to fiddle with it anymore tonight. |
| 06:21.26 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:41.16 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:41.18 | schmidts | good morning |
| 06:41.34 | ChannelZ | It's a beautiful day in the neighborhood |
| 06:43.07 | schmidts | ChannelZ here its raining so the best weather for a productive monday :D |
| 06:45.13 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 06:46.01 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 06:48.55 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 06:49.15 | *** join/#asterisk mutex7c (~mutex7c@HSI-KBW-095-208-202-191.hsi5.kabel-badenwuerttemberg.de) |
| 06:50.03 | *** join/#asterisk james_zhu (~Administr@183.16.215.92) |
| 06:55.28 | *** join/#asterisk mutex7c (~mutex7c@HSI-KBW-095-208-202-191.hsi5.kabel-badenwuerttemberg.de) |
| 06:57.00 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:58.15 | *** join/#asterisk coppice (~coppice@m121-202-19-249.smartone-vodafone.com) |
| 07:01.07 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f04a:4bcf:670c:6bd0) |
| 07:06.39 | *** join/#asterisk mintos (~mvaliyav@114.143.164.195) |
| 07:09.50 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 07:15.29 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 07:18.19 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 07:44.55 | *** join/#asterisk hariom (~hariom@117.195.188.238) |
| 07:45.57 | hariom | How to play a prompt (.wav file) that is not located on the PC where asterisk is running. |
| 07:53.33 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 07:55.13 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 08:11.18 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 08:11.36 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 08:12.24 | mandla | Hello. |
| 08:14.10 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 08:20.37 | schmidts | hariom you have to have access through your filesystem like nfs |
| 08:27.09 | *** join/#asterisk dexteruk (~kvirc@78.142.1.236) |
| 08:29.27 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 08:29.36 | hariom | schmidts: and in case of record, is it possible to get the incoming audio on the system other than running asterisk. I am trying out fagi |
| 08:30.13 | dexteruk | i want to dial group of sip phones but i need to do a dundi lookup on all the devices to find out where they are |
| 08:30.38 | dexteruk | normally you dial Dial(SIP/1000&SIP/1001) |
| 08:31.02 | dexteruk | but these devices are elsewhere not on the local machine |
| 08:31.23 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 08:31.51 | dexteruk | Am i missing something simple |
| 08:32.21 | singler | dexteruk: do a lookup before dial and construct dial string from variables |
| 08:32.35 | dexteruk | can you give me an example? |
| 08:33.23 | singler | I did not use dundi, so no, but if you would pastebin your lookup dialplan, I could |
| 08:42.59 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:43.06 | jacc0 | hi all! good morning :) |
| 08:43.17 | schmidts | morning jacc0 |
| 08:48.04 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 08:52.16 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 08:54.39 | *** join/#asterisk jits1998 (b75260e6@gateway/web/freenode/ip.183.82.96.230) |
| 08:56.25 | jits1998 | hi guys.. we are trying to setup asterisk to be used for training across our centers .. the system will be using softphones only... we are facing problem with echo .. any hints how we can resolve this.. the problem is increased as distance increases.. |
| 08:57.34 | jacc0 | @jits1998 : use headphones |
| 08:57.39 | petern_ | do your softphones have any echo cancellation? |
| 08:57.45 | petern_ | indeed, headsets :) |
| 08:57.55 | jacc0 | :P |
| 08:59.03 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 08:59.04 | jits1998 | we are looking at using nate client.. that does not seem to have any echo cancellation |
| 08:59.41 | jits1998 | jacc0: even headphones don't help .. |
| 09:00.21 | jits1998 | i meant Yate client |
| 09:00.53 | coppice | headsets do help. they just aren't a 100% fix |
| 09:01.51 | jits1998 | coppice: the trainer can use headsets... but on the other site it will be a large number of students.. so we want to use the classroom audio system . |
| 09:02.40 | jacc0 | use a push-to-talk microphone |
| 09:02.56 | coppice | roger |
| 09:03.43 | jits1998 | jacc0: we get echo with all microphones switched off :-| |
| 09:05.05 | jacc0 | maybe you have configured the soundcard to use mixed audio is your audio source |
| 09:05.24 | jacc0 | *as |
| 09:05.37 | jits1998 | client is on windows 7 machine .. can you tell me exactly what to look for ? |
| 09:05.57 | jits1998 | btw when i am using skype, there is no echo at all .. |
| 09:08.06 | jacc0 | what clients are you using ; x-lite? |
| 09:09.04 | jits1998 | jacc0: yate client |
| 09:09.28 | jacc0 | right click on the speaker icon in your taskbar and select "recording devices" |
| 09:10.13 | jacc0 | and make sure you selected the microphone as default input device and not "sterio Mix" |
| 09:10.23 | jacc0 | *stereo |
| 09:10.43 | jacc0 | try x-lite from counterpath as a client |
| 09:11.49 | jacc0 | btw it's a good idea talk about echo cancelation in class :) |
| 09:12.06 | jits1998 | don't have the option do to stereo mix ... let me check x-lite .. |
| 09:13.11 | jits1998 | we tried x-lite but picked yate becuase it gives the option to have auto-answer configured as default option during startup .. |
| 09:13.28 | jacc0 | Hmm, asterisk is takingup 83% mem on a 1gb machine |
| 09:14.52 | jacc0 | what to do to findout what is taking up all the memory? |
| 09:15.44 | hariom | in response to agi command, if the response is 200 result=1 what does this '1' shows? |
| 09:15.50 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-zjwqomggsrtrzedc) |
| 09:16.06 | jacc0 | ok |
| 09:16.36 | jacc0 | Action completed successfully |
| 09:17.57 | jacc0 | @hariom: do you see anything in CLI at that moment? |
| 09:18.20 | hariom | jacc0: AGI Tx >> 200 result=1 |
| 09:18.46 | jacc0 | in "asterisk -rvvvv" does it show anything? |
| 09:18.59 | jacc0 | like "broken pipe" |
| 09:19.12 | hariom | no |
| 09:19.33 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-231-28.w86-204.abo.wanadoo.fr) |
| 09:19.50 | merlin8282 | Hi. I've problems setting up a correct connection to sipgate... |
| 09:20.10 | jits1998 | jacc0: will phone.conf help me ? |
| 09:20.33 | merlin8282 | I have an * 1.6.2.9-2+squeeze2 behind a full cone NAT, and I've one-way audio: only inward works. |
| 09:20.45 | hariom | jacc0: so when result=0 this means failed. result=-1 failed? |
| 09:20.50 | merlin8282 | SIP port 5060 and RPT ports 10000:20000 are forwarded |
| 09:24.04 | merlin8282 | it seems that the problem is, that the local IP is used in SDP, because asterisk sends SDP packets with "Audio is at 192.168.X.Y port 14878" |
| 09:25.50 | jacc0 | @hariom: I'm not sure what the 1 means, but the 200 means "Action completed successfully" |
| 09:25.51 | merlin8282 | I've already set "localnet=192.168.0.0/255.255.0.0" (and the 3 other that are in the example, also they're all local nets), externip = [extern IP from NAT]:5060 and nat = yes |
| 09:27.00 | tzafrir | merlin8282, 'nat=yes' helps when you're the external and the other party is behind nat |
| 09:27.11 | merlin8282 | also, SIP seems to work fine (invite, 100 trying, 180 ringing, 200 ok, sending ack, etc.) |
| 09:27.30 | hariom | jacc0: not exactly. 200 means command processed but was that successful or not is known only using 'result'. If result=-1, that means though command is process successfully but may not result as intended. |
| 09:28.08 | merlin8282 | tzafrir: ah, ok. But well; to be honest I tried also nat=route and nat=no, both with the same results. |
| 09:28.33 | hariom | jacc0: eg: playing file where file could not be read |
| 09:29.06 | singler | merlin8282: by saying that inward works, do you mean that you receive audio from provider? |
| 09:29.59 | merlin8282 | singler: yes |
| 09:30.39 | singler | check with tcpdump or simmilar program if your rtp packets are sent to correct destination |
| 09:31.00 | merlin8282 | singler: they are. I ran tcpdump on the NAT gateway |
| 09:31.21 | merlin8282 | it seems that the provider does not get the correct ip:port to send audio/RTP to |
| 09:32.11 | singler | but providers does send audio to you correctly |
| 09:32.18 | jacc0 | ? " do you mean that you receive audio from provider?" -> singler: yes |
| 09:32.23 | merlin8282 | singler: right |
| 09:33.13 | jacc0 | try setting qualify to 29 seconds; that way it will function as a kind of nat-keep-alive |
| 09:35.14 | merlin8282 | jacc0: tried it: nok. |
| 09:35.21 | merlin8282 | But sorry: inward audio does NOT work |
| 09:35.27 | merlin8282 | it's outgoing that is working. |
| 09:36.01 | *** join/#asterisk syntaxx (~patvan@unaffiliated/syntaxx) |
| 09:36.38 | syntaxx | hi can anyone suggest a good sip sotfphone client that supports video and have windows and linux compatibility? |
| 09:36.53 | merlin8282 | syntaxx: ekiga ? |
| 09:37.09 | syntaxx | merlin8282, does it runs on windows? i thinks its only on linux? |
| 09:37.18 | merlin8282 | syntaxx: yes, it runs under windows |
| 09:37.25 | merlin8282 | set it up last week on a test computer ;) |
| 09:37.47 | merlin8282 | I didn't test video, but audio works fine on both OSes |
| 09:37.56 | syntaxx | merlin8282, ok.. ill try to install ekiga on both |
| 09:37.58 | syntaxx | thanks.. |
| 09:41.35 | merlin8282 | jacc0, singler, tzafrir no further idea why it's not working ? |
| 09:42.43 | jacc0 | no clue; can you forward udp 5060 and 10000-20000 to the client behind nat? |
| 09:42.44 | tzafrir | merlin8282, I guess that the gateway messes up the port numbers and such |
| 09:42.58 | singler | not really.. you could try analyze rtp traffic with wireshark to check if audio is really in the packets, also setup some test "provider" and check if it works |
| 09:44.31 | merlin8282 | jacc0: the asterisk and the clients are in a LAN, and the asterisk tries to connect through a NAT to sipgate. |
| 09:45.05 | jacc0 | then set reinvite=no |
| 09:45.06 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 09:46.19 | jacc0 | sorry: canreinvite=no for the clients and forward udp port 5060 and 10000-20000 to your asterisk box |
| 09:49.00 | syntaxx | merlin8282, doesn't seem to work for me :( |
| 09:52.30 | merlin8282 | syntaxx: what does not work exactly ? |
| 09:52.37 | syntaxx | merlin8282, we cant hear each other |
| 09:53.41 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 09:53.45 | jacc0 | what about my memory usage problem? any tips on how to locate the memory leak? |
| 09:53.46 | syntaxx | we have this softphone client jitsi is the name but its under beta.. sometimes we can hear each other sometimes not =/ |
| 09:54.30 | jacc0 | when does it work? |
| 09:54.40 | jacc0 | if you call both ways within the minute? |
| 09:54.43 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 09:54.51 | syntaxx | jacc0, yes |
| 09:54.57 | jacc0 | nat-keep-alive |
| 09:55.14 | syntaxx | we are on a local lan |
| 09:55.24 | merlin8282 | jacc0: "canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does." --> I give it a try |
| 09:56.12 | jacc0 | Hmm, I'll have to fix it in my asterisk install; I have 1.8.5 and still use canreinvite |
| 09:57.36 | *** join/#asterisk derRichard (~derRichar@pippin.sigma-star.at) |
| 09:57.38 | derRichard | hi |
| 09:57.45 | merlin8282 | jacc0: i've set both canreinvite and directmedia to no, doesn't change anything :-( |
| 09:58.14 | kaii | merlin8282: please verify that it is really set by issuing a "sip show peer <name>" |
| 09:59.04 | jacc0 | canreinvite=yes sets DirectMedia : Yes |
| 09:59.08 | jacc0 | :) |
| 09:59.39 | derRichard | i'm looking for asterisk based office communication solutions like microsoft lync. (with support). can you recommend something? |
| 09:59.54 | kaii | jacc0: actually, yes is just the default. |
| 10:00.15 | jacc0 | canreinvite=no sets directmedia : no |
| 10:00.20 | kaii | ok. |
| 10:00.26 | jacc0 | 1.8.5 |
| 10:00.37 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 10:00.49 | merlin8282 | jacc0: it's set correctly: DirectMedia : No |
| 10:04.11 | catphish | how could i go about debugging a reproducible freeze in asterisk? |
| 10:04.41 | catphish | my making a large number of calls i can reach a state where i can still connect to asterisk's console but it won't accept any calls, or log anything, or restart |
| 10:05.30 | jacc0 | @catphish:use dahdi |
| 10:05.33 | *** join/#asterisk Godfather_ (~estanteri@90.162.100.241) |
| 10:05.49 | catphish | jacc0: how? |
| 10:06.37 | catphish | i'm just upgrading 1.8.4.4 to 1.8.5 |
| 10:07.09 | jacc0 | catphish:https://issues.asterisk.org/jira/browse/ASTERISK-18166 |
| 10:07.21 | jacc0 | set noload => res_timing_timerfd.so |
| 10:07.50 | catphish | it's likely a timing deadlock? |
| 10:08.03 | jacc0 | could be |
| 10:08.09 | jacc0 | I'm still investegating it |
| 10:08.10 | catphish | this is actually running on a kvm VM |
| 10:08.18 | catphish | i'll try your suggestion |
| 10:08.22 | catphish | and the upgrade to 1.8.5 |
| 10:08.26 | catphish | thanks |
| 10:08.33 | catphish | otherwise i'll try to gather some more info |
| 10:09.11 | jacc0 | yes; and add it to the bug report : https://issues.asterisk.org/jira/browse/ASTERISK-18166 (because I think it's the same thing) |
| 10:09.55 | merlin8282 | mmm... we're going to give the asterisk server an external IP, so there should be no more problem :/ |
| 10:10.32 | jacc0 | that is the best thing to do |
| 10:12.59 | catphish | "set noload => res_timing_timerfd.so" made no difference |
| 10:13.15 | catphish | i have autoload=yes so a lot of other stuff could be loaded |
| 10:13.21 | catphish | i'll check the locks |
| 10:13.59 | jacc0 | in CLI: core show locks |
| 10:14.28 | catphish | No such command 'core show locks' |
| 10:14.51 | *** join/#asterisk rutski (~rutski@ool-45708688.dyn.optonline.net) |
| 10:14.54 | rutski | Hey there guys |
| 10:14.56 | schmidts | catphish you have to compile asterisk with some compiler flags to get the core show locks command |
| 10:15.00 | catphish | ok |
| 10:15.01 | rutski | I've got this machine, but I can't physically get to it |
| 10:15.17 | rutski | It has 8 PSTN lines going into it |
| 10:15.26 | rutski | When SIP users use said machine to dial out, it uses a certain PSTN line by default |
| 10:15.37 | rutski | (I'm guessing it's just the first one plugged into the first port on the DAHDI card) |
| 10:15.43 | rutski | But I really need to make it use a different line by default |
| 10:15.45 | *** part/#asterisk derRichard (~derRichar@pippin.sigma-star.at) |
| 10:15.51 | rutski | but I can't physically get to the machine to plug different lines into the first port |
| 10:15.52 | rutski | any ideas? |
| 10:16.31 | jacc0 | dial(dahdi/2/${EXTEN}) to use line 2? |
| 10:16.50 | rutski | currently I have things like: |
| 10:16.53 | catphish | schmidts: do you know which flag? |
| 10:16.54 | rutski | exten => _XXXXXXX,3,Dial(DAHDI/G0/1914${EXTEN} |
| 10:16.59 | rutski | I thought you had to put a group number there |
| 10:17.03 | rutski | but you can put a line number? Interesting. |
| 10:17.07 | rutski | What if that line is in use? |
| 10:17.08 | *** join/#asterisk obruT (~turbo@bunika.babuncic.com) |
| 10:17.09 | jacc0 | not sure |
| 10:17.15 | rutski | Well, worth a try |
| 10:18.16 | schmidts | catphish i have to take a look |
| 10:18.22 | catphish | DEBUG_THREADS |
| 10:18.23 | schmidts | catphish which version? |
| 10:18.23 | catphish | i got iy |
| 10:18.28 | schmidts | yes ;) |
| 10:18.30 | catphish | 1.8.5 |
| 10:19.18 | schmidts | ok then you should nee debug_threads and imho there should also be a debug locks flag |
| 10:19.37 | jacc0 | there is |
| 10:20.19 | obruT | hello everyone... i found out that call waiting in asterisk is enabled by dialing *70... what I dont know, how does the dialplan should look like to catch that special extension and how to trigger execution of it ? |
| 10:22.39 | schmidts | obruT take a look at features.conf i guess you will find *70 in there |
| 10:25.05 | catphish | schmidts: jacc0: http://paste.codebasehq.com/pastes/flzp459c4ryw |
| 10:25.14 | catphish | any help would be appreciated |
| 10:28.14 | jacc0 | catphish: are you using monitor? |
| 10:28.30 | *** join/#asterisk james_zhu (~Administr@183.16.215.92) |
| 10:29.02 | Tuju | where i should map sip accounts and line numbers? |
| 10:29.28 | catphish | jacc0: in places yes, for this call, no |
| 10:29.40 | Tuju | extensions.conf ? |
| 10:30.45 | obruT | schmidts: no reference to *70 or call waiting in that file... I just googled for features.conf and call waiting, nothing... |
| 10:31.34 | jacc0 | catphish: do_monitor is waiting for a lock on channel 0x7fc1bc029e50 that is already locked by the pbx_thread |
| 10:32.16 | catphish | i see that, but i'm afraid that's the limit of my skill |
| 10:32.30 | catphish | i'm doing to disable res_monitor for now and see if it prevents the crash |
| 10:33.00 | jacc0 | okay, let me know the result and file a bug report please |
| 10:33.05 | Tuju | exten => tuju,5551,Dial(SIP/tuju); should that work? |
| 10:33.09 | catphish | but i guess the pbx_thread lock would break other things |
| 10:34.04 | *** join/#asterisk oktay (~oktay@81.215.202.193) |
| 10:34.28 | oktay | hi. anybody know how to do a firmware upgrade on a dlink DVG ? (2102-s) |
| 10:35.41 | catphish | err, it seems that do_monitor has nothing to do with res_monitor |
| 10:35.43 | catphish | it's in sip |
| 10:36.54 | jacc0 | hmm |
| 10:37.24 | jacc0 | I guess you will have to file a bug report |
| 10:37.35 | jacc0 | issues.asterisk.org/jira |
| 10:37.37 | catphish | https://bugs.digium.com/view.php?id=15349&nbn=12 |
| 10:37.40 | catphish | perhaps |
| 10:42.48 | schmidts | jacc0 do_monitor has nothing to do with the monitor app, its just the function in chan_sip.c which loops endless and handle incoming and scheduled messages |
| 10:43.05 | schmidts | catphish which version do you use cause it looks like a know bug to me |
| 10:43.23 | catphish | it looks like a known bug |
| 10:43.29 | catphish | i'm using 1.8.5 |
| 10:43.46 | catphish | looks like this problem was fixed in 1.6.2 |
| 10:43.52 | schmidts | ah ok, cause there will be a patch for this but imho it will be in 1.8.6 |
| 10:44.41 | catphish | why would it be so much later? looks like it was fixed in 2009? |
| 10:44.49 | schmidts | catphish you could try the svn checkout of 1.8 branch |
| 10:45.05 | schmidts | you are talking about another deadlock problem ;) there are many out there :D |
| 10:45.10 | catphish | makes sense |
| 10:45.15 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 10:45.38 | catphish | hopefully if there's a patch in svn i can apply it to 1.8.5 |
| 10:46.08 | schmidts | give me a moment and i can say you the reviewboard url, there you can download the patch directly |
| 10:46.30 | catphish | that would be awesome |
| 10:47.20 | schmidts | https://reviewboard.asterisk.org/r/1313/ |
| 10:51.31 | *** join/#asterisk wonderworld (~ww@port-92-201-171-153.dynamic.qsc.de) |
| 10:52.45 | catphish | that diff doesn't seem happy to apply to 1.8.5 at all |
| 10:53.27 | schmidts | maybe you have to patch it by hand |
| 10:53.41 | catphish | perhaps, it's huge though |
| 10:53.42 | schmidts | i guess there was some changes between 1.8.5 and the svn rev this base on |
| 10:53.48 | catphish | makes sense |
| 10:54.00 | catphish | i might just test against trunk |
| 10:55.29 | schmidts | just use the svn checkout of 1.8 its allready in there ;) |
| 10:55.45 | catphish | i'm trying that now |
| 10:57.09 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 10:57.58 | catphish | oops, i'm compiling trunk not 1.8 |
| 10:58.12 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:58.26 | schmidts | :D |
| 10:58.33 | catphish | http://svn.asterisk.org/svn/asterisk/branches/1.8 :) |
| 10:58.43 | catphish | actually reading things helps |
| 10:59.47 | Tuju | how do i redirect my call 5551 into sip:6661@example.com ? |
| 11:00.22 | schmidts | Tuju exten => 5551,1,Dial(SIP/6661@example.com) |
| 11:00.26 | Tuju | i tried: exten => 5551,1,Dial(SIP/6661@example.com) but it whines about 0 being first number. |
| 11:00.44 | Tuju | hmmm.... |
| 11:01.21 | Tuju | <PROTECTED> |
| 11:02.20 | schmidts | tuju this looks more like a message when you do a goto but not a dial |
| 11:02.42 | catphish | apparantly i'm missing defaults.h |
| 11:03.14 | Tuju | schmidts: hmmm... that could be, i tried goto earlier. |
| 11:03.29 | Tuju | and now i get this one (more weird error) app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 11:04.26 | schmidts | tuju that looks better for me ;) you should try to enable sip debug to see if you get something back from your other side |
| 11:07.44 | catphish | make all helps |
| 11:11.44 | singler | Tuju: it whines that first number must be more than 0, connect to asterisk with -vvvvvvvvvv and pastebin output above your SIP channel error |
| 11:12.13 | Tuju | schmidts: ack, good to know. this is my first time in my life i'm hacking the asterisk (although waited for years, former ser & others user) and I'm very excited :) |
| 11:12.39 | Tuju | singler: schmidts was right, that came from goto which is now gone. |
| 11:13.56 | catphish | schmidts: my bug is still present in 1.8-trunk |
| 11:14.38 | catphish | i'll try 2.10 |
| 11:18.12 | jacc0 | there is no 2.10 |
| 11:22.45 | catphish | don't worry, i know what i mean :) |
| 11:23.30 | catphish | 10.0-trunk is what i mean :) |
| 11:28.25 | catphish | 10.0 exhibits the same behaviour, wonder if it's the combination of my VM and the test i'm using |
| 11:29.11 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 11:31.34 | johnnyasterisk | Hi is there a way I can allow any registrations from a specific ip address. When any calls come from that ip address I then want to specify that the calls from that ip address below to a specific extension |
| 11:32.17 | catphish | if you know the IP, the device doesn't need to register at all afaik |
| 11:32.29 | catphish | you set up a peer for it, and specify a context for calls from that peer |
| 11:34.08 | johnnyasterisk | well when a call comes from that ip address it will be from XXXX@1.2.3.4 |
| 11:34.17 | johnnyasterisk | the XXXX can be any 4 digit extension |
| 11:37.11 | *** join/#asterisk brezular (~brezular@adsl-dyn-206.95-102-98.t-com.sk) |
| 11:37.53 | jacc0 | then do as catphish adviced and use execif($["CALLERID(num)" = 10]?whatever) |
| 11:37.55 | catphish | schmidts jacc0 I believe you were right about res_timing_timerfd.so |
| 11:38.04 | jacc0 | :) |
| 11:38.13 | catphish | timerfd was being loaded despite my noload |
| 11:38.18 | catphish | deleting it helped |
| 11:38.33 | catphish | asterisk is still failing heavily with my test, but not locking in the process :) |
| 11:38.37 | *** join/#asterisk cerienjean (~iper@ALamentin-106-1-12-213.w90-43.abo.wanadoo.fr) |
| 11:38.58 | jacc0 | thank leifmadsen for it |
| 11:40.06 | jacc0 | could you still add your output from core show locks to the bug report? |
| 11:40.25 | jacc0 | and can you tell how you reproduced it? |
| 11:40.37 | *** join/#asterisk james_zhu (~Administr@72.11.141.154) |
| 11:40.43 | jacc0 | dev. team will need it to fix the bug |
| 11:41.08 | catphish | i will get some info together |
| 11:41.24 | catphish | what timing should i be using? |
| 11:41.45 | catphish | i'll drop the other modules for now |
| 11:43.32 | catphish | will try dahdi only for now |
| 11:43.44 | *** join/#asterisk GreatSUN (~greatsun@88-117-0-211.adsl.highway.telekom.at) |
| 11:44.24 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.39) |
| 11:44.40 | GreatSUN | hi all |
| 11:44.52 | GreatSUN | short feature question: |
| 11:45.34 | GreatSUN | is it possible to generate conferences (hold one party, dial another one and make a conference with those) on asterisk server side? |
| 11:54.37 | jacc0 | @catphish: you should use dahdi |
| 11:54.52 | catphish | ok i'm doing that now |
| 11:55.29 | catphish | i'm confused that my ram usage has suddenly increased enourmously |
| 11:56.24 | jacc0 | :) I'm looking at something simular right now |
| 11:57.44 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 11:57.58 | catphish | would setting dahdi as my timing source have caused a large increase in ram usage? |
| 11:58.17 | catphish | i may just be going mad of course |
| 12:00.03 | jacc0 | I have asterisk takingup 83% of mem on a 1gb machine :S |
| 12:01.03 | *** join/#asterisk james_zhu (~Administr@183.16.215.92) |
| 12:01.26 | catphish | mine's using 46% on a 512 VM |
| 12:02.50 | catphish | but it was using < 10 before i started looking at the deadlock |
| 12:08.36 | *** part/#asterisk james_zhu (~Administr@183.16.215.92) |
| 12:10.36 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:19.03 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 12:19.52 | jacc0 | inotify_daemon : name = '\000' <repeats 2856 times> |
| 12:20.11 | jacc0 | at stdtime/localtime.c:290 |
| 12:20.52 | *** join/#asterisk brah (be88a535@gateway/web/freenode/ip.190.136.165.53) |
| 12:22.22 | jacc0 | that is 4 bytes repeated 2856times = 11424 bytes |
| 12:25.06 | jacc0 | it seems to be amemory leak issue with the timing module |
| 12:26.06 | singler | jacc0: it appears that I am also having issues mentioned in 181166 and 18142 (asterisk v1.8.5). I have backtraces, but debug log is not prepared yet, should I upload my backtraces? |
| 12:27.10 | jacc0 | yes upload as much as you can; even if it's not usefull - just to let them know there are more people having the same issue - so they wont close it |
| 12:27.30 | jacc0 | :P |
| 12:27.36 | *** join/#asterisk EmbouNT (~DanteAggo@77.73.161.250) |
| 12:27.45 | singler | to which bug I should upload them? :) |
| 12:27.51 | singler | or maybe both? |
| 12:29.36 | EmbouNT | hello, i have a qestion, there are any repository of asterisk 1.8.5 for CentOS 6.0? |
| 12:30.13 | singler | EmbouNT: why don't you compile it from source? |
| 12:30.15 | EmbouNT | in packages.asterisk.org i only get 4 or 5, but centos 6.0 repo isn't listed |
| 12:30.32 | EmbouNT | i'm compiling from source, but i need mysql capabilities, and get me errors |
| 12:30.48 | EmbouNT | withour res_mysql, cdr_mysql i can compile perfect |
| 12:31.01 | EmbouNT | but making make menuconfig and selecting mysql, gave me errores |
| 12:31.02 | singler | install mysqlclient-dev or simmilar package |
| 12:31.05 | EmbouNT | errors* |
| 12:31.44 | singler | on debian I would tell you exact name, but not for centos.. |
| 12:32.20 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 12:33.20 | EmbouNT | ok i'll try to install these packages |
| 12:34.11 | EmbouNT | thanks |
| 12:35.19 | Tuju | does anyone here have a 100% working cisco 7975G sip-configuration? |
| 12:35.29 | Tuju | I've tried quite many of them and constantly have problems with getting it to start registration. I got it once working, but then changed config and that state has long gone. |
| 12:35.51 | Tuju | compared to the old 7960 those java based ones are real pita to get working |
| 12:36.29 | tzafrir | Anybody here uses a Sangoma BRI device with DAHDI drivers? |
| 12:36.49 | tzafrir | If so: what's the output of: cat /proc/dahdi/* #? |
| 12:40.06 | singler | tzafrir: are you sure Sangoma BRI can be used with dahdi? Last time I used it, it used Woomera channel |
| 12:43.37 | *** join/#asterisk coppice (~coppice@m121-202-19-249.smartone-vodafone.com) |
| 12:46.27 | jacc0 | tzafrir: what asterisk version are you using? |
| 12:47.42 | *** join/#asterisk marclaporte (~Miranda@69-165-165-53.dsl.teksavvy.com) |
| 12:48.37 | tzafrir | jacc0, why should it matter? |
| 12:48.39 | jacc0 | 1.8.X is not supported by sangoma |
| 12:48.48 | jacc0 | so? |
| 12:48.52 | tzafrir | I care about the DAHDI version |
| 12:49.06 | tzafrir | But why should the care? |
| 12:49.35 | jacc0 | you n eed to use woomera ; woomera doesn't support 1.8.x |
| 12:50.14 | singler | then SMGv3 should be used I guess |
| 12:50.19 | jacc0 | yes |
| 12:50.24 | jacc0 | that is what they adviced me |
| 12:50.30 | jacc0 | SMGv3 |
| 12:50.39 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 12:50.44 | jacc0 | asterisk 1.8.x is no longer supported |
| 12:51.11 | jacc0 | SO? why is it so hard to tell your version; I'm not helping you anymore |
| 12:51.20 | singler | tzafrir: soon Sangoma support should be online, you can try waiting in #sangoma |
| 12:51.28 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:c88c:d19d:790c:7f39) |
| 12:51.35 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 12:51.50 | catphish | jacc0: thanks for all your help |
| 12:52.00 | catphish | i will try to file a bug report regarding the timeing lock |
| 12:53.10 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:53.10 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:56.07 | Tuju | i've many clients, but i'm only one person. should i create more accounts or is there a way to keep me as person/account separate from devices and their lines? |
| 12:59.24 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 13:00.17 | GreatSUN | short feature question: |
| 13:00.19 | GreatSUN | is it possible to generate conferences (hold one party, dial another one and make a conference with those) on asterisk server side? |
| 13:00.39 | jacc0 | @tuju: you need more accounts - FollowMe might do what you want |
| 13:00.49 | schmidts | GreatSUN maybe take a look at local channels |
| 13:01.58 | Tuju | jacc0: ack, i look into that FollowMe (sounds something presence related) |
| 13:02.20 | GreatSUN | schmidts: local channels? |
| 13:02.53 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:03.27 | oej | Afternoon |
| 13:04.27 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 13:04.38 | jacc0 | GreatSUN: combining Conference/Meetme, bridge and local channels you should be able to make something you want |
| 13:04.47 | *** join/#asterisk frawd (~francois@23.Red-81-38-28.dynamicIP.rima-tde.net) |
| 13:05.28 | jacc0 | local channel is a channel you can bridge a channel to; it is not some externel line but a context in your extensions.conf |
| 13:07.24 | Katty | hi |
| 13:07.49 | schmidts | GreatSUN you can use a local channel to start a new call leg in your dialplan |
| 13:09.27 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 13:10.14 | jacc0 | hi katty |
| 13:14.24 | beek | Hi Katty |
| 13:17.15 | schmidts | Hi Katty |
| 13:20.36 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 13:25.49 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:32.49 | *** join/#asterisk fim (~fim@planet-algol.com) |
| 13:34.28 | *** join/#asterisk ruyo (~psantos@a213-22-10-69.cpe.netcabo.pt) |
| 13:36.54 | fim | hello. I'm having a weird issue with Asterisk 1.6.2.18.2. When I start it using asterisk -c all modules are loaded successfully but if I start it using the init script, connect using -r and try to load a module (specifically chan_sip), I get "Unable to load module chan_sip.so". Any ideas where to start looking? |
| 13:39.07 | *** join/#asterisk coppice (~chatzilla@116.92.39.71) |
| 13:44.25 | jacc0 | @fim: what command do you use to reload sip? sip reload? |
| 13:44.51 | fim | jacc0: module load chan_sip |
| 13:45.08 | fim | jacc0: sip commands aren't available in the cli since the sip module isn't loaded in the first place |
| 13:45.41 | jacc0 | ok, I have no experiance with not loading sip initialy |
| 13:46.30 | jacc0 | you get the same error if the sip modul eis already loaded |
| 13:47.30 | fim | jacc0: in order to get sip autoloaded do you need to put it in modules.conf? |
| 13:47.43 | jacc0 | yes |
| 13:48.29 | jacc0 | by default it is configured to autoload=yes -> that will load all available modules |
| 13:50.17 | fim | jacc0: I get 0 modules :P |
| 13:56.10 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:56.12 | EmbouNT | trying to start asterisk with /etc/init.d/asterisk start or service asterisk start give me <ASTERISK_ETC_DIR>/asterisk.conf not found. STOP, anyone knows why? In etc/asterisk/asterisk.conf all is OK |
| 13:56.14 | EmbouNT | :( |
| 13:58.28 | *** join/#asterisk skrusty (~skrusty@83.166.171.74) |
| 13:58.49 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:58.52 | *** join/#asterisk lcat (~lcat@187.45.252.108) |
| 14:01.26 | tzafrir | EmbouNT, could you please pastebin your /etc/asterisk/asterisk.conf ? |
| 14:02.09 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:02.18 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:02.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.55 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 14:08.16 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 14:08.26 | BenC[UK] | guys, anyone here know anything about phpagi ? |
| 14:08.30 | *** join/#asterisk shine (~stroll@lamantin.achamo.net) |
| 14:08.46 | BenC[UK] | or probably any agi script... I want to carry on in my code after a dial - is that possible? |
| 14:10.05 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:12.48 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:14.26 | jacc0 | @BenC[UK]: do you mean after pickup or after hangup? |
| 14:15.00 | BenC[UK] | after pickup would be good |
| 14:15.09 | BenC[UK] | just so Ican update the db and say it was answered |
| 14:15.29 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:15.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:16.32 | BenC[UK] | actually, ignore that I can do it another way |
| 14:16.52 | jacc0 | :) |
| 14:17.26 | jacc0 | it is in the CDR records |
| 14:18.03 | jacc0 | it is already the database if you enable cdr mysql |
| 14:20.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:21.47 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 14:22.56 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 14:23.04 | krion | hi |
| 14:23.11 | jacc0 | hi krion |
| 14:23.26 | krion | i'm having a trouble with voicemail recording |
| 14:24.01 | krion | my rtp flux is fine (using wireshark for troubleshoot), but when the voicemail is recorder to wav, there is something wrong |
| 14:24.13 | krion | the message is in a sort of fast "fast forward" |
| 14:24.37 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:24.43 | krion | it's weird... try to reload app_voicemail.so with no luck |
| 14:24.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:26.48 | EmbouNT | how i can test in the cli if the mysql module is up? |
| 14:26.51 | krion | i got some channel.c: Scheduling timer at 138 sample intervals and channel.c: Scheduling timer at 0 sample intervals |
| 14:27.05 | krion | but not sure it's related |
| 14:29.29 | schmidts | krion do you save your voicemails over an nfs link? |
| 14:29.56 | schmidts | if yes you have to start asterisk with a special option, dont know which one now, to save the voicemail localy and then move it to the final directory |
| 14:30.04 | schmidts | i also had this once when i save my voicemails over nfs |
| 14:31.08 | schmidts | krion its the -t option: Record soundfiles in /var/tmp and move them where they belong after they are done. |
| 14:32.54 | krion | yes nfs links |
| 14:33.08 | schmidts | start asterisk with the -t option and everything will be fine |
| 14:34.35 | krion | schmidts: i already recording into a tmp but in nfs |
| 14:35.17 | krion | and the trouble didn't appear before (like first august) as the settings where the same |
| 14:37.57 | schmidts | krion maybe your nfs is getting slower by some cause |
| 14:39.41 | Tuju | argh, this java phone is real pita. |
| 14:39.48 | Tuju | really hard to get the config right. |
| 14:41.13 | krion | schmidts: not sure... i've two other asterisk and they recording fine |
| 14:41.30 | krion | i'll try to rester the asterisk process tonight, hope this will fixe it |
| 14:41.32 | krion | fix |
| 14:41.41 | schmidts | krion i can only tell you that i had just the same problem and with the -t option it was gone |
| 14:41.52 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:41.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:41.55 | Tuju | now i get it into 'Registring...' state but sniffer doesn't show any traffic anyway. |
| 14:42.15 | krion | schmidts: ok thanks a lot for the hint i'll try |
| 14:42.26 | schmidts | your welcome ;) |
| 14:42.53 | krion | but i have to wait tongiht, damn you user who's calling in august ! get holiday ! |
| 14:43.02 | schmidts | LOL |
| 14:52.35 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:53.52 | catphish | i'm impressed, asterisk on my core2quad can handle 1150 calls in a very simple test |
| 14:54.02 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 14:54.54 | aberrios | Just to check I'm not going mad, or confirm that I am. I'm having a mini argument with a provider. I said I wanted a SIP Trunk with 10 channels. He said "you can't have 10 channels on a trunk, you can have 1 trunk with 10 extensions".... Would a provider call the channels on a trunk 'extensions'? |
| 14:55.20 | catphish | people use all kinds of words |
| 14:55.21 | _Corey_ | catphish: What kind of test are you using to get that number? |
| 14:55.24 | catphish | why not keep it simple |
| 14:55.45 | catphish | use 'concurrent calls (incoming / outgoing)' and 'numbers' |
| 14:55.49 | _Corey_ | aberrios: Say "call paths", may help |
| 14:56.24 | aberrios | I'm just hoping they don't send me 10 different accounts to setup... |
| 14:57.07 | catphish | _Corey_: using sipp to dial Milliwatt, looping back the RTP data for 30 seconds |
| 14:57.31 | leifmadsen | aberrios: they might depending on how their accounts/system are setup |
| 14:57.41 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 14:57.43 | catphish | 1000 seems like a pretty standard number of calls people have reported on quad core systems |
| 14:57.47 | leifmadsen | aberrios: although ya, use the path of least resistance and say "concurrent calls" |
| 14:58.05 | krion | schmidts: but you're right anyway, i ln the tmp dir in my voicemail nfs share to the /tmp on my asterisk host and the problem is gone |
| 14:58.17 | krion | the thing is it was working earlier... with the same setup |
| 14:58.21 | catphish | i'm about to provision a host with 2 x 6-core opterons, looking forward to seeing the call numbers on that |
| 14:58.25 | krion | i'll try to umount remount the nfs |
| 14:58.47 | _Corey_ | catphish: Hmmm, in my experience there are other factors (i/o, etc.) but that's cool |
| 14:59.19 | catphish | _Corey_: i'd expect network interrupts to become an issue at that point |
| 14:59.29 | catphish | that's 100Mbit in each direction |
| 14:59.47 | catphish | but basically at 1150 calls, asterisk maxes out the core2quad |
| 15:00.00 | catphish | network drivers will have an impact |
| 15:00.11 | catphish | disk IO and RAM are totally unused |
| 15:00.19 | _Corey_ | catphish: Usually we reach the upper limit on a system before that happens... audio gets intermittently lost on the channels (asterisk bridging in this scenario) etc |
| 15:01.08 | catphish | well i did a few tests, i discovered that my SRX240 firewall started dropping packets before asterisk did |
| 15:01.14 | jacc0 | I think i've fond the source of a memory leak: inotify_daemon (data=0x0) at stdtime/localtime.c |
| 15:01.29 | _Corey_ | catphish: Yeah, that's a lot of bandwidth :) |
| 15:01.33 | catphish | but in the end CPU was the only limit |
| 15:01.34 | jacc0 | see this gdb (part): http://pastebin.com/z7Ayayhh |
| 15:02.03 | jacc0 | it seems to be taking up more and more memory over time |
| 15:02.04 | catphish | well 100Mbit * 2 isn't that much bandwidth really, but initiating 80 new UDP streams a second probably started to stress the firewall |
| 15:02.48 | catphish | either that or the small UDP packets were just too much for it |
| 15:03.01 | catphish | either way, i won't be using it behind a stateful firewall |
| 15:03.06 | catphish | just a nice ACL |
| 15:03.10 | *** join/#asterisk Yedidya (~chatzilla@host86-137-84-71.range86-137.btcentralplus.com) |
| 15:05.23 | Yedidya | HELP! need to have a call join a confbrige AND have some white noise played to the outbound leg of THIS call only. Any ideas? (I'm otherwise a pro with extentions.conf and can manage php (for agi)). |
| 15:05.24 | jacc0 | don't use asterisk internel timer or you will run out of memory or end up in a deadlock! |
| 15:05.55 | jacc0 | I'll report this tomorrow; office hours are over for today :) |
| 15:06.00 | catphish | jacc0: I think that's the moral of today's story, yes |
| 15:06.06 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:07.01 | Qwell | jacc0: what makes you think that the inotify stuff is taking up memory? |
| 15:07.15 | catphish | i wonder how pthreads compares to dadhi timing |
| 15:07.32 | catphish | since i only have dummy driver running |
| 15:08.31 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:08.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:08.47 | The_Boy_Wonder | catphish: pthreads is not an efficient timing source. it works, but is expensive |
| 15:08.53 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:09.01 | catphish | i'll carry on using ztdummy then :) |
| 15:09.07 | catphish | works nicely |
| 15:09.24 | jacc0 | The_Boy_Wonder@ it takes up more and more memory over time |
| 15:09.44 | The_Boy_Wonder | really?! umm. is there a bug report open for that? |
| 15:09.54 | jacc0 | <PROTECTED> |
| 15:10.00 | jacc0 | <PROTECTED> |
| 15:10.10 | jacc0 | I will file a bug report tomorrow |
| 15:10.15 | Kobaz | so there's a memory leak in timing pthreads? |
| 15:10.22 | jacc0 | office hours are over for me |
| 15:10.29 | catphish | the pthread timer causes a deadlock too? |
| 15:10.37 | catphish | :| |
| 15:10.47 | The_Boy_Wonder | catphish: i have not heard of a deadlock occurring in pthread timing |
| 15:10.59 | catphish | i think that's what jacc is talking about |
| 15:11.03 | The_Boy_Wonder | the timerfd one is well known |
| 15:11.33 | catphish | yeah, i ran into the timerfd one today, even in 10.0 trunk, but happy to use dadhi instead |
| 15:11.42 | jacc0 | okay, pthreads is not part of timerfd? |
| 15:11.58 | Kobaz | no |
| 15:12.00 | catphish | no, there's 4 timer sources |
| 15:12.08 | jacc0 | okay |
| 15:12.10 | catphish | dadhi, pthreads, fd and something else |
| 15:12.20 | Kobaz | on a production system you shouldn't use anything other than dahdi |
| 15:12.29 | catphish | fd sadly seems to be the default and is hard not to load without deleting / not compiling it |
| 15:12.46 | Kobaz | noload => res_timing_timerfd.so |
| 15:12.49 | Kobaz | in modules.conf |
| 15:12.53 | catphish | Kobaz: that doesn't work |
| 15:13.00 | catphish | at least for me, it loaded anyway |
| 15:13.01 | Kobaz | then you typed it wrong |
| 15:13.13 | catphish | ...testing again |
| 15:13.14 | jacc0 | hehehe |
| 15:13.15 | jacc0 | bye all |
| 15:13.24 | Kobaz | or you have a conflicting option... like load => of the module, and then a noload |
| 15:13.34 | catphish | i have autoload on |
| 15:13.36 | catphish | then noload |
| 15:13.38 | Kobaz | yeah that's fine |
| 15:13.41 | Kobaz | that's what i do |
| 15:13.50 | catphish | let me compile it and see |
| 15:15.29 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:17.33 | anonymouz666 | anything else to make realtime queue_log to work? odbc status OK (connected), table sourced from contrib, nothing being insert into queue_log table. |
| 15:19.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:21.06 | catphish | Kobaz: i can't reproduce the problem of not being able to disable the timerfd module now so maybe i was being stupid |
| 15:21.20 | catphish | i originally assumed that another module was calling it overriding my noload |
| 15:22.51 | anonymouz666 | fixed. |
| 15:22.58 | anonymouz666 | there was a class not defined |
| 15:24.11 | anonymouz666 | leifmadsen: migrating a 400 seats callcenter from version 1.4 to 1.8 and hopping for the best ;) |
| 15:24.35 | leifmadsen | anonymouz666: I'd suggest hoping instead of hopping, but that's just me :) |
| 15:25.16 | anonymouz666 | distributed device state is something that I can't live without it |
| 15:25.23 | leifmadsen | ya it's pretty amazing |
| 15:25.28 | aberrios | "'faith, hop and charity and the greatest of these is 'hop'" |
| 15:25.51 | anonymouz666 | leifmadsen: any problem running the XMPP even for low latency links? |
| 15:26.03 | leifmadsen | AIS works well (and is easier to get setup because it requires no external server) if you're using it in a LAN environment; XMPP works well too and allows WAN interconnectivity but is more difficult as you have to setup an external service |
| 15:26.23 | leifmadsen | anonymouz666: I think you have it inversed in your mind, as low latency is ideal |
| 15:26.32 | leifmadsen | there would be no problems having better connectivity with XMPP :) |
| 15:26.40 | Qwell | leifmadsen: blasphemy |
| 15:26.44 | leifmadsen | Qwell: nub |
| 15:26.50 | Qwell | packets would arrive too early |
| 15:27.01 | leifmadsen | Qwell: perhaps even in the past |
| 15:27.07 | beek | or before they were sent |
| 15:27.38 | anonymouz666 | heh |
| 15:30.32 | *** join/#asterisk irroot (~irroot@197.108.225.246) |
| 15:31.01 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 15:36.05 | p3nguin | I'm sure this has been discussed over and over, and I'm not trying to beat a dead horse, but is Asterisk 10 the equivalent of 1.10 or 2.0? I'm only looking for a simple authoritative answer, not a debate. |
| 15:36.18 | beek | yes |
| 15:36.26 | anonymouz666 | 1.10 |
| 15:36.35 | beek | since there's not to be a 2.0 |
| 15:36.37 | Qwell | It is equivalent to Asterisk 10. |
| 15:36.46 | anonymouz666 | Qwell: you troll a lot :P |
| 15:36.48 | Qwell | http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 15:37.03 | beek | Was there pot involved? |
| 15:37.04 | Qwell | Who's trolling? The premise of the question is flawed. |
| 15:37.23 | p3nguin | I'm sure it is, but I don't know any other way to word it to get the answer I'm trying to get. |
| 15:37.27 | Gugge | its equivelent to the version after 1.8 :) |
| 15:37.34 | Qwell | Gugge: Exactly. |
| 15:37.43 | Qwell | The point is, it really just doesn't matter. |
| 15:37.44 | Gugge | nothing else |
| 15:37.47 | p3nguin | gugge: That's kind of my point. What's after 1.8? 1.10 or 2.0? |
| 15:37.53 | Gugge | p3nguin: 10.0 is |
| 15:37.54 | Qwell | 10 is after 1.8 |
| 15:37.55 | anonymouz666 | Asterisk 2.0 will be rewritten totally in PHP with hiphop from facebook. :P |
| 15:37.59 | Gugge | in asterisk version numbers |
| 15:38.31 | Gugge | in other software it could be 1.9, 1.81, 1.10, 2.0, or whatever they like :) |
| 15:39.20 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 15:40.12 | Yedidya | PLease peeps, as you seem to be here, HELP! need to have a call join a confbrige AND have some white noise played to the outbound leg of THIS call only. Any ideas? (I'm otherwise a pro with extentions.conf and can manage php (for agi)). |
| 15:41.40 | catphish | does asterisk (sip channels) have any intelligence to end calls where the remote end simply disappears? |
| 15:41.52 | Qwell | catphish: There's an RTP timeout option. |
| 15:42.12 | catphish | that would be ideal |
| 15:43.27 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 15:43.41 | p3nguin | So, according to the blog post, 10 was going to be 1.10. You could have just said that when I asked. |
| 15:44.37 | catphish | 1.2 => 1.4 => 1.6 => 1.8 => 10.0 |
| 15:44.44 | catphish | isn't that obvious? |
| 15:44.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:44.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:45.30 | chazzam | except it isn't 1.6 |
| 15:45.51 | Qwell | 1.2 > 1.4 > 1.6.0 > 1.6.1 > 1.6.2 > 1.8 > 10 |
| 15:45.53 | chazzam | it was 1.2 => 1.4 => 1.6.0 => 1.6.1 => 1.6.2 => 1.8 => 10 |
| 15:45.55 | chazzam | yeah |
| 15:45.58 | chazzam | that |
| 15:46.19 | catphish | oh well, obviously! |
| 15:46.27 | catphish | :) |
| 15:46.29 | Gugge | It should have been AsteriskX |
| 15:46.34 | Qwell | no |
| 15:46.35 | Gugge | no one would questien that :) |
| 15:46.36 | irroot | you missed 0.99 1.0 :P |
| 15:46.45 | p3nguin | Or Asterisk X, like Mac OS X. |
| 15:47.01 | chazzam | as some have mention 10 is 2 in binary |
| 15:47.13 | Gugge | Stupid names isnt a problem, number apparently are :) |
| 15:47.19 | anonymouz666 | XMPP Tigase sux to setup. :~ |
| 15:47.31 | leifmadsen | Puppet just went from 0.25.x to 2.6.x, so who cares about version numbers? |
| 15:47.36 | catphish | 10 is 2 in binary, but since none of that other revisions are in binary i'm not buying it :) |
| 15:47.42 | p3nguin | If you read the blog post, they've decided to drop the 1. prefix. That makes 1.10 turn into 10. |
| 15:47.45 | Gugge | leifmadsen: a lot of people it seems :) |
| 15:47.46 | leifmadsen | anonymouz666: that's why I said AIS was easier :) |
| 15:47.50 | Qwell | leifmadsen: I bet they feel like idiots now that Linux is 3.0 |
| 15:47.55 | leifmadsen | Gugge: a lot of people have nothing better to do |
| 15:48.09 | p3nguin | So all that needed to be said when I asked was that 10 is what 1.10 would have been. |
| 15:48.28 | leifmadsen | p3nguin: yes I didn't see the question :) |
| 15:48.33 | leifmadsen | 10 == 1.10 with 1. missing |
| 15:48.34 | catphish | i'd have said that, but i wasn't listening, so meh |
| 15:48.47 | leifmadsen | it's really just that simple |
| 15:48.50 | tzanger | so that means we're going to see asterisk 10, 11, 12, 13? |
| 15:48.59 | Gugge | or 1.8 is what 1.6.3 whould have been, and 10 is what 1.6.4 would have been? :P |
| 15:49.00 | leifmadsen | tzanger: exactly |
| 15:49.01 | Qwell | tzanger: yes |
| 15:49.07 | leifmadsen | Gugge: yes :) |
| 15:49.11 | leifmadsen | Gugge: someone who gets it :D |
| 15:49.38 | p3nguin | The only reason I was asking was so I had an authoritative answer for when the question arises later, in the absence of the authority on the matter. See: yesterday. |
| 15:49.40 | Gugge | its just a number :) |
| 15:49.58 | tzanger | hm, I'm still running 1.4.23.1 |
| 15:50.02 | aberrios | Its not a number its a free software pbx! |
| 15:50.05 | leifmadsen | p3nguin: yep, more information / authoritative answer available in the Kevin P. Fleming post on blogs.digium.com |
| 15:50.29 | leifmadsen | p3nguin: you can point that to people who ask you about version numbers as then you don't have to keep repeating yourself :) |
| 15:50.32 | leifmadsen | ~asterisk10 |
| 15:50.32 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 15:50.34 | _Corey_ | aberrios: Nice prisoner reference... |
| 15:51.04 | p3nguin | or insurance commercial reference. |
| 15:51.07 | p3nguin | Nationwide is on your side. |
| 15:51.18 | p3nguin | NationPam is on your s...am. |
| 15:51.44 | tzanger | jesus 2004 was when asterisk 1.0 came out... it does not feel like that long ago |
| 15:52.06 | Qwell | Only 2004? Feels like 1980. |
| 15:54.23 | tzanger | heh I like the comments... "Asterisk XP" haha |
| 15:54.29 | tzanger | let's hope we don't see an Asterisk ME |
| 15:54.44 | Gugge | Or Asterisk Vista |
| 15:54.53 | catphish | awesome |
| 15:57.37 | anonymouz666 | leifmadsen: did you install openais using the tarball? |
| 15:57.58 | p3nguin | Hey, now... Asterisk Me would be awesome! |
| 15:58.23 | anonymouz666 | res_ais is not being recognized by menu select after the make install |
| 15:59.19 | anonymouz666 | damn |
| 15:59.30 | anonymouz666 | needs to stop being so smart |
| 15:59.42 | anonymouz666 | installed the server in one machine, and trying to find it in another |
| 16:02.25 | leifmadsen | anonymouz666: :) |
| 16:02.57 | leifmadsen | Learn Asterisk Me at Asterisk U! |
| 16:02.59 | *** join/#asterisk frawd (~francois@23.Red-81-38-28.dynamicIP.rima-tde.net) |
| 16:03.09 | leifmadsen | (that is not a real tag line) |
| 16:05.41 | p3nguin | It is now! HAHAHAHA!!!!!! |
| 16:06.54 | p3nguin | is still looking for qwell on g+ |
| 16:07.02 | Qwell | You'll never find me. |
| 16:07.13 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 16:07.35 | p3nguin | Don't be skeerd of it. |
| 16:12.26 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
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| 16:39.21 | *** join/#asterisk kraptv (ryan@mail.skylab.org) |
| 16:39.39 | kraptv | I _love_ Asterisk and am looking forward to 10! |
| 16:40.34 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 16:41.04 | *** join/#asterisk Hive (~Adam@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 16:41.52 | chazzam | Qwell: g+ has cookies... |
| 16:47.44 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 16:48.49 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.39) |
| 16:52.20 | Yedidya | How do i neject sound into 1 leg of a call? |
| 16:52.37 | Yedidya | typo# How do i inject sound into 1 leg of a call? |
| 16:52.47 | p3nguin | ChanSpy() will do it. |
| 16:52.57 | p3nguin | Use the whisper mode. |
| 16:53.16 | WIMPy | If the sound comes from a channel. |
| 16:54.15 | anonymouz666 | p3nguin: things start to become nice when you have to inject into callee and caller, make something periodic, etc. |
| 16:54.47 | WIMPy | anonymouz666: You found a way to do that? |
| 16:55.11 | anonymouz666 | of course there is a way, but it's not a trivial task. |
| 16:55.45 | anonymouz666 | russell made a proposal in devel list, but nobody ever implement the idea |
| 16:56.04 | anonymouz666 | the cookbook has something in that way, did you see it? |
| 16:56.28 | WIMPy | So, not for "users". |
| 16:56.33 | WIMPy | Nope |
| 16:57.04 | WIMPy | We tried here ast week and found out that you can't pair dilapan apps like Playback with ChanSpy. |
| 16:57.11 | WIMPy | last |
| 16:57.41 | *** join/#asterisk bmint (~bmint@h174.92.190.173.static.ip.windstream.net) |
| 16:58.15 | bmint | Is there an agi command to put all channel variables for a call into an array? |
| 16:58.45 | WIMPy | DumpChan? |
| 16:58.47 | kraptv | Does Asterisk still reply on DADHI for timing and other channel functions or is it redone with its own software timer? (i.e. no zaptel/dahdi drivers for OS X - what to do then) |
| 16:58.58 | Kobaz | dumpchan just prints to the console |
| 16:59.01 | bmint | yes like dumpchan but how do I use that in phpagi |
| 16:59.23 | WIMPy | It doesn't go to the script? Hmm. bad luck :-( |
| 16:59.25 | Kobaz | bmint: there isn't one |
| 16:59.44 | WIMPy | kraptv: Partially. You don;t need dahdi timing, but MeetMe() and Page() need it. |
| 17:00.14 | bmint | So the only variables I can use are variables from the return array? |
| 17:01.38 | kraptv | Well, does ChanSpy work? I imagine not as it is very zap* focused. |
| 17:02.10 | bmint | Sorry request array? |
| 17:02.17 | WIMPy | kraptv: Yes, that works. |
| 17:02.30 | kraptv | Wow, cool. thanks, WIMPy! |
| 17:02.40 | Bipul | p3nguin, hi |
| 17:07.23 | p3nguin | <PROTECTED> |
| 17:07.29 | p3nguin | dammit |
| 17:07.35 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 17:07.52 | p3nguin | bipul: Hi. |
| 17:08.30 | Bipul | p3nguin, yes i win the presentation ;) |
| 17:08.37 | p3nguin | <PROTECTED> |
| 17:08.43 | p3nguin | Grr, did it again! |
| 17:08.44 | Bipul | that's what i want's to say :D |
| 17:09.15 | catphish | is there any advantage to using digium hardware on a purely aip asterisk host |
| 17:09.19 | p3nguin | I thought it was just a presentation, not a competition. |
| 17:09.23 | Bipul | But i having issue with outgoing voice |
| 17:09.33 | Bipul | yes i got the best presentation |
| 17:09.50 | Bipul | and my faculty told me to dig more on Asterisk |
| 17:10.00 | Bipul | it will helpful for you career alot. |
| 17:10.03 | p3nguin | Are you in secondary school? |
| 17:10.19 | Bipul | Nops Engineering college. |
| 17:10.28 | p3nguin | Oh, post-secondary. |
| 17:10.45 | p3nguin | I didn't know. |
| 17:10.52 | Bipul | Engineering ( University"). |
| 17:10.56 | p3nguin | yeah |
| 17:11.22 | Bipul | p3nguin, so Thank's once again.. |
| 17:11.41 | p3nguin | I'm happy that you did well in the presentation. |
| 17:12.34 | Bipul | Yes Now every one know about a2infotech.com |
| 17:12.56 | Bipul | and also Asterisk technology specially our profesors he would like to work with me |
| 17:13.03 | p3nguin | I guess that's good. |
| 17:13.22 | Bipul | It's not Good it's awsome. |
| 17:13.37 | *** join/#asterisk _jem_ (~jem@wikimedia/-jem-) |
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| 17:25.36 | Tuju | http://www.888voip.com/configuring-cisco-7975-ip-phones-for-sip/ I cannot begin to stress how picky these phones are with their configuration files. |
| 17:26.03 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 17:26.27 | Kobaz | oh, yes |
| 17:30.44 | *** join/#asterisk Sertys (~cwalker@hotel-palas.com) |
| 17:31.22 | leifmadsen | Tuju: I sold my 7970 years ago because of how picky the configuration was |
| 17:31.37 | Tuju | well, i've four of them here. :-( |
| 17:31.49 | Tuju | and i already got rid of those 7960's |
| 17:31.52 | p3nguin | I'll stick to my 7960 for now. |
| 17:32.11 | Tuju | you can even telnet inside those and hack them like ios. |
| 17:32.40 | Tuju | we should make python class for this config and serialize it. |
| 17:38.51 | Sertys | lol |
| 17:39.21 | Tuju | leifmadsen: did you ever get it working, at least once? |
| 17:39.38 | Tuju | and if you did, do you have any configs laying around somewhere? |
| 17:39.42 | leifmadsen | Tuju: kind of, but that was a long time ago |
| 17:39.49 | leifmadsen | I definitely don't :) |
| 17:40.07 | Tuju | I got it working once, but then changed the config and - well, here I am now. |
| 17:40.20 | leifmadsen | yep, it's so incredibly picky I gave up after days of trying |
| 17:40.44 | leifmadsen | sold the phone, bought polycoms, and decided I saved $1000 doing that (assuming my time was worth > $0) |
| 17:41.24 | Tuju | one huge issue was that they changed protocol from udp to tcp in one release upgrade |
| 17:41.46 | coppice | cisco phones are great for their intended use - land fill. |
| 17:41.49 | Tuju | you could have gone around it by adding <transportLayerProtocol>4</transportLayerProtocol> into config - just not sure does it have to be 2 or 4 for udp |
| 17:42.36 | Tuju | i kind of got bit chill feelings once noticed that it has java inside it |
| 17:42.52 | Tuju | as i typically stay *far* away that crap. |
| 17:43.07 | Tuju | then again, having same VM and code it should run, right+ |
| 17:43.33 | Tuju | but i guess java's problems are more associated with the reasons why it got selected, not as a technology. |
| 17:44.02 | Tuju | you got bunch of idiots coding so you need easier language, hence this is so picky with config files. |
| 17:44.36 | Tuju | they could have dropped the X from xml, as it certainly is not eXtensible. :) |
| 17:45.03 | raden | Katty, :D :D :D :D :D :D :D |
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| 17:51.51 | leifmadsen | Tuju: generally the reason those phones don't have configuration files that are well documented is, as I understand it anyways, that the configurations in those phones are now generated from CCM directly and not created by the administrator by hand |
| 17:52.40 | Tuju | leifmadsen: i missed your point |
| 17:52.48 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 17:52.50 | leifmadsen | point is, good luck! |
| 17:53.06 | Tuju | i'm afraid that's not enough this time. |
| 17:53.23 | leifmadsen | then you'll probably want to find someone who has a CCM who can generate you some configs for those phones |
| 17:53.40 | leifmadsen | I literally spent 3 days trying to make a 7970 work and failed |
| 17:53.45 | Tuju | that doesn't sound like me :) |
| 17:53.59 | Tuju | i got it register once already. |
| 17:54.15 | Tuju | just was stupid enough to change the config until i took a copy of it |
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| 18:01.45 | jpcansa | is there any way that Monitor() will remove non-accepted chararcters when creating a .wav file?? |
| 18:03.33 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:03.45 | sunfone | Does anyone know if Polycom (or any 802.3af phone) will work with an SMI POE switch? |
| 18:04.23 | p3nguin | |
| 18:08.28 | Tuju | ha! i can see sip traffic now! |
| 18:10.47 | *** part/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
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| 18:13.34 | stevekstevek | Hola Asterisk Room. |
| 18:14.19 | stevekstevek | I haven't been here in a _very_ long time -- and I haven't been involved with asterisk development for a really long time.. |
| 18:15.38 | stevekstevek | Anyway, my question is: I need to build a pretty scalable conference bridge, and if I were to do that these days, which conference app would be the best in terms of scalability: app_meetme, app_confbridge (using new conference infrastructure), or app_conference (which I actually did help develop ages ago). |
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| 18:18.08 | jeffspeff | has anybody tried setting up a polycom cx600 to work with asterisk? |
| 18:19.08 | _Corey_ | jeffspeff: That's the Microsoft model |
| 18:19.19 | pabelanger | stevekstevek: confbridge was rewritten for Asterisk 10 |
| 18:19.49 | jeffspeff | _Corey_, i was hoping to be able to use some of the lync features, but still use * as the pbx backend |
| 18:19.51 | stevekstevek | pabelanger: yeah, I read that -- but I think that was more the front-end and configuration stuff rather than the core bridging code. |
| 18:21.33 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 18:21.53 | *** join/#asterisk x1user (~chatzilla@95.43.28.77) |
| 18:22.43 | pabelanger | stevekstevek: no, it was a rewrite of the code. New media, better performance, etc. |
| 18:22.48 | pabelanger | confbridge > meetme |
| 18:23.20 | stevekstevek | pabelanger: hmm, I though that was for 1.8, and for 10.0 the configuration stuff was changed. |
| 18:23.35 | pabelanger | nope, 10 |
| 18:24.05 | stevekstevek | basically, that app_confbridge was a thin/configuration front end to enhanced bridging capabilities, in more-or-less the same way that app_meetme was a front-end to the conferencing engine inside of zaptel/dahdi.. |
| 18:24.42 | leifmadsen | stevekstevek: ConfBridge in 1.8 vs 10 is very different. In 1.8 it's only a front-end to the bridging interface basically |
| 18:24.53 | leifmadsen | ConfBridge() in 10 is significantly enhanced |
| 18:25.00 | leifmadsen | (not even really the same thing at all anymore) |
| 18:25.10 | stevekstevek | right -- but does it still uses the internal bridging engine? |
| 18:25.24 | leifmadsen | it uses the bridging modules |
| 18:25.34 | leifmadsen | which are also relatively new |
| 18:26.01 | stevekstevek | so -- have people benchmarked these things anywhere? |
| 18:26.05 | pabelanger | https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 18:26.56 | stevekstevek | way back when I wrote app_conference, I did it because app_meetme has O(n^2) algorithms, and scaled really poorly. I see that, at some point there was a thing added they called "talker optimization", which probably improved things a bit. |
| 18:27.29 | leifmadsen | stevekstevek: ya ConfBridge() and MeetMe() are not built in the same way a all |
| 18:27.47 | stevekstevek | I know I can build a bridge for hundreds of callers in a single conference with app_conference -- I assume that the internal bridging modules are at least as efficient.. |
| 18:27.51 | leifmadsen | I've done some testing and gotten about 10x better performance with ConfBridge() over MeetMe() |
| 18:28.24 | leifmadsen | well I wouldn't refer to them as "internal" because the bridging modules are exposed as modules under the Bridging Modules section of menuselect |
| 18:28.26 | stevekstevek | awesome: what kinds of numbers have you been able to test to? |
| 18:28.32 | leifmadsen | (to me, internals seems to mean "hidden") |
| 18:28.45 | leifmadsen | stevekstevek: I was doing something like 200 channels or something I think |
| 18:28.52 | leifmadsen | and the box wasn't super powerful |
| 18:28.59 | stevekstevek | and it broke at those numbers? |
| 18:29.06 | stevekstevek | or you stopped testing more. |
| 18:29.14 | leifmadsen | I stopped testing |
| 18:29.26 | stevekstevek | 'cause I know I've tested to 800 or so, with app_conference, 5 years ago.. |
| 18:29.29 | stevekstevek | ok. |
| 18:29.37 | stevekstevek | so, you'd definitely recommend starting that way.. |
| 18:30.21 | pabelanger | stevekstevek: yes, I would test with Asterisk 10 beta1, and report results. I'm sure the developers would be interested in seeing them |
| 18:30.29 | leifmadsen | oh ya for sure, it's not at all the same thing as MeetMe() or done in the same way at all |
| 18:30.42 | leifmadsen | definitely look at Asterisk 10 though, that's where we were doing all our testing |
| 18:30.51 | leifmadsen | we spent a good 2-3 weeks testing it internally |
| 18:31.05 | stevekstevek | I will. |
| 18:31.13 | stevekstevek | It will probably work into my schedule if I do that. |
| 18:31.15 | leifmadsen | (well longer than that, but 2-3 weeks intensely) |
| 18:31.17 | leifmadsen | nice |
| 18:31.21 | beek | leifmadsen: You're going to need to do another revision of the book for 10! |
| 18:31.26 | leifmadsen | nah |
| 18:31.31 | leifmadsen | we'll see how well it sells for now |
| 18:31.33 | stevekstevek | BTW: I'm really glad to see what y'all have done with the bridging interfaces, and now reading the conference apps. |
| 18:31.36 | leifmadsen | we tend to focus on LTS |
| 18:32.00 | stevekstevek | This is all stuff I remember talking about at the first astricon, and even before that.. |
| 18:32.15 | stevekstevek | from way-back before the different asterisk forks forked off :) |
| 18:32.22 | anonymouz666 | leifmadsen: between two boxes, do you use IAX2 or SIP? |
| 18:32.30 | leifmadsen | SIP always |
| 18:32.32 | leifmadsen | I never use IAX2 |
| 18:32.38 | anonymouz666 | distributed device state working fine |
| 18:32.40 | anonymouz666 | it is amazing |
| 18:32.42 | leifmadsen | :) |
| 18:32.42 | leifmadsen | yep |
| 18:32.51 | anonymouz666 | openais it is really EASY |
| 18:33.11 | voxter | I may just convert all my IAX peers to SIP this year |
| 18:33.20 | voxter | for no other reason other than to be able to use 3rd party call reporting tools |
| 18:33.36 | voxter | and to avoid transcoding when handing off to non IAX peers. |
| 18:33.37 | leifmadsen | anonymouz666: yep :) |
| 18:33.42 | voxter | its worked gloriously for me internally |
| 18:34.11 | stevekstevek | leifmadsen: pabelanger: Thanks, gentlemen! |
| 18:34.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v008-062.mobile.uci.edu) |
| 18:34.34 | anonymouz666 | my problem is I have in both machines the SIP peers |
| 18:34.42 | anonymouz666 | so I can register in one box and in another |
| 18:34.54 | anonymouz666 | that makes impossible to dial using sip from one to another |
| 18:35.08 | anonymouz666 | cause the nature of chan_sip it matches the friend first |
| 18:35.36 | leifmadsen | that's why you need to split the friends into peers and users |
| 18:36.44 | anonymouz666 | ooh |
| 18:36.48 | anonymouz666 | gonna test this now |
| 18:44.16 | voxter | leifmadsen: thats actually suggested? Using peers and users? |
| 18:44.28 | *** join/#asterisk JokerMx (~JokerMx@200.71.215.104) |
| 18:44.32 | voxter | I have a perfectly working friend setup, maybe i have some unknown limitation? |
| 18:44.33 | leifmadsen | it is if you need to carefully control matching |
| 18:44.44 | leifmadsen | if you don't have issue with matching, then using friends is fine |
| 18:44.54 | leifmadsen | it's only if you have specific issues with matching on specific peers |
| 18:45.10 | JokerMx | hola |
| 18:45.11 | leifmadsen | breaking out to users and peers seems to be necessary to match on username vs IP |
| 18:45.22 | voxter | nod |
| 18:45.27 | JokerMx | necesito ayuda con asterisk |
| 18:45.28 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 18:45.30 | voxter | Ive had issues with that in the past |
| 18:45.41 | voxter | specifically when i have two peers to one place, and one is locked to ulaw only, and one is locked to g729 only |
| 18:45.53 | leifmadsen | JokerMx: English please, or see #asterisk-br |
| 18:46.05 | voxter | since codec 'selection' amongst a list of multiple available codecs doesn't exactly "work" in asterisk |
| 18:46.14 | voxter | at least it didn't in the version of 1.4 i was running when i did that. |
| 18:46.37 | leifmadsen | and it probably still doesn't -- that's a long standing issue that requires some core changes |
| 18:46.51 | voxter | yeah. transcoded to the first in the list regardless of if you wanted to use it not |
| 18:46.55 | voxter | that was a surprise on my cpu. :) |
| 18:46.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:48.55 | JokerMx | I need to handle messages that I deliver advertising (E1/PRI) |
| 18:49.16 | anonymouz666 | leifmadsen: sounds like spanish not portuguese for #asterisk-br :) |
| 18:49.27 | leifmadsen | anonymouz666: ya I think that's what I meant to point at ;) |
| 18:51.37 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 18:53.57 | JokerMx | I need to handle messages that I deliver advertising (E1/PRI) |
| 18:54.31 | WIMPy | JokerMx: Maybe you should try to re-phrase that. |
| 18:55.16 | Katty | i have a scenario i need help with. |
| 18:55.24 | Katty | location a has an asterisk server. |
| 18:55.31 | Katty | location b, is a remote user...at home. |
| 18:55.42 | Katty | usually opening ports and forwarding stuff works fine |
| 18:55.43 | WIMPy | One that doesn't work at all? |
| 18:56.00 | Katty | so location a to locationb is super. |
| 18:56.09 | Katty | now location c wants in |
| 18:56.13 | Katty | location c has 2 phones. |
| 18:56.33 | Katty | and you can't forward a single port to two internal LAN devices. |
| 18:56.48 | Katty | assuming they won't pay for a vpn... |
| 18:57.03 | Katty | how do you get two remote phones on a single lan, back to the main location |
| 18:58.09 | WIMPy | That obviousely depends on the router at loc c. |
| 18:58.34 | Katty | elaborate. |
| 18:58.45 | Katty | you cannot forward a single port to two individual, internal, ip addresses |
| 18:58.59 | WIMPy | Correct. |
| 18:59.04 | Katty | so how do you do it |
| 18:59.12 | WIMPy | But maybe you don't have to. |
| 18:59.21 | WIMPy | That obviousely depends on the router at loc c. |
| 18:59.31 | Katty | elaborate |
| 18:59.40 | WIMPy | And in the worst case on the phones as well. |
| 18:59.58 | Katty | could you be more specific |
| 19:00.03 | *** join/#asterisk REdOG (~REdOG@gentoo/user/redog) |
| 19:00.19 | WIMPy | With routers that have connection tracking (e.g. linux) you shouldn't need to forward anything. |
| 19:00.45 | WIMPy | If you need to do static forwarding, you have to use different ports. But not only for SIP, but also for RTP. |
| 19:00.55 | WIMPy | That's where the phones config comes in. |
| 19:01.14 | WIMPy | And then there is everything in between, off course. |
| 19:01.16 | Katty | i imagine i will have to use the equipment that the client already has. |
| 19:01.20 | Katty | cheap, linksys routers |
| 19:01.30 | Katty | the kind you buy from staples for 100 bucks |
| 19:01.57 | WIMPy | That probably means Linux, so I'd try to do nothing and hope that it just works. |
| 19:02.19 | Tuju | now i get this: Registration from '<sip:mato@tuju.fi>' failed for '213.219. |
| 19:02.26 | WIMPy | With nat=yes on the Asterisk side. |
| 19:02.32 | Katty | the routers are appliances |
| 19:02.33 | p3nguin | You should be able to have multiple devices on the same LAN connecting out to a remote Asterisk system, and no ports need to be forwarded on the LAN where those devices are. |
| 19:02.35 | Katty | not linux boxes. |
| 19:02.50 | Tuju | my account username is 'tuju', if i change that 'mato' --> 'tuju', i don't get anything into asterisk side anymore. |
| 19:03.00 | WIMPy | Most plastic routers are Linux. |
| 19:03.07 | Katty | p3nguin: how do you not forward 5060 and rtp ports, from the client location, through the firewall, to the IP? |
| 19:03.14 | Tuju | am i supposed to use some phone number instead of my registering username? |
| 19:03.16 | Katty | p3nguin: and expect it to work properly |
| 19:03.20 | p3nguin | Just don't forward them. |
| 19:03.37 | p3nguin | I have phones in remote LANs without ports being forwarded. They work fine. |
| 19:03.38 | Katty | and then ...the rtp ports won't get forwarded through the firewall to the internal devices |
| 19:03.49 | Katty | which means no audio |
| 19:03.50 | WIMPy | Katty: Because the phone will send out packets and the connection tracking will take care of the replies. |
| 19:04.10 | Katty | how do you initiate a call from the asterisk server, to the phone at the remote location, if there are no ports opened |
| 19:04.14 | p3nguin | This is typical "phone behind NAT with asterisk in another network" configuration. |
| 19:04.21 | Tuju | now i get this: Registration from '<sip:5551@tuju.fi>' failed for '213.219................. - No matching peer found |
| 19:04.21 | WIMPy | Just try it. It probably just works without having to do anything. |
| 19:05.13 | p3nguin | You'll just have to make sure you use type=friend for the devices in that single LAN, and make sure you configure all the NAT stuff appropriately. |
| 19:09.31 | Tuju | this is weird - if i put 'wrong' settings there, it just doesn't register. if i put those i think are correct, i get ICMP destination port unreachable |
| 19:12.06 | raden | Katty, :D :D :D :D :D :D |
| 19:12.14 | raden | gives Katty huge hugs |
| 19:13.17 | raden | Katty, you trying to setup some SIP phones at a remote location with crappy routers ? |
| 19:21.53 | raden | Im out |
| 19:23.24 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:25.59 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:26.02 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 19:30.29 | Katty | raden: i do whatever my company tells me. |
| 19:32.10 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:32.32 | anonymouz666 | anyone has a script to convert all old ExecIf syntax to the new one? ;) |
| 19:33.18 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 19:35.43 | *** join/#asterisk x1user (~chatzilla@hotel-palas.com) |
| 19:36.13 | p3nguin | How many ExecIfs do you have that need to be changed? |
| 19:36.48 | x1user | Hi, i have the following problen when loading module [Aug 9 01:34:38] WARNING[3535]: chan_mobile.c:2756 sdp_register: Failed to connect sdp and create session. |
| 19:37.59 | anonymouz666 | p3nguin: TONS |
| 19:38.26 | anonymouz666 | 181 |
| 19:44.56 | p3nguin | Changing the ,Set, to ?Set( would be easy, but I'm not sure how to go about adding the closing ) after the app data. |
| 19:45.19 | p3nguin | or whatever your app is |
| 19:45.38 | p3nguin | (Set is what I most often use with ExecIf) |
| 19:46.30 | p3nguin | If you know awk, I'm sure you'll be able to handle the task with ease. |
| 19:50.45 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 19:51.36 | anonymouz666 | 241 active channels |
| 19:51.37 | anonymouz666 | 202 active calls |
| 19:51.47 | anonymouz666 | nice, isn't ? |
| 19:51.52 | anonymouz666 | it |
| 19:53.52 | p3nguin | How much RAM and CPU is asterisk using with those calls active? |
| 19:54.26 | anonymouz666 | 8 GB RAM |
| 19:54.32 | anonymouz666 | about 50% in use |
| 19:54.42 | p3nguin | Holy crap. I typically use 40 MB. |
| 19:55.14 | anonymouz666 | load 3.04 |
| 19:55.18 | anonymouz666 | 8 cores |
| 19:55.21 | p3nguin | How does asterisk use 8G memory?! |
| 19:55.30 | anonymouz666 | memory is cheap. |
| 19:55.38 | anonymouz666 | it's there |
| 19:55.41 | anonymouz666 | for OS to use. |
| 19:55.43 | p3nguin | Your response does not make sense. |
| 19:56.06 | anonymouz666 | another thing, I make use of query cache. |
| 19:57.15 | anonymouz666 | sip, iax2 and dahdi channels all together |
| 19:57.30 | anonymouz666 | with my last talk with leifmadsen, iax2 is gonna away |
| 19:57.41 | leifmadsen | anonymouz666: that's not at all what I just said |
| 19:57.51 | leifmadsen | or was implying |
| 19:57.52 | _Corey_ | anonymouz666: Just curious if you have a moment, can you do a "cat /proc/YOURASTERISKPID/status" and pastebin the output? |
| 19:58.06 | leifmadsen | What I said, is I never use IAX2 |
| 19:59.12 | anonymouz666 | yeap, I don't need to use it either, there's no benefit |
| 19:59.20 | anonymouz666 | it makes things more complicated in case of debugging etc. |
| 19:59.25 | p3nguin | Sure there's benefit. |
| 19:59.29 | p3nguin | SIP can't do trunking. |
| 19:59.47 | anonymouz666 | I don't need trunking :-) |
| 19:59.58 | anonymouz666 | sorry I was speaking on my setup only. |
| 20:00.05 | p3nguin | Just because you don't need it doesn't mean there's no benefit of it. |
| 20:00.15 | anonymouz666 | correct. |
| 20:00.23 | p3nguin | But I'll accept your addendum. |
| 20:01.29 | anonymouz666 | _Corey_: do you wanna see some line specifically? |
| 20:02.17 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 20:03.26 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 20:05.44 | _Corey_ | anonymouz666: SleepAVG. VMsize, Threads, etc. just curious w/your concurrent calls |
| 20:08.54 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 20:09.52 | anonymouz666 | SleepAVG: 98% VmSize: 1034072 kB Threads: 256 - the calls down to 175 calls |
| 20:10.53 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 20:11.15 | *** part/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 20:11.38 | p3nguin | Hmm, you told me asterisk was using 8G, but it's only using 1G. |
| 20:12.32 | anonymouz666 | ahh I told you the result of top command |
| 20:13.04 | p3nguin | Even top wouldn't say asterisk is using 8G if it is only using 1G. |
| 20:13.11 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 20:13.25 | anonymouz666 | I told you the O.S. usage |
| 20:13.33 | p3nguin | *sigh* |
| 20:13.56 | drift- | p3nguin! |
| 20:14.06 | drift- | the man i need to see :D |
| 20:14.18 | _Corey_ | anonymouz666: Cool, interesting thx |
| 20:25.49 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 20:29.55 | anonymouz666 | _Corey_: I think these indicators will improve, because I have to update this system from 1.4 to 1.8. |
| 20:34.05 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:57.32 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:57.32 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 21:58.45 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 22:03.28 | *** join/#asterisk tully` (~tully@66.76.60.154) |
| 22:04.06 | tully` | is there a way to make asterisk start an EAGI() function but not wait for completion of the script? |
| 22:09.20 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 22:38.54 | *** join/#asterisk damageless (~damageles@68.178.118.142) |
| 22:47.16 | nny | shouldn't Gotoif($[${count}!=1]?trap) mean If number doesn't =1 go to trap? |
| 22:50.15 | leifmadsen | nny: yes, unless ${count} is null |
| 22:50.28 | leifmadsen | nny: which would make that an invalid statement |
| 22:51.22 | nny | leifmadsen: ok gotcha |
| 22:52.14 | leifmadsen | GotoIf($["${count}" != 1]?trap) is better, or alternatively if you're counting or comparing using > or <, then you could do something like GotoIf($[0${count} >= 1]?trap) |
| 22:52.50 | leifmadsen | (or check on the preceding line using ISNULL() or EXISTS()) |
| 22:53.02 | nny | leifmadsen: thanks, engineering a doosy here. Have 2 users in a conference, when user B connects, I want to play a sound file only to User A. I am using an originate command and trying to figure out what the channel value is for the meetme room. Heh |
| 22:53.17 | nny | leifmadsen: yeah just confirming my meetme room has 1 participant |
| 22:56.29 | nny | any way to see the channel name of a meetme room? |
| 22:58.11 | leifmadsen | meetme isn't a channel |
| 22:58.18 | leifmadsen | it's just the end point for a channel |
| 22:59.52 | nny | sorry what I mean to ask is see what channel(s) are specifically connected to meetme room X |
| 23:00.00 | nny | it's a long shot. |
| 23:00.32 | nny | mayeb can just call the app directly from originate |
| 23:00.33 | nny | nm |
| 23:00.42 | nny | let me read up on how originate works |
| 23:00.52 | nny | it's changed since 1.4 |
| 23:01.35 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:01.49 | *** join/#asterisk shine (~stroll@lamantin.achamo.net) |
| 23:03.06 | nny | leifmadsen: thanks think I have it figured out, some hackery needed |
| 23:03.12 | leifmadsen | np |
| 23:09.22 | *** join/#asterisk shine (~stroll@lamantin.achamo.net) |
| 23:11.10 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:11.30 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:14.19 | nny | leifmadsen: so GotoIf($[0${count} >= 1]?trap) means if 0 (or 00) or 02 then ? trap else continue? |
| 23:14.28 | leifmadsen | yes |
| 23:14.39 | leifmadsen | not 02 |
| 23:14.39 | nny | k thanks, i see how that fixes null |
| 23:14.43 | leifmadsen | 02 >= 1 |
| 23:15.01 | leifmadsen | in that case it returns trap |
| 23:15.05 | nny | what about GotoIf($[0${count} != 1]?trap) |
| 23:15.15 | nny | er |
| 23:15.17 | nny | 01 != 1? |
| 23:15.23 | leifmadsen | if 01 isn't returns, then true |
| 23:15.24 | nny | is it literal or case matching? |
| 23:15.29 | leifmadsen | what case? |
| 23:15.43 | leifmadsen | 01 == 1 |
| 23:16.07 | leifmadsen | 02 is greater than or equal to 1 |
| 23:16.17 | leifmadsen | 00 or 02 != 1 |
| 23:16.18 | nny | hmm. yeah but 01 isn't 1 |
| 23:16.23 | leifmadsen | 01 does equal 1 |
| 23:16.25 | leifmadsen | yes it is |
| 23:16.29 | leifmadsen | you're comparing numbers |
| 23:16.32 | nny | ahh ok |
| 23:16.33 | leifmadsen | not strings |
| 23:16.34 | nny | that's what I mean |
| 23:16.36 | nny | meant* |
| 23:16.47 | leifmadsen | 000000001 still equals 1 |
| 23:16.58 | nny | yes, that's what I meant by literal |
| 23:17.08 | leifmadsen | your use of "literal" is incorrect |
| 23:17.12 | nny | yeah heh |
| 23:17.29 | nny | I figured that, should have said, is it comparing the string or the number |
| 23:18.11 | nny | i'll re-read the gotoif section again. Always messes with me (or me with it) |
| 23:25.18 | *** join/#asterisk Yudaisrael1984 (~Yudaisrae@80.179.161.117.static.012.net.il) |
| 23:25.37 | Yudaisrael1984 | guys is there anyone who can helkp me with basic linux commands i messed up and i need a fix |
| 23:25.55 | Yudaisrael1984 | i wrote by mistake mv /* /var/www/html/test/* |
| 23:26.08 | Yudaisrael1984 | and now i cant do anything |
| 23:26.19 | Yudaisrael1984 | yet im still in the system |
| 23:26.27 | Yudaisrael1984 | is there anyway to mv it back???? |
| 23:27.12 | Yudaisrael1984 | anyone? |
| 23:28.41 | p3nguin | I wish you good luck. |
| 23:28.54 | Yudaisrael1984 | damn it |
| 23:29.16 | nny | ouch |
| 23:29.20 | p3nguin | You can probably do it using a rescue CD, but I doubt you'll do it on the current session. |
| 23:29.20 | nny | do this |
| 23:30.01 | nny | hah maybe /var/www/html/test/bin/mv /var/www/html/test/* / |
| 23:30.03 | p3nguin | You could try /var/www/html/test/bin/mv /var/www/html/test/* / |
| 23:30.07 | nny | as in call mv from it's new location |
| 23:30.09 | p3nguin | its |
| 23:30.14 | nny | its |
| 23:30.16 | nny | lol indeed |
| 23:30.19 | nny | HERE COMES AN S! |
| 23:30.30 | nny | p3nguin: nice mirrored response, I beat ya though :D |
| 23:30.55 | Yudaisrael1984 | tried i get an error |
| 23:31.03 | p3nguin | If that fails or doesn't work as expected, the rescue CD can certainly help. |
| 23:31.08 | nny | yeah |
| 23:31.49 | Yudaisrael1984 | i get lib/ld-linux.so.2 |
| 23:32.21 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:32.25 | Yudaisrael1984 | bad elf interpreter |
| 23:33.16 | drudge` | damn keelber elfs |
| 23:33.31 | drudge` | pequnio keebleros, no beuno |
| 23:33.41 | nny | Yudaisrael1984: better to mount a cd and move it with a non toasted os |
| 23:34.39 | Tuju | how do i force asterisk to send the SIP responses to 5060, instead of return port? |
| 23:40.27 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:40.44 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:49.46 | p3nguin | I would expect it will always send to the standard SIP port of 5060 unless you explicitly tell it to go somewhere else. |
| 23:52.19 | brdude | I want to get a sip trunk and DID for brazil any idea where I should go. Was looking it up on voip-info.org but the site is down. |
| 23:53.23 | jeffspeff | I'm having a pretty bad brainfart why won't this following line work? --> exten=200,n,Dial(201&202&203&204&205&206) <-- the numbers are extensions defined in another context, and did an include=phone context to make sure the link was there between the two contexts |
| 23:53.23 | Tuju | p3nguin: it doesn't |
| 23:53.50 | Tuju | rfc says that replys must come back to 5060, but asterisk sends them into UDP src port. |
| 23:54.21 | Tuju | there is plenty of similar cases in net and something related to this was fixed in asterisk around 1.4 |
| 23:55.01 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:55.04 | Tuju | i found this <voipControlPort>5060</voipControlPort> and set it but it still sends using src port +40k |
| 23:55.14 | Tuju | and asterisk uses those for responses. |
| 23:55.20 | Tuju | which are not open in cisco |
