| 00:15.49 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:18.49 | hardwire | moo |
| 00:28.02 | *** join/#asterisk Chinorro (~chino@84.127.234.228.static.user.ono.com) |
| 00:29.33 | *** join/#asterisk DelphiWorld (~Miranda@openvpn/user/DelphiWorld) |
| 00:30.24 | *** part/#asterisk DelphiWorld (~Miranda@openvpn/user/DelphiWorld) |
| 00:36.17 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 00:36.25 | timeshell | Greetings |
| 00:36.31 | timeshell | I got a major issue here. |
| 00:39.07 | timeshell | Phones logging into asterisk 1.6.0.26 |
| 00:39.18 | timeshell | All phones log in fine at first. |
| 00:39.32 | timeshell | But one by one asterisk debug reports "SIP registration auth loop caused by stale nonce" |
| 00:39.34 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 00:40.09 | timeshell | And then every phone is reported by asterisk as "SIP/2.0 401 Unauthorized" |
| 00:41.27 | timeshell | Additionally, the server cannot ping the phones either. |
| 00:41.46 | timeshell | When the phone is able to connect it is able to ping it. |
| 00:42.20 | timeshell | However, afterward, when the phone is being reported as unauthorized, it no longer recieves ping replies. |
| 00:43.25 | timeshell | I'm baffled at the situation. Rebooting the phones allow them to work briefly, but then afterward they return to an unauthorized, unpingable state. |
| 00:43.39 | timeshell | I have rebooted the asterisk server, the network switches and the phones. |
| 00:56.48 | *** join/#asterisk [Outcast] (~anonymous@westford-nat.juniper.net) |
| 01:00.05 | *** join/#asterisk fofware (~fabian@host148.186-124-230.telecom.net.ar) |
| 01:07.23 | *** join/#asterisk [Outcast]_ (~anonymous@pool-108-7-70-195.bstnma.fios.verizon.net) |
| 01:08.58 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 01:17.23 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 01:18.50 | *** join/#asterisk emsLinux (~emsLinux@190.165.101.38) |
| 01:18.59 | emsLinux | hello people |
| 01:19.44 | emsLinux | guys, anyone knows if there is a way to send a message to every extension in asterisk, something like an announce to let clients know something? |
| 01:25.50 | p3nguin | That might be quite useless, considering many extensions don't Dial phones. |
| 01:28.01 | emsLinux | i think you got it wrong, or maybe i did, my intention is to send a message to every extension in the server, something like a broadcast message but personalized for every extension |
| 01:28.37 | p3nguin | I didn't get it wrong. |
| 01:33.55 | emsLinux | so i dont understand what you mean with your comment, my english is not very good |
| 01:35.51 | p3nguin | What I mean is that you don't want to send this message to "extensions" as you put it, but rather send the message to PHONES or other DEVICES. |
| 01:38.37 | emsLinux | no, in fact is to extension, i have a SIP service and i need to remember the pay date to my clients, just a reminder, is not telemarketing |
| 01:39.01 | emsLinux | so i dont need to call outside my server, just to the extensions |
| 01:39.23 | emsLinux | what about cmd Originate, may this work? |
| 01:39.25 | p3nguin | You just don't get it. |
| 01:39.36 | p3nguin | So I'm not going to bother trying to explain it again. |
| 01:39.47 | emsLinux | srry, i don't, told you my english is not the best and im kind of new in asterisk |
| 01:40.30 | p3nguin | You want to send a message to a PHONE, not to an extension. |
| 01:40.50 | emsLinux | ok, how can i do it, what do i need? |
| 01:41.43 | p3nguin | You probably need some pre-recorded message and a mechanism to create call files. |
| 01:44.28 | emsLinux | yeah, the mechanist is the problem, i dont know how to do it, and i can't find any usefull info on Internet |
| 01:48.32 | *** join/#asterisk Chinorro (~chino@84.127.234.228.static.user.ono.com) |
| 01:53.36 | *** join/#asterisk coppice (~coppice@m121-202-23-28.smartone-vodafone.com) |
| 01:53.58 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:54.03 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 01:54.21 | *** join/#asterisk jackmcbarn (~jackmcbar@2002:47c7:7dd2::2) |
| 01:56.48 | jackmcbarn | how can i play around with asterisk without buying any hardware? |
| 01:57.38 | p3nguin | If you already have a computer running a Linux-based OS, install Asterisk on it. |
| 01:57.49 | jackmcbarn | okay |
| 01:57.54 | jackmcbarn | but doesnt asterisk need phones? |
| 01:58.09 | p3nguin | That depends... |
| 01:58.16 | p3nguin | If you want to make phone calls, yes it does. |
| 01:58.49 | p3nguin | But the phones can be software phones. |
| 02:02.55 | p3nguin | If you don't run a Linux-based OS and you have a spare computer, you can install AsteriskNOW on the spare computer and have a dedicated Asterisk box. |
| 02:05.35 | p3nguin | AsteriskNOW is built on CentOS, with Asterisk and an optional choice of GUI (which I don't recommend using). AsteriskNOW with no GUI is a reasonable starting point for trying to come into the industry. |
| 02:30.21 | timeshell | can somone please explain how I can deal with "chan_sip.c:11360 check_auth: Correct auth, but based on stale nonce received from '<" |
| 02:31.52 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:31.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 02:34.45 | *** join/#asterisk adolfomaltez (~taro@190.86.145.104) |
| 02:41.49 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 02:50.58 | qop | is afraid of leifmadsen |
| 02:55.44 | *** join/#asterisk sourcode (~code@ppp-61-90-39-166.revip.asianet.co.th) |
| 02:56.33 | leifmadsen | qop: as you should be |
| 02:56.39 | leifmadsen | runs off to bed |
| 02:57.36 | qop | asterisk -r >> cdr status command not found? |
| 02:57.42 | qop | lol |
| 03:11.34 | *** join/#asterisk gravin (~gravin@217.71.50.60.brf01-home.tm.net.my) |
| 03:26.54 | *** join/#asterisk coppice (~coppice@m121-202-23-28.smartone-vodafone.com) |
| 03:28.15 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 03:34.55 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.216.126) |
| 03:39.05 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 03:54.51 | timeshell | can somone please explain how I can deal with "chan_sip.c:11360 check_auth: Correct auth, but based on stale nonce received from '<" |
| 03:57.18 | *** join/#asterisk remx (~remx@corporate.runecho.com) |
| 04:01.08 | jql | what do you mean by "deal" with it? |
| 04:01.30 | jql | a normal UA would just accept the new challenge and proceed as usual |
| 04:10.40 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
| 04:23.47 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 04:38.30 | *** part/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 04:43.00 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 04:43.19 | timeshell | can somone please explain how I can deal with "chan_sip.c:11360 check_auth: Correct auth, but based on stale nonce received from '<" |
| 04:43.50 | timeshell | I'm desperately need help to resolve this. |
| 04:44.12 | p3nguin | Will saying it over and over and over and over really help? |
| 04:44.32 | timeshell | if someone decides to help, yes |
| 04:44.34 | p3nguin | (2301.08) <jql> what do you mean by "deal" with it? |
| 04:44.37 | p3nguin | (2301.31) <jql> a normal UA would just accept the new challenge and proceed as usual |
| 04:44.52 | timeshell | Missed that |
| 04:44.59 | timeshell | It doesn't proceed |
| 04:45.02 | p3nguin | I know. That's why I pasted it now. |
| 04:45.29 | timeshell | asterisk proceeds with <--- Transmitting (NAT) to 10.20.15.187:5060 ---> |
| 04:45.30 | timeshell | SIP/2.0 401 Unauthorized |
| 04:46.03 | timeshell | And then the phone just repeatedly tries to register and keeps getting this message |
| 04:46.24 | timeshell | Just started today. We've had this working for 2 or 3 years now. |
| 04:46.39 | timeshell | I have rebooted everything. |
| 04:46.45 | timeshell | I don't know what else to try. |
| 04:47.44 | timeshell | I wasn't logged in when jql replied...that's why I missed it. |
| 04:47.58 | timeshell | Probably when I was rebooting my switches./ |
| 05:00.07 | *** join/#asterisk gravin (~gravin@175.136.252.105) |
| 05:00.35 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 05:06.07 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 05:14.23 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 05:14.29 | eject_ck | Hi all |
| 05:15.07 | eject_ck | What's best way to make smtp to fax gateway with asterisk ? Fetchmail + FFA ? |
| 05:15.25 | *** join/#asterisk gravin (~gravin@175.136.252.105) |
| 05:22.33 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 05:25.48 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 05:31.39 | *** join/#asterisk gravin (~gravin@175.136.252.105) |
| 05:43.17 | *** join/#asterisk hfb (~hfb@cpe-98-151-252-78.socal.res.rr.com) |
| 05:44.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-aupsjrglkqfsbhcx) |
| 05:50.46 | *** join/#asterisk Dr{Who} (~mathewss@c-98-232-143-236.hsd1.or.comcast.net) |
| 05:53.29 | Dr{Who} | if I have a call file with maxtries set to 2 does it call an extension upon the final failure. I am making a nagios notificaiton caller system to help make my life a little easier and I am trying to prevent it from calling me again if another down event occures so I am adding my call to a database that I want to delete only when all call attempts have been made. |
| 05:56.59 | jql | Dr{Who}: yeah, you can do something when the final failure occurs |
| 05:57.50 | jql | you can either archive the call file with its final status appended, or grab the failed extension |
| 05:58.23 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-219.telkomadsl.co.za) |
| 06:00.22 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:01.32 | *** join/#asterisk _R0N_ (~ron@vpns-001.videoandmedia.eu) |
| 06:01.40 | Dr{Who} | currently i have the failure extension hook set but it gets call after every attempt. I am readying pbx_spool.c but dont see anyplace it tells asterisk that the call has finnaly failed that gives me a way to capture that event and do something. |
| 06:02.21 | Dr{Who} | so it modifies the call file so my daemon could find the final call files and parse it up. ok thats one plan thanks. |
| 06:02.22 | jql | what's the actual extension line? (one line only, please) |
| 06:02.45 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:03.30 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:03.37 | Dr{Who} | ok im a little new to this and my extension is Huge so far 140 lines. |
| 06:03.57 | jql | yeah no don't want that. heh |
| 06:03.59 | p3nguin | A single extension is 140 lines? Good grief!!! |
| 06:04.15 | jql | points at AGI briefly |
| 06:04.18 | p3nguin | Not extensions.conf |
| 06:04.21 | p3nguin | just one extension. |
| 06:04.37 | jql | thank goodness for n |
| 06:04.41 | p3nguin | I doubt I have more than 20 lines on any single extension. |
| 06:04.48 | Dr{Who} | Its an automated process that lets me acknoloedge a host problem in nagios so lots of voice prompts etc. |
| 06:05.29 | jql | I think I have one up around 40 lines, but it's just a giant switch statement. Goto(${VAR}), n(x1), n(x2), n(x3), n(x4) |
| 06:05.52 | Dr{Who} | im using visual dialplan so its not too painfull to read |
| 06:05.56 | p3nguin | I don't even know how that's necessary. |
| 06:06.23 | Dr{Who} | lots of db queries updates/deletes to fill tables with stuff to send back to nagios |
| 06:06.54 | jql | whatever floats your boat, man |
| 06:07.07 | Dr{Who} | so it sounds like the only way is to look for the completed call file and use archive |
| 06:09.44 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:09.46 | schmidts | good morning |
| 06:09.57 | timeshell | what's so good about it/ |
| 06:10.11 | schmidts | atleast its friday ;) |
| 06:10.25 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:10.40 | timeshell | It'll be good if I get to sleep. |
| 06:10.52 | schmidts | :D |
| 06:10.53 | *** join/#asterisk _R0N_ (~ron@vpns-001.videoandmedia.eu) |
| 06:11.13 | timeshell | Fighting with this stupid * server all night |
| 06:11.18 | irroot | got to be out the house 6:30 tommorow takes the "thrill" off friday |
| 06:11.28 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 06:11.38 | irroot | timeshell it fighting back ?? |
| 06:11.41 | timeshell | Big time |
| 06:11.52 | irroot | what up |
| 06:12.11 | timeshell | chan_sip.c:11360 check_auth: Correct auth, but based on stale nonce received from |
| 06:12.16 | timeshell | asterisk proceeds with <--- Transmitting (NAT) to 10.20.15.187:5060 --->SIP/2.0 401 Unauthorized |
| 06:12.21 | timeshell | endless loop |
| 06:12.54 | irroot | only one extension trying to register with that user |
| 06:12.58 | timeshell | yes |
| 06:13.05 | timeshell | Every phone is doing it |
| 06:13.18 | timeshell | At least every phone on a specific switch is doing it. |
| 06:13.28 | timeshell | Logs in. Then after a short while starts doing that. |
| 06:13.51 | timeshell | We're talking about 20 phones. |
| 06:13.55 | timeshell | At least. |
| 06:14.00 | irroot | the switch not doing nat or something the problem is the nooce |
| 06:14.12 | irroot | s/nooce/noonce/ |
| 06:14.13 | timeshell | You asking or telling? |
| 06:14.21 | schmidts | timeshell you will get a stale nonce when an account is still registered and tries to reregister from another port |
| 06:14.31 | timeshell | Fine. |
| 06:14.35 | timeshell | How do you fix it? |
| 06:14.35 | schmidts | timeshell do you have qualify=yes? maybe its a problem with closed nat ports |
| 06:14.44 | timeshell | Shouldn't be nating |
| 06:14.49 | irroot | also try insecure=port,invite |
| 06:14.56 | p3nguin | s/noonce/nonce/ |
| 06:15.11 | timeshell | The server and phones are on the same network. |
| 06:15.21 | irroot | lo there p3nguin |
| 06:15.27 | timeshell | Same VLAN |
| 06:15.38 | p3nguin | I'm a bit intoxicated, so don't mind me. |
| 06:16.02 | timeshell | Seems to happen whether I have NAT set on the user or not. |
| 06:16.04 | p3nguin | Absolut! |
| 06:16.43 | irroot | is jealous of p3nguin .... 08h00 is bit early |
| 06:16.50 | timeshell | Please help. |
| 06:16.54 | timeshell | I'm running out of options. |
| 06:17.04 | p3nguin | 01.17 over here. |
| 06:17.34 | timeshell | It all seemed to start when I disconnected another switch connected to the switch that the phones are connected to. The phones haven't changed and no other settings on the switch have changed. |
| 06:17.38 | p3nguin | I should be sleeping, but drinking instead. |
| 06:17.48 | timeshell | I even put back the switch, but it didn't restore functionality. |
| 06:17.52 | wdoekes2 | timeshell: does the CSeq increase for the register packets? are the nonces stale? |
| 06:18.14 | wdoekes2 | can you tell whether the phones receive anything at all from the asterisk? |
| 06:18.56 | timeshell | wdoekes2 On reboot, they register at first. |
| 06:19.04 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 06:19.14 | timeshell | Shortly afterwards they go into this No Auth. |
| 06:19.23 | timeshell | I've been using this config for about 2 or 3 years. |
| 06:19.27 | timeshell | I'm baffled. |
| 06:19.30 | wdoekes2 | I read that.. and the answer to my questions? |
| 06:19.43 | timeshell | Right now the phones can call each other. |
| 06:19.53 | timeshell | So yes, they have to be communicating with asterisk. |
| 06:20.00 | timeshell | CSeq |
| 06:20.02 | timeshell | Checking |
| 06:20.17 | timeshell | Kinda hard to track... lots of phone messages. |
| 06:20.26 | wdoekes2 | you can sip set debug ip <ip> |
| 06:20.51 | timeshell | It seems to be behaving right now. |
| 06:21.10 | timeshell | I upgraded from 1.6.0.26 to 1.6.2.19 to try to resolve this. |
| 06:21.17 | irroot | timeshell can i make a sugestion i have seen a oddity like this b4 |
| 06:21.17 | timeshell | At first it didn't seem to work. |
| 06:21.29 | irroot | there is a bridge on the VLAN to native lan ?? |
| 06:21.43 | timeshell | But now I see a lot less No Auth messages. |
| 06:21.57 | timeshell | irroot There shouldn't be. |
| 06:22.04 | irroot | where the phones flip flop between networks |
| 06:22.16 | timeshell | That's what's been in my mind all aloong |
| 06:22.39 | timeshell | However, I only have a bridge across 2 vlans on another switch, VLAN's 1 and 5. The phones are on Vlan 15 |
| 06:22.43 | wdoekes2 | well.. if they happen again, you'll want to know whether the phones got the 401 or not (you can tell that they got them if they're using the new nonce and an increased cseq on the 2nd register) |
| 06:23.15 | timeshell | I'll watch for that. |
| 06:23.17 | timeshell | Thanks |
| 06:23.20 | timeshell | But even if that is so |
| 06:23.24 | irroot | had a site with 2 network points one for phone one for net in some places the phone was vlan some fool unpluged laptop plugged it into the pc port on the phone so phone bridged the 2 it was a mess |
| 06:23.24 | timeshell | How to fix it? |
| 06:23.51 | wdoekes2 | in that case the problem is either that (1) asterisk sends the 401 to the wrong location or (2) some switch/router/whatever is eating or misrouting your packets |
| 06:23.51 | timeshell | irroot, For me, I force the phone vlan to be tagged |
| 06:24.00 | timeshell | That prevents accidental usage. |
| 06:24.23 | timeshell | wdoekes2 I suspect it is 2 |
| 06:24.38 | timeshell | Because when I was getting the No Auth I also was unable to get a ping reply from the phone. |
| 06:24.47 | timeshell | Which was also baffling me. |
| 06:24.56 | wdoekes2 | it is however unlikely that you can still get any other packets to the phones at that point => you shouldn't be able to setup a call either |
| 06:24.58 | schmidts | timeshell maybe you can trace this with ngrep or tcpdump on the other network, if you see some packets coming in there from the phones you know whats going wrong |
| 06:25.00 | irroot | if you have a bad non vlan switch / hub on the net it can mess it up |
| 06:25.22 | timeshell | schmidts I have wireshark running right now trying to figure that out. |
| 06:25.27 | irroot | p3nguin why sleep when you can drink :P |
| 06:25.52 | timeshell | irroot ?? Explain |
| 06:26.46 | irroot | a old 4 port dlink under someones desk is the root of much evil |
| 06:26.57 | irroot | or a rougue wifi router |
| 06:27.13 | timeshell | Neither are likely here unless I put it in. |
| 06:27.19 | timeshell | Nobody here is savvy enough. |
| 06:28.56 | timeshell | wdoekes2 With respect to point 2, like I said earlier, I removed a switch and plugged a different switch into its port. I suspected that may cause MAC issues, but I rebooted all the switches and it didn't resolve anything. |
| 06:29.29 | schmidts | timeshell maybe you forget some vlans to configure on the new switch? |
| 06:29.56 | timeshell | No, the new switch was already configured and on the network. Just relocated. |
| 06:30.07 | timeshell | Same trunk settings as the one that was removed. |
| 06:30.17 | timeshell | And I already verified it too. |
| 06:30.19 | timeshell | :p |
| 06:30.28 | timeshell | brb |
| 06:30.34 | timeshell | Gotta plug more phones back in |
| 06:31.07 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 06:31.23 | jacc0 | good morning all! |
| 06:31.28 | jacc0 | :) |
| 06:31.33 | jacc0 | almost weekend!! |
| 06:31.58 | schmidts | Morning jacc0 |
| 06:32.18 | irroot | morning jacc0 |
| 06:32.30 | timeshell | back |
| 06:32.34 | timeshell | So far so good |
| 06:32.54 | timeshell | Still getting Unauth messages, but they register right after now |
| 06:33.16 | timeshell | What a weird day |
| 06:33.20 | timeshell | Never seen this before. |
| 06:33.31 | schmidts | ;) |
| 06:33.44 | timeshell | I've beeen working on this for 10 hours now. |
| 06:34.07 | timeshell | And there |
| 06:34.12 | wdoekes2 | you should get unauth messages, once per register |
| 06:34.20 | timeshell | will be hell from the boss if it's not working by office openb |
| 06:34.28 | timeshell | Well that's good |
| 06:34.41 | timeshell | I was only getting them repeatedly before. |
| 06:34.48 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 06:34.54 | timeshell | I didn't want to do a blind upgrade to 1.6.2.x either. |
| 06:35.10 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 06:35.15 | timeshell | But looks like there was a patch to deal with this a couple years ago so I thought I better try it. |
| 06:35.37 | timeshell | MOH isn't working correctly now |
| 06:35.39 | timeshell | :p |
| 06:35.54 | timeshell | shoot me now |
| 06:36.08 | irroot | nah wait for boss to do it :P |
| 06:36.13 | schmidts | :D |
| 06:36.39 | timeshell | Well, it definitely working better now |
| 06:37.08 | timeshell | I'm building a replacement server just in case. |
| 06:37.26 | timeshell | This has to work in the morning. |
| 06:37.44 | timeshell | well, in 4.5 hours |
| 06:38.34 | jacc0 | I have asterisk loosing its binding to the network port :S |
| 06:38.55 | jacc0 | the asterisk proces seems to be running fine |
| 06:39.15 | jacc0 | tshark shows incomming INVITEs |
| 06:39.23 | jacc0 | asterisk is not responding to them |
| 06:39.36 | jql | does netstat show that the port is open? |
| 06:40.28 | jacc0 | I'm not sure; I've automated the proces to check if asterisk is still responding to SIP INVITES; if not is restarts the asterisk process |
| 06:40.40 | jql | netstat -an | grep 5060 |
| 06:40.43 | jacc0 | I now need to automate the proces of gadering debug info |
| 06:40.58 | jacc0 | (i'm not there when this happens |
| 06:41.16 | jacc0 | It's realy a battle to keep aasterisk functining |
| 06:41.53 | jql | are you testing from localhost or over the network? |
| 06:41.55 | jacc0 | we now use safe_asterisk, astcanary, monit and flip1405 in a fail over config to keep things online :S |
| 06:42.14 | jacc0 | we are checking if asterisk is rezsponding to SIP INVITE from local host |
| 06:42.27 | jacc0 | using a Monit script |
| 06:42.33 | jql | well, that removes a whole slew of failure modes |
| 06:42.40 | jacc0 | ;) |
| 06:42.51 | jql | yeah, you're kinda boned |
| 06:43.09 | jacc0 | monit reports ; "no route to host" when trying to connect to 5060 |
| 06:43.15 | jacc0 | (from local host) |
| 06:44.20 | jacc0 | I'll add "netstat -an | grep 5060" to the automated debug info gathering script |
| 06:44.45 | jacc0 | allong with "gdb" and ps aux/top/df |
| 06:44.54 | jacc0 | more sugestions? |
| 06:44.57 | jql | I wish you luck. I'd be spazzing out at that |
| 06:44.59 | jql | Last reload: 24 weeks, 5 hours, 14 minutes, 34 seconds |
| 06:45.09 | jql | I know it can work better than that |
| 06:45.25 | jacc0 | I have like 20 asterisk production env. in the feeld |
| 06:45.41 | jacc0 | trunked with differend PBXes |
| 06:47.26 | jacc0 | debug info i was able to gather from the event : http://pastebin.com/PgnftcvK |
| 06:48.10 | jacc0 | it is an Ascotel PBX |
| 06:48.24 | jacc0 | one thing that stands out in the asterisk log is : cel.c: Invalid AMA flag: 1885958147 |
| 06:49.27 | jacc0 | I'm starting to think it hase something to do with CDR in combination with Originate() and Bridge() |
| 06:50.52 | jacc0 | Originate() and Bridge() might be confusing asterisk cdr |
| 06:52.37 | jacc0 | other PBXes show other asterisk crashes/deadlocks :S It's a strugle to keep asterisk up and running |
| 06:52.52 | jacc0 | But we'll manage ;) |
| 06:55.36 | wdoekes2 | jacc0: you may want to add a -p to to that netstat.. someone else could be using your 5060 |
| 06:58.24 | jacc0 | ty for the tip |
| 06:59.17 | singler | also you can add -l if we are at it :) |
| 07:01.29 | jacc0 | <PROTECTED> |
| 07:03.07 | *** join/#asterisk gravin (~gravin@175.139.239.3) |
| 07:10.38 | wdoekes2 | is asterisk eating 100% cpu? |
| 07:10.42 | jacc0 | nope |
| 07:11.08 | wdoekes2 | and do you get reply from an options packet? |
| 07:11.15 | wdoekes2 | (e.g. with sipsak(1)) |
| 07:12.17 | jacc0 | no reply |
| 07:12.32 | jacc0 | it looks like it lost its binding |
| 07:13.16 | wdoekes2 | but netstat says udp 5060 is still bound? |
| 07:15.19 | wdoekes2 | gdb -p `pidof asterisk` -ex 'thread apply all bt' -ex detach -ex quit > tmp.txt |
| 07:17.18 | jacc0 | I hav'n't been able to do a netstat :S |
| 07:17.29 | jacc0 | scripts make asterisk restart before i can have a look |
| 07:17.40 | jacc0 | I have to automate the process of debug info gathering |
| 07:18.12 | jacc0 | more on this next week :) |
| 07:18.26 | jacc0 | now setting up automated debug gathering scripts :) |
| 07:18.39 | jacc0 | ~backtrace |
| 07:18.40 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 07:21.45 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:25.12 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:26.53 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-182-151.pools.arcor-ip.net) |
| 07:30.45 | *** join/#asterisk Aven (~user@95.52.229.140) |
| 07:35.00 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-182-151.pools.arcor-ip.net) |
| 07:53.05 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 07:55.49 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 07:59.24 | timeshell | <PROTECTED> |
| 07:59.28 | timeshell | CSeq: 1 for SIP/2.0 401 Unauth and for Register |
| 07:59.32 | timeshell | I think I made it happen when I disconnected the switch while fixing wires. |
| 07:59.36 | timeshell | So, how do I fix it? |
| 07:59.53 | timeshell | How do you force nonce's to reset or play nice again? |
| 07:59.58 | timeshell | To properly increment the CSEQ? |
| 08:00.46 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:02.17 | irroot | timeshell not a ip conflict perhaps ?? |
| 08:02.23 | timeshell | nope |
| 08:02.45 | timeshell | Seemed to be caused just by unplugging the trunk port on the switch and plugging it back in. |
| 08:02.52 | timeshell | It's all the phones again |
| 08:03.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:04.50 | timeshell | So, how??? |
| 08:05.06 | timeshell | There's gotta be a way to force the server and the phones to play nice again |
| 08:05.23 | *** part/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
| 08:05.28 | timeshell | [2011-07-08 04:05:20] NOTICE[31575]: chan_sip.c:13329 check_auth: Correct auth, but based on stale nonce received from |
| 08:05.35 | timeshell | I'm getting tons of these now |
| 08:05.41 | irroot | timeshell using realtime ?? |
| 08:05.48 | irroot | if so prune all the peers |
| 08:05.53 | irroot | they will register |
| 08:05.54 | timeshell | debug |
| 08:06.27 | timeshell | sip prune realtime all? |
| 08:07.14 | timeshell | got "No peers found to prune" |
| 08:08.11 | timeshell | "Not a realtime peer" |
| 08:08.20 | timeshell | crap |
| 08:08.24 | timeshell | after all that |
| 08:08.28 | timeshell | It was all working |
| 08:08.34 | timeshell | I was just cleaning up wires. |
| 08:09.20 | irroot | super glue it into the port :P |
| 08:09.33 | timeshell | haha |
| 08:09.37 | timeshell | I really need to fix this |
| 08:09.59 | irroot | modulue unload chan_sip.so |
| 08:10.06 | irroot | and load it ?? |
| 08:10.11 | timeshell | Well sure |
| 08:10.19 | timeshell | Just restart asterisk then? |
| 08:10.23 | timeshell | That doesn't seem to fix it. |
| 08:10.41 | jacc0 | ??? |
| 08:10.57 | jacc0 | set reregister timeout for sip devices to 30 sec. |
| 08:12.22 | timeshell | reregister timeout? Or register timeout? |
| 08:12.40 | jacc0 | what devices are you using ? comon once/ |
| 08:12.42 | jacc0 | ? |
| 08:12.54 | timeshell | polycom 500, 501, 430, 301, 601, 650 |
| 08:13.48 | jacc0 | let me look it up for you |
| 08:14.14 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 08:14.31 | timeshell | Yah, unloading/loading chan_sip.so didn't help |
| 08:14.49 | timeshell | They're still Cseq 1 and looping unauth 401 |
| 08:15.21 | timeshell | Why have I never had this problem before? |
| 08:15.41 | timeshell | It's not the first time I've moved stuff around on the switch. |
| 08:16.42 | jacc0 | restarting the sip client solves the problem? |
| 08:17.09 | timeshell | jacc0 The phones? |
| 08:17.12 | timeshell | Hasn't yet |
| 08:17.18 | jacc0 | yeah |
| 08:17.31 | timeshell | No, doesn't seem to matter what I restart. |
| 08:17.32 | jacc0 | restart the phone; deos that fix the problem? |
| 08:17.52 | jacc0 | okay, somting might be wrong in astdb then |
| 08:17.53 | timeshell | Didn't before. I even formatted the phones filesystem |
| 08:18.20 | timeshell | astdb? |
| 08:18.47 | timeshell | You mean in the config files? |
| 08:18.50 | jacc0 | stop asterisk |
| 08:18.52 | jacc0 | cd /;rm `find -name astdb` |
| 08:18.58 | jacc0 | and start asterisk |
| 08:19.10 | jacc0 | astdb is regenerated when asterisk starts |
| 08:20.38 | timeshell | where is astdb in the filesystem? |
| 08:20.51 | timeshell | nm |
| 08:21.03 | jacc0 | find -name astdb |
| 08:21.14 | timeshell | restarted |
| 08:21.14 | jacc0 | will tell you where it is |
| 08:21.26 | schmidts | /usr/lib/asterisk or /var/spool/asterisk typically ;) |
| 08:21.29 | timeshell | Yah |
| 08:21.29 | irroot | /var/lib/asterisk/astdb |
| 08:21.31 | timeshell | Didn't work |
| 08:22.55 | jacc0 | maybe instaal tshark and have a look wath is happening on the wire |
| 08:23.02 | jacc0 | tshark -R sip |
| 08:23.14 | jacc0 | even better tshark -R sip -V |
| 08:23.53 | jacc0 | are you positive the phones are not registerd? |
| 08:24.43 | timeshell | I see the SIP/2.0 401 Unauthorized messages whizing by |
| 08:24.43 | jacc0 | could it be that the phones first try to register using "digest" resulting in a 401 |
| 08:24.56 | timeshell | And they are all CSeq 1 |
| 08:24.59 | jacc0 | and if the get a 401 they try it in "plain text" ? |
| 08:25.20 | jacc0 | and sucseed |
| 08:25.23 | jacc0 | ? |
| 08:25.28 | timeshell | No |
| 08:25.51 | timeshell | The phones don't ring when called on a working phone (which usually isn't working long) |
| 08:26.08 | timeshell | Just get blank air or voicemail |
| 08:28.03 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:29.22 | wdoekes2 | timeshell: if the phones cseq's aren't increasing they're not getting your 401. you probably can't ping the phones either |
| 08:29.47 | timeshell | When it's like this, you're right I can't |
| 08:30.00 | timeshell | When they first boot up they log in and I can |
| 08:30.10 | timeshell | But when another phone comes on, they seem to disappear. |
| 08:30.17 | timeshell | Can't figure that out. |
| 08:31.05 | wdoekes2 | anything odd in dmesg? does the arp table look ok? (the phone's mac addresses?) any odd arp messages flying around? |
| 08:32.00 | timeshell | a number of the ip addresses in the arp table have "incomplete" HW address |
| 08:33.53 | wdoekes2 | they probably match exactly with the machines that you cannot reach ;) |
| 08:34.13 | timeshell | no doubt |
| 08:34.20 | timeshell | So why were they all working 30 mins ago? |
| 08:34.29 | wdoekes2 | because they were getting good arp back then |
| 08:34.38 | wdoekes2 | you need to kick your switch |
| 08:34.47 | timeshell | Bother |
| 08:35.06 | timeshell | I pulled the network cable that connects it to the server about 30 mins ago |
| 08:35.24 | timeshell | Why isn't the switch responding? |
| 08:36.00 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 08:36.46 | timeshell | All the other computers on the switch appear to be functioning. |
| 08:37.25 | timeshell | And now all the ips in the arp have macs |
| 08:37.51 | jacc0 | flush the arp table? |
| 08:37.57 | timeshell | No... |
| 08:38.08 | timeshell | It seems to be working now |
| 08:38.12 | timeshell | I did delete a couple of them. |
| 08:38.23 | timeshell | Good grief |
| 08:38.26 | timeshell | I hate layer 2 |
| 08:39.04 | timeshell | So that's been the issue all along. |
| 08:39.09 | timeshell | Bah |
| 08:39.27 | timeshell | Well, thanks guys |
| 08:39.43 | timeshell | I flushed the arp on the switch earlier too |
| 08:39.52 | timeshell | But didn't check the arp on the server. |
| 08:40.04 | timeshell | I wouldn't have thought of that (at least not for a while) |
| 08:40.52 | jacc0 | :) |
| 08:40.52 | *** join/#asterisk nightrid3r (~nightrid3@91.176.134.63) |
| 08:40.58 | timeshell | lol, I was so busy trying to fix the problem, it probably would have fixed itself if I waited long enough. :p |
| 08:41.12 | jacc0 | hehehe, true |
| 08:41.29 | timeshell | Well, there's 12 hours wasted. |
| 08:41.47 | timeshell | And McDonald's is closed for renovations. :( |
| 08:41.56 | timeshell | (there's only one here) |
| 08:42.04 | wdoekes2 | good.. that'll save you some stomach ache |
| 08:42.05 | timeshell | I'll have to go to the neighbouring town. |
| 08:42.34 | timeshell | bah |
| 08:42.40 | timeshell | I'm annoyed now |
| 08:43.05 | timeshell | I think I need to go bang my head on something. |
| 08:44.17 | timeshell | At least I don't need to worry about building that second server now. |
| 08:44.40 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 09:00.12 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 09:11.12 | *** join/#asterisk wonderworld (~ww@port-92-201-232-249.dynamic.qsc.de) |
| 09:13.51 | *** join/#asterisk gokulnath (~chatzilla@115.111.177.122) |
| 09:13.59 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 09:17.37 | *** join/#asterisk sourcode (~code@ppp-61-90-39-166.revip.asianet.co.th) |
| 09:19.08 | *** join/#asterisk gravin (~gravin@175.139.239.3) |
| 09:27.47 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 09:32.22 | _R0N_ | I have a dialplan problem.. is there a guru in the house? |
| 09:36.28 | _R0N_ | no one? |
| 09:38.56 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 09:39.59 | jacc0 | ãs |
| 09:40.02 | jacc0 | ~ask |
| 09:40.03 | 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. |
| 09:53.49 | _R0N_ | I want the dialplan to continue after a hangup.. |
| 09:54.02 | _R0N_ | but it does not wait for the hangup.. |
| 09:54.18 | _R0N_ | have now: |
| 09:54.21 | _R0N_ | exten => _X.,1,Ringing |
| 09:54.21 | _R0N_ | exten => _X.,2,Dial(SIP/${EXTEN},20,tgh) |
| 09:54.21 | _R0N_ | exten => _X.,3,Hangup |
| 09:54.22 | _R0N_ | exten => h,1,NoOP(${DIALSTATUS}) |
| 09:54.58 | _R0N_ | but after answer it goes directly to 'h' |
| 09:55.10 | _R0N_ | I use asterisk 1.8.4.4 |
| 09:59.40 | jacc0 | use goto: h,1,goto(${EXTEN},4) |
| 09:59.46 | jacc0 | maybe that works |
| 09:59.57 | jacc0 | that is howI do it |
| 10:00.50 | jacc0 | or do not use ; _X.,3,Hangup |
| 10:01.11 | _R0N_ | should it not wait until the call i sdisconnected? |
| 10:01.32 | _R0N_ | ${DIALSTATUS} gives ANSWER not HANGUP |
| 10:03.21 | _R0N_ | removed _X,3 still the same.. |
| 10:03.54 | jacc0 | F([[context^]exten^]priority): When the caller hangs up, transfer |
| 10:03.54 | jacc0 | <PROTECTED> |
| 10:03.54 | jacc0 | <PROTECTED> |
| 10:04.17 | jacc0 | no, that is not it sorry |
| 10:04.23 | _R0N_ | :) |
| 10:15.16 | irroot | there is a problem with dialstatus in the h extension but cant recall what / why / when |
| 10:17.47 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-252-002.pools.arcor-ip.net) |
| 10:18.06 | irroot | http://asteriskfaqs.org/2010/12/20/asterisk-users/dialstatus-on-cancel.html |
| 10:18.37 | irroot | http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause |
| 10:18.48 | irroot | _RON_ ^^ |
| 10:26.14 | TobSnyder | when trying to use the command "asterisk -rx 'dahdi restart'" I am getting |
| 10:26.14 | TobSnyder | No such command 'dahdi restart' (type 'help dahdi restart' for other possible commands) |
| 10:26.31 | TobSnyder | any Idea why asterisk does not now anything about dahdi / pri commands anymore? |
| 10:26.45 | irroot | mmmm you have not got dahdi loaded |
| 10:28.39 | irroot | module show like dahdi |
| 10:28.49 | irroot | chan_dahdi.so DAHDI Telephony Driver w/PRI & SS7 & MFC 0 |
| 10:33.33 | TobSnyder | chan_dahdi.so DAHDI Telephony w/PRI 0 |
| 10:33.59 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 10:35.07 | _R0N_ | irroot: The problem is not the value of the variable but that it does not wait for the hangup.. |
| 10:36.02 | TobSnyder | ok so I have done "module load chan_dahdi.conf" |
| 10:37.10 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 10:38.07 | TobSnyder | but still there is no dahdi command available? |
| 11:15.48 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 11:16.04 | *** join/#asterisk jsman (~jsman@unaffiliated/jsman) |
| 11:18.37 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 11:24.02 | eject_ck | Hi all |
| 11:27.30 | jacc0 | hi |
| 11:34.43 | schmidts | maybe a stupid question but isnt there a "core show keys" in 1.8 or did i missed something necesary ;) |
| 11:38.46 | leifmadsen | schmidts: no :) |
| 11:38.48 | leifmadsen | schmidts: keys show |
| 11:38.56 | leifmadsen | keys import |
| 11:38.57 | leifmadsen | etc |
| 11:39.09 | schmidts | didnt have "keys" ;) |
| 11:39.12 | leifmadsen | huh |
| 11:39.19 | leifmadsen | do you have res_crypto.so loaded? |
| 11:39.27 | schmidts | aaahh i will check |
| 11:39.36 | leifmadsen | without it you won't have keys :) |
| 11:39.41 | leifmadsen | keys init is another command |
| 11:40.33 | schmidts | ok looks like i have disabled this on this test machine ;) |
| 11:41.10 | leifmadsen | :) |
| 11:41.29 | schmidts | ah no i just havent openssl installed thats why :D |
| 11:42.22 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 11:43.52 | leifmadsen | that'd do it :) |
| 11:48.15 | schmidts | ok strange things happens |
| 11:48.41 | schmidts | installed openssl package, have done ./configure but i still see res_crypto needs openssl ? |
| 11:49.43 | leifmadsen | you didn't install the -devel pkg |
| 11:50.50 | schmidts | i want to go home :( its friday no mind left for something like this today |
| 11:51.54 | schmidts | and btw debian packages are sometimes not very clever named openssl <-> libssl-dev thats why i havent found it by searching for openssl :( |
| 11:56.06 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-131-153.twcny.res.rr.com) |
| 12:01.00 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 12:01.34 | timeshell | wdoekes2 AH!! |
| 12:01.40 | timeshell | Still issues. |
| 12:01.55 | timeshell | When the ARP times out, it still takes far too long to renew. |
| 12:02.01 | timeshell | Phones get lost again |
| 12:02.13 | timeshell | Still having trouble tracking down the cause. |
| 12:02.45 | timeshell | irroot ^^^^ |
| 12:03.18 | wdoekes2 | well.. I did not think that the fix for the issue was just doing nothing |
| 12:03.25 | wdoekes2 | you probably still need to kick your switch |
| 12:03.42 | irroot | ^^^ |
| 12:03.54 | timeshell | lol |
| 12:03.55 | timeshell | I didn' |
| 12:03.59 | timeshell | I didn't just do nothing |
| 12:04.02 | irroot | also worth having a look at STP |
| 12:04.04 | timeshell | I just didn't do the right thing |
| 12:04.51 | timeshell | I'd like to kick my switch |
| 12:05.00 | timeshell | I don't know why it's suddenly misbehaving |
| 12:05.30 | wdoekes2 | it could be broken, it could be misconfigured.. the problems started when you installed this one, right? |
| 12:05.38 | timeshell | NO |
| 12:05.45 | timeshell | I removed one that was attached to it |
| 12:06.02 | wdoekes2 | oh.. the missing link ;) |
| 12:06.02 | timeshell | This main switch that all the phones are on hasn't actually changed |
| 12:06.07 | timeshell | I put it back |
| 12:06.09 | timeshell | It's there now |
| 12:06.13 | timeshell | It didn't resolve the issue |
| 12:06.29 | timeshell | Something else is going on |
| 12:06.41 | wdoekes2 | I'd watch out for any out of the ordinary traffic just before it stops behaving normally |
| 12:07.36 | wdoekes2 | run a tcpdump -w tmp.pcap 'not tcp and not udp' or something on the asterisk machine and look at the output when you know the issue has just occurred.' |
| 12:08.38 | wdoekes2 | that might give you some clues |
| 12:09.39 | wdoekes2 | and.. does the problem recover after a fixed period of time (say 2 minutes (STP)) |
| 12:09.50 | timeshell | iI know that the issue doesn't originate at the switch that the server is connected to as other switches that connect to it still communicate just fine |
| 12:10.26 | timeshell | don't know yet |
| 12:10.33 | wdoekes2 | the fact that others can use the switch fine, does not exclude the switch from blame |
| 12:10.38 | timeshell | NONO |
| 12:10.44 | timeshell | There are a number of switches |
| 12:11.03 | timeshell | The phones on one particular switch is where the problem is |
| 12:11.11 | timeshell | the phones on all the other switches are fine |
| 12:11.16 | wdoekes2 | oh.. ok |
| 12:11.27 | timeshell | There are at least 5 switches involved |
| 12:11.38 | timeshell | 4 connected to a central switch |
| 12:11.47 | timeshell | 1 of the 4 is a problem |
| 12:13.50 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:8178:2b0:8030:7ee9) |
| 12:15.10 | *** join/#asterisk avruser (~user@91.196.77.26) |
| 12:17.41 | timeshell | k, so the arp just cleared (timed out I suppose) on the server. However, when the phones re-registered the arp entries all came back |
| 12:18.19 | avruser | Hello, people! Can i change dialed extension and then handle changed extension in context? Thanks. |
| 12:18.39 | leifmadsen | avruser: huh? |
| 12:18.55 | timeshell | That sounds like a yes |
| 12:19.06 | leifmadsen | I can think of so many things that could be referring to.... |
| 12:19.11 | leifmadsen | avruser: please be more specific :) |
| 12:19.16 | timeshell | hey leif |
| 12:19.19 | timeshell | What's up |
| 12:20.23 | leifmadsen | a large list of things to get done before leaving today |
| 12:21.11 | timeshell | I'm only in a slight panic because our main office switch apparently isn't playing nice with ARPs. |
| 12:23.15 | avruser | leifmadsen: Then my users dial numbers, asterisk search appropriate extension in user context. Can i convert user's dialed number into another number, and then search context again? |
| 12:24.11 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 12:24.14 | leifmadsen | avruser: yes, just use Goto() |
| 12:24.34 | leifmadsen | exten => _2XX,1,Goto(5551${EXTEN},1) |
| 12:25.13 | leifmadsen | if you dialed 211 it would try to Goto() 5551211 |
| 12:26.09 | avruser | leifmadsen: Yes, It's that i need. :) Thank you! |
| 12:26.32 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 12:34.11 | mickecarlsson | leifmadsen ping |
| 12:34.42 | leifmadsen | mickecarlsson: pong? |
| 12:34.52 | mickecarlsson | Is this internal: https://issues.asterisk.org/jira/browse/SWP-3619 |
| 12:35.25 | leifmadsen | yes |
| 12:35.37 | mickecarlsson | OK, because I went to this link: ASTERISK-18099 |
| 12:35.49 | mickecarlsson | And it changed that to the other |
| 12:35.50 | leifmadsen | it's just used for tracking time at Digium |
| 12:35.56 | mickecarlsson | Ah, OK |
| 12:35.58 | leifmadsen | it shouldn't "change" you |
| 12:36.06 | leifmadsen | it should just show as a clone to your original issue |
| 12:36.16 | mickecarlsson | Hmm, let me check |
| 12:37.08 | mickecarlsson | OK, I stretched that a bit by going to https://issues.asterisk.org/jira/browse/ASTERISK-18099 |
| 12:37.24 | mickecarlsson | And that was replaces by the SWP |
| 12:38.05 | leifmadsen | I don't know why it's redirecting automatically |
| 12:38.18 | mickecarlsson | btw, astdb.sqlite3, I have done some testing, no problems at all |
| 12:39.01 | leifmadsen | excellent |
| 12:39.07 | leifmadsen | twilson will be happy to hear that |
| 12:39.55 | mickecarlsson | One minor issue spotted (and it s really minor/trivial) |
| 12:40.12 | *** join/#asterisk cerberus_za (~coert@dsl-185-103-205.dynamic.wa.co.za) |
| 12:40.22 | mickecarlsson | database does not sort alphabetically like the old astdb was doing |
| 12:40.32 | mickecarlsson | database show that is |
| 12:40.44 | mickecarlsson | I can live with that:-) |
| 12:41.37 | mickecarlsson | gtg, coffee time |
| 12:41.46 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 12:43.12 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:45.20 | leifmadsen | mickecarlsson: I don't know where ASTERISK-18099 went |
| 12:46.08 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
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| 12:46.31 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 12:54.15 | timeshell | wow, the issue seems to go away while I'm monitoring the vlan |
| 12:54.26 | timeshell | When I disconnect it comes back after a while. |
| 12:54.34 | timeshell | When I'm packet sniffing, everything is working |
| 12:55.01 | timeshell | HTF do you track down a problem that goes away when your trying to trace it? |
| 12:56.02 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 12:56.19 | wdoekes2 | heisenbug++ |
| 12:56.31 | jacc0 | I guess it's a quantum mechanics law |
| 12:57.20 | timeshell | What do you use to read the bin file from tcpdump? |
| 12:57.30 | wdoekes2 | tcpdump :) |
| 12:57.36 | wdoekes2 | or wireshark |
| 12:58.53 | timeshell | tcpdump <filename> ? |
| 12:59.02 | wdoekes2 | man 1 tcpdump |
| 12:59.25 | timeshell | Forgive me, I don't use commandline Linux as often as I used to. |
| 13:00.01 | *** join/#asterisk Aven (~user@95.52.229.140) |
| 13:00.07 | SunTsu | timeshell: then you probably want to use wireshark |
| 13:00.22 | timeshell | thanks |
| 13:01.07 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 13:01.24 | SunTsu | timeshell: it's a packet analyzer with a neat gui and great features like following tcp streams and such |
| 13:01.38 | timeshell | Yes |
| 13:01.40 | timeshell | I use it alot |
| 13:02.36 | timeshell | I've become less familiar with Linux commandline tools since I changed jobs and had less time for them |
| 13:05.01 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pgmlyatwvypbibtw) |
| 13:05.11 | *** join/#asterisk dfamorato (~dfamorato@2001:470:5:630:cabc:c8ff:fe9b:26b7) |
| 13:05.23 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pgmlyatwvypbibtw) |
| 13:09.47 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 13:10.48 | timeshell | The strange thing still is that while I'm packet sniffing on yet another switch on the affected VLAN, the problem doesn't reappear |
| 13:11.01 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:11.12 | timeshell | It's almost like my sniffing is forcing arp'ing to work |
| 13:11.19 | timeshell | Does that make sense? |
| 13:13.11 | timeshell | The polycom phones are making gratuitous arps. |
| 13:13.18 | irroot | what color pill you take timeshell ?? |
| 13:13.31 | timeshell | eh? |
| 13:13.58 | irroot | the matrix :P |
| 13:14.06 | timeshell | Just saying what I see. :p |
| 13:15.47 | timeshell | So, why would the switch hold onto the arp reply unless it knows I'm listening for it on a different switch? |
| 13:19.24 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 13:19.31 | fullstop | Good morning and happy Friday! |
| 13:20.48 | fullstop | I'm trying to set up t38modem with asterisk and, as usual, I am downright mentally challenged when it comes to sip.conf and peers. |
| 13:21.38 | fullstop | I set up a peer with a host, a secret, a default user.. and I attempt to get t38modem to register.. |
| 13:21.44 | fullstop | but receive this: Registration from '<sip:T38modem:tfaxmodem@127.0.0.1>' failed for '127.0.0.1' - No matching peer found |
| 13:22.01 | fullstop | Is it attempting to use '127.0.0.1' as the user name? |
| 13:22.55 | timeshell | no |
| 13:23.02 | timeshell | Server address |
| 13:23.17 | timeshell | T38modem is username |
| 13:23.22 | timeshell | tfaxmodem is password |
| 13:23.25 | fullstop | Which is what I have specified.. |
| 13:23.35 | fullstop | in the [T38modem] peer.. |
| 13:23.42 | fullstop | so I am a bit perplexed |
| 13:23.48 | irroot | fullstop t38modem v1 / 2 ?? |
| 13:23.53 | fullstop | v2 |
| 13:24.20 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 13:24.26 | irroot | /usr/bin/t38modem --listenport 1722 -u t38modem --old-asn -i 192.168.25.1 -g 192.168.25.1 -D ulaw -P alaw\ |
| 13:24.42 | irroot | that --old-asn is important |
| 13:24.59 | irroot | oh scratch that you using opal/sip |
| 13:25.16 | fullstop | Yeah |
| 13:25.23 | fullstop | old-asn is not an option for me |
| 13:26.33 | fullstop | sip exchange: http://pastebin.com/DtvXEMUy |
| 13:27.57 | fullstop | and the peer definition output from the cli: http://pastebin.com/w9f9zQrG |
| 13:28.40 | fullstop | I am not allowing guest sip sessions |
| 13:34.20 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 13:47.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.15 | timeshell | wdoekes2 Asking you because you've been so helpful. So, why would the switch hold onto the arp reply unless it knows I'm listening for it on a different switch? |
| 13:56.46 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
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| 14:00.50 | Linuturk | Asterisk 1.8.2.4 When trying to dial certain 1800 numbers, the call never connects. For one number, it will connect after a delay of about 45 seconds. Other toll free numbers disconnect. Still other toll free numbers work just fine, with no delay. On the ones that delay, I've noticed Progress cause code 127 in the cli. This is a Freepbx system (2.9) and we can't quite figure out the issue. I've tried from different phones, and even redirected an |
| 14:00.57 | jacc0 | timeshell: you could try to use tshark on the linux commandline; you can open the dump with wireshark |
| 14:01.24 | jacc0 | tshark -R "sip || rtp" -V -w output.cap |
| 14:02.00 | WIMPy | Linuturk: What do you mean by "Progress cause code 127"? |
| 14:02.37 | Linuturk | WIMPy: I see that in the asterisk cli when the call is trying to connect. apparently, it means an internetworking issue |
| 14:02.48 | Linuturk | it's a code returned by the pri |
| 14:03.27 | Linuturk | http://www.google.com/search?aq=f&sourceid=chrome&ie=UTF-8&q=PROGRESS+with+cause+code+127+received << I've been all over this page trying the various things |
| 14:03.29 | WIMPy | Ok, so it's got nothing to do with progress, but it just terminates for an unknown reason. |
| 14:04.07 | Linuturk | well, that's the last message I see when it is trying to connect, so that's where I started researching |
| 14:04.15 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:04.23 | Linuturk | but, I'm going bald pulling out my hair on this |
| 14:05.05 | WIMPy | Maybe your telco lost connectivity? |
| 14:05.18 | Linuturk | I had them check their switches and such |
| 14:05.24 | Linuturk | they report green lights on their end |
| 14:06.05 | WIMPy | If the call takes time to terminate, that doesn't look like an issue at your end. |
| 14:06.44 | Linuturk | well, this started happening right after a pbx switchout |
| 14:06.50 | Linuturk | asterisk 1.2 to 1.8 |
| 14:06.55 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 14:07.01 | Linuturk | and, it's happening for a few numbers |
| 14:07.06 | Linuturk | not just the one in that log |
| 14:07.12 | catphish | will asterisk accept incoming sip invites from totally unknown peers? |
| 14:07.22 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 14:08.22 | WIMPy | Linuturk: Maybe you're sending something you didn;t do before, but the real issue seems to be between your telco and the destinations telco. |
| 14:08.44 | Linuturk | why would it work before the switchout then? |
| 14:08.53 | WIMPy | You can try pri debug to find out what exactely you're sending and if something looks out of place. |
| 14:09.09 | Linuturk | pri debug is at the end of my original comment |
| 14:09.16 | Linuturk | PRI debug: http://pastebin.com/npe0wjMn SIP&PRI Debug: http://pastebin.com/0h4fXKaL |
| 14:11.48 | WIMPy | debug 2 would be easier to read. |
| 14:12.02 | Linuturk | ok |
| 14:12.36 | WIMPy | But it does not look like there's anything out of place. |
| 14:14.00 | WIMPy | Ok. Looks like a bug. |
| 14:14.29 | *** join/#asterisk atheos (~atheos@208.119.68.16) |
| 14:14.34 | WIMPy | It's Asterisk that terminates the call. |
| 14:14.43 | Linuturk | http://pastebin.com/EAbKwPdq << debug 2 |
| 14:14.50 | WIMPy | Must be a misinterpretation of that Progress message. |
| 14:15.16 | Linuturk | ok |
| 14:15.20 | Linuturk | that's just peachy |
| 14:15.30 | Linuturk | at least it isn't my fault! |
| 14:16.31 | WIMPy | Ah, two calls. |
| 14:16.46 | Linuturk | the system is really busy right now |
| 14:16.56 | Linuturk | so, anything I give you might have two calls in it |
| 14:17.07 | Linuturk | at night I can get cleaner logs |
| 14:17.12 | Linuturk | but, you know how it goes |
| 14:17.34 | WIMPy | It's ok, I just didn't see that in the first moment. |
| 14:18.10 | WIMPy | Looks like you should try a different libpri. |
| 14:18.23 | Linuturk | how do I check what version I currently have |
| 14:18.24 | Linuturk | ? |
| 14:19.00 | WIMPy | pri show version |
| 14:19.43 | Linuturk | <PROTECTED> |
| 14:20.03 | Linuturk | not much different |
| 14:20.36 | WIMPy | Try 1.4.12 and see if its' fixed there. |
| 14:21.04 | Linuturk | is there a changelog or something I can look at for the different versions? |
| 14:21.18 | Linuturk | and, I'm assuming I should do this after hours |
| 14:21.19 | *** join/#asterisk RickB17 (~RickB17@c-69-242-23-245.hsd1.pa.comcast.net) |
| 14:22.15 | WIMPy | It's in the download directory. |
| 14:23.57 | *** part/#asterisk gokulnath (~chatzilla@115.111.177.122) |
| 14:25.02 | WIMPy | But it's not the first time, I see libpri taking a call down without any need. |
| 14:26.10 | Linuturk | ok |
| 14:35.41 | catphish | is it possible to configure multiple sip peers to the same host, and then match one based on the number in the invite rather than the IP? |
| 14:36.43 | catphish | ideally i want different numbers originating from the same host to go into different contexts |
| 14:37.48 | WIMPy | No. That only works with type=user. |
| 14:38.10 | WIMPy | Or you have to do it in the dialplan. |
| 14:40.31 | catphish | thanks WIMPy I suspected that was the case |
| 14:40.49 | catphish | my remote thinks it's too important to have to authenticate itself |
| 14:40.54 | catphish | so i will have to use a dialplan |
| 14:43.36 | WIMPy | I have no idea if that works, but you could try a user without password. |
| 14:45.29 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:48.52 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:51.12 | *** join/#asterisk ssureshot (~digitolx@rrcs-24-142-174-146.central.biz.rr.com) |
| 14:54.26 | fullstop | Does anyone here use t38modem? I've got around my initial woes.. |
| 14:54.44 | fullstop | but it looks like it is using g.711 without attempting t.38 first |
| 14:55.19 | *** join/#asterisk mcreedjr (434d72cc@gateway/web/freenode/ip.67.77.114.204) |
| 14:56.34 | mcreedjr | I'm having issues with delayed MWI to Polycom 330s on Asterisk SVN intermittently. I have added pollmailboxes to voicemail.conf. Any other ideas? |
| 15:05.02 | *** join/#asterisk gxdssoft (~gxdssoft@190.235.168.112) |
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| 15:32.41 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 15:33.37 | ollii | hey, im using asterisk 1.4.40 and someone used my voicemailbox to set up a call to some countries...so my question is: is it possible to deactivate transfer via voicemail system? |
| 15:33.45 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pgmlyatwvypbibtw) |
| 15:39.28 | ssureshot | can music on hold play mp3 files by default? |
| 15:42.10 | russellb | ssureshot: if you install format_mp3, yes. |
| 15:43.58 | ssureshot | russellb: format mp3 isn't an option in make menuconfig? |
| 15:45.11 | Qwell | ollii: Do you have a dialout context specified? |
| 15:45.40 | Qwell | or perhaps an operator context with poor includes |
| 15:45.45 | ollii | ssureshot: yes it is, in contrib is a script to fetch the needed sources |
| 15:46.18 | ollii | Qwell: yes i have a context called internal which includes a dialout context ... now i've learned that this isnt a really good idea |
| 15:47.23 | russellb | ssureshot: it's in the addons section |
| 15:49.16 | ssureshot | ah, let me take a look and see if I can figure this out right quick.. thank you both :) |
| 15:50.06 | ssureshot | and yes I do see it now... hmm now I have to figure out how to compile that in to a working system |
| 15:50.54 | ssureshot | do I have to run make and make install again? |
| 15:51.46 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 15:53.07 | Qwell | yes |
| 15:54.11 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 15:55.21 | ssureshot | roger that I think I'm on my way again... now to run some tests.. |
| 15:56.29 | ssureshot | I just read a post that said the modules directory needed removed before "make install" is this stil the case? |
| 15:56.56 | ssureshot | I've already moved past that just hoping that doesn't hurt me |
| 15:57.35 | Qwell | ssureshot: it's always a good idea to, sure |
| 15:57.44 | Qwell | I actually just do `make uninstall install`, and it takes care of that for me |
| 15:58.05 | Qwell | I switch between versions a lot though, so YMMV. |
| 15:58.48 | ssureshot | roger that |
| 15:59.19 | ssureshot | I want this to be the last reload before the live tests start so I'm trying to be careful and have the system as clean as possible |
| 15:59.56 | *** part/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 16:10.30 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:11.39 | *** join/#asterisk generalhan (435a4002@about/windows/staff/generalhan) |
| 16:13.55 | generalhan | hey guys ... im having an issue with my PRI after our services went down. my DAHDI status is showing all PRI lines as being in use, "1 T1 Clear (In use) (SWEC: MG2)" <--- for all 23 channels. even after a restart of all services, and the machine itself. would most of you assume that this is an issue with the provider ? |
| 16:18.45 | WIMPy | What status? Do 'dahdi show status'. |
| 16:21.16 | generalhan | they all show up in the asterisk console ... but when checking them from the system CLI (not asterisk console) is when they all show as in use |
| 16:21.57 | WIMPy | In use by asterisk, yes. |
| 16:23.44 | seanbright | "No session refresh received after 1768s (expiration period=1800s), stopping session now!" |
| 16:23.54 | seanbright | should asterisk be sending something to my UA that it's not? |
| 16:24.25 | russellb | the UA says that? |
| 16:24.28 | seanbright | yeah |
| 16:24.55 | russellb | then yeah. you can turn session timers off in sip.conf so that they won't even be negotiated if it's screwing up |
| 16:25.39 | generalhan | the issue is that all calls are showing congestion hangup cause 34 circuit busy |
| 16:25.48 | seanbright | i guess i could disable timers on the client side as well |
| 16:26.06 | generalhan | i checked with the provider to see if they left the hardware in a loopback for testing and they assure me that all is up and running on their end |
| 16:26.09 | *** join/#asterisk irroot (~irroot@41.54.197.228) |
| 16:33.46 | *** join/#asterisk afink (~chatzilla@wsip-98-172-137-175.om.om.cox.net) |
| 16:35.32 | *** join/#asterisk nW44b_ (~Schnitzel@unaffiliated/benwa) |
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| 16:41.25 | afink | is this the correct basic configuration for a dual span t1? http://pastebin.com/smtny5ig |
| 16:49.44 | ssureshot | when I press the hold button on my cisco 7940, does that symbolize a parked call? |
| 16:50.19 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 16:51.22 | paulc | ssureshot: no - parking is not the same as holding |
| 16:51.46 | ssureshot | ah, |
| 16:51.59 | paulc | ssureshot: parking typically means "holding within the system, to be retrieved from another extension, typically by dialing a code (or parking slot number)" |
| 16:52.15 | paulc | curses his double use of the word typically (it's too early still) |
| 16:53.40 | ssureshot | ah so as long as I get the music playing when I dial the default of 700 within asterisk.. I assume I would transfer the call to an extension to park them? |
| 16:55.06 | *** join/#asterisk tcpip4000 (~fulano@200.118.180.111) |
| 16:55.06 | paulc | Usually it works like this... you (A) are on a call talking to (B). You want to park the call. You hit transfer and dial 700. The system says "7.. 0.. 1.." then you get music. |
| 16:55.18 | paulc | You hang up, and the caller (B) is parked, with nice music on hold. |
| 16:55.32 | paulc | Now you walk over to Sally's desk.. and dial 701.. and now Sally's phone is connected to caller (B) |
| 16:59.23 | *** join/#asterisk tcpip4000 (~tcpip4000@200.118.180.111) |
| 17:01.12 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
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| 17:12.13 | *** part/#asterisk tcpip4000 (~tcpip4000@200.118.180.111) |
| 17:16.08 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:16.55 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:21.06 | hardwire | would be all goobery inside if somebody knew of a quick FastAGI <-> HTTP gateway |
| 17:24.12 | RickB17 | I currently support my companies asterisk installation, but am moving on. Does anyone know of a good hosted SIP solution that allows call queueing with agents and such? |
| 17:25.32 | hardwire | Some of those can get complicated. I can do remote maint if you can't find a good online resource. |
| 17:25.42 | hardwire | <- shameless |
| 17:25.49 | RickB17 | yeah i could offer that too on the side |
| 17:25.55 | irroot | RickB17 its a bit iffy to rely on data link well here its stupid :P |
| 17:25.57 | RickB17 | but they want it to be "hands-off" |
| 17:26.15 | _Corey_ | Rickb17: You may want to look into hosted Switchvox or something like that |
| 17:26.26 | hardwire | That might work as well. |
| 17:26.36 | hardwire | I haven't diven into switchvox at all. |
| 17:27.00 | paulc | I talked to a guy yesterday who bought service from rentpbx.com - might be worth a look |
| 17:27.20 | paulc | but would maybe require them to still maintain it, rather than "call helpdesk for changes" etc |
| 17:28.02 | _Corey_ | There are providers that offer full-service asterisk and switchvox out there |
| 17:31.53 | *** join/#asterisk sourcode (~code@ppp-58-8-238-17.revip2.asianet.co.th) |
| 17:37.42 | The_REV | hey yall... does anyone have a good write up on howto get ztdummy working in a XEN virtual machine? |
| 17:46.00 | irroot | The_REV ztdummy is deprecated as is dahdi_dummy |
| 17:46.29 | The_REV | ok... how do I get my meet me rooms to work then? |
| 17:46.30 | irroot | you should find plenty info on voip-info still |
| 17:46.48 | irroot | the new versions of asterisk dont requrie it |
| 17:46.56 | irroot | there alternate timing modules |
| 17:47.34 | The_REV | Ive been looking there and cannot find anything... and the system is on 1.6 and cannot be moved to 1.8 |
| 17:49.27 | The_REV | but thank you for responding irroot |
| 17:49.53 | irroot | 1.6 should be fine look at res_timing** |
| 17:50.30 | The_REV | ok ill do some research |
| 17:52.25 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 17:52.47 | jmls | anyone know where I can get SIP firmware for Samsung smt-i3100 ? |
| 17:55.07 | generalhan | using Xorcom Astribank with a T1, status shows no alarms and all 23 channels as "in use", but all outgoing calls fail with hangup cause 34 and circuit busy/ congested. Xorcom unit has a steady blinking light for the TE port from my T1 which seems to me should mean that it is all up and working. Provider is telling me that their end is perfect. can anyone suggest something to test to verify |
| 17:55.07 | generalhan | that i am good on my end ? |
| 18:04.00 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 18:07.07 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 18:07.22 | kuku | HELP. I've got one way audio after switching to a different ISP. Any suggestions ? |
| 18:08.16 | irroot | mmm |
| 18:08.22 | irroot | kuku sip ? |
| 18:08.54 | kuku | yes |
| 18:08.57 | irroot | nat ? |
| 18:09.09 | kuku | inside out doesnt work. outside in voice works |
| 18:09.10 | kuku | yes |
| 18:09.18 | kuku | It was natted before |
| 18:09.42 | kuku | I change externalip= in sip_nat.conf |
| 18:12.36 | irroot | mmm they might be natting it on there path seen this on wi-fi routes b4 |
| 18:16.13 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 18:16.43 | kuku | k |
| 18:16.45 | kuku | thanks |
| 18:17.19 | acidfoo | hey folks, i'm getting an error with Dial(), not sure what would be the correct way of using it... |
| 18:17.22 | acidfoo | [Jul 8 14:09:56] -- Executing [dial_from_queue@user:5] Dial("Local/1113@default-958d;2", ",30,") in new stack |
| 18:17.23 | acidfoo | [Jul 8 14:09:56] WARNING[16917]: app_dial.c:1923 dial_exec_full: Dial requires an argument (technology/number) |
| 18:17.58 | acidfoo | I don't know what is the purpose of the "-958d;2" part |
| 18:18.02 | irroot | kuku set up a soft client out side network and use that to register to server / test |
| 18:18.36 | leifmadsen | acidfoo: for that error, that means nothing (It's a unique channel identifier) |
| 18:18.41 | leifmadsen | the error you're getting is because you're using Dial() wrong |
| 18:19.09 | leifmadsen | essentially, you're not passing anything to Dial() (except a timeout) so it has no idea what end point to try calling |
| 18:19.12 | acidfoo | what would be somthing that look good |
| 18:19.19 | acidfoo | ah ok |
| 18:19.29 | leifmadsen | "what would be somthing that look good" does not parse |
| 18:19.30 | irroot | leifmadsen [i typed slow] how i get hold of your book ?? |
| 18:19.32 | acidfoo | so the ",30," |
| 18:19.45 | acidfoo | would be -> "something,30,something" |
| 18:19.47 | leifmadsen | infobot: tell irroot about thebook |
| 18:19.56 | irroot | ~thebook |
| 18:19.56 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 18:19.57 | leifmadsen | acidfoo: core show application Dial |
| 18:20.12 | leifmadsen | irroot: you got a msg from infobot :) |
| 18:20.16 | irroot | lol problem is ebay/amazon do not deliver |
| 18:20.49 | acidfoo | good |
| 18:21.10 | generalhan | ok setting pri debug on span 1 still gives me no interesting information other than that the channel is circuit-busy and everyone is busy/congested at this time. |
| 18:21.25 | generalhan | provider is being of no help and i swear everything on my end is setup properlly |
| 18:21.50 | ectospasm | generalhan: if you never receive or send a SETUP message, the call won't begin |
| 18:22.10 | ectospasm | ...if "pri show spans" doesn't show span 1 as Provisioned, Up, Active, then it won't work. |
| 18:22.21 | leifmadsen | irroot: don't feel bad -- I haven't gotten my copies yet either |
| 18:23.03 | *** join/#asterisk jo3sm1th (~IceChat7@66.0.165.57) |
| 18:23.34 | generalhan | ectospasm: PRI span 1/0: Provisioned, Up, Active |
| 18:23.48 | jo3sm1th | Anyone here do technicak support? I need help configuring/installing asterisk on CentOS to work with flowroute, I can pay with Paypal. |
| 18:24.28 | ectospasm | generalhan: are these inbound or outbound calls you're testing with? |
| 18:24.40 | irroot | jo3sm1th mmm intresting concept |
| 18:24.56 | irroot | set up paypal theother day have not used it |
| 18:25.41 | generalhan | ectospasm: outbound ... inbound shows no data at all before the "all circuits are busy" or "no route found" messages |
| 18:26.00 | ectospasm | generalhan: do you see the outbound setup message? |
| 18:26.12 | ectospasm | er, SETUP message? |
| 18:26.23 | jo3sm1th | It would only take a skilled person 10 minutes to do it, I got like $50 Paypal right now |
| 18:27.47 | ectospasm | generalhan: essentially, you should see >SETUP, <CALL PROGRESS or <ALERTING... then a <DISCONNECT with a cause code. |
| 18:29.32 | seanbright | russellb: wasn't asterisk's fault |
| 18:29.41 | seanbright | russellb: twas firewall |
| 18:31.02 | russellb | seanbright: lamez0r |
| 18:31.34 | seanbright | agreed |
| 18:34.57 | *** join/#asterisk serafie (~erin@nat/digium/x-ysiboyhhdyodwenz) |
| 18:38.30 | jo3sm1th | is VIM and Nano both to accomplish the same thing? |
| 18:38.54 | ectospasm | jo3sm1th: yes |
| 18:39.03 | ectospasm | jo3sm1th: they're both text editors |
| 18:39.25 | ectospasm | nano is more nub friendly |
| 18:39.41 | ectospasm | ...closer to notepad.exe than vim |
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| 18:44.16 | generalhan | ectospasm: i dont see anything like that in my debug |
| 18:45.26 | jo3sm1th | yum install asterisk ... why doesnt this work to install asterisk? |
| 18:46.33 | TuNoD_68 | @jo3sm1th: it does if you have the repo setup |
| 18:47.14 | jo3sm1th | i did... |
| 18:47.19 | jo3sm1th | http://codeghar.wordpress.com/2009/03/08/how-to-install-asterisk-on-centos-package-installation/ |
| 18:48.29 | TuNoD_68 | @jo2sm1th: you might want to give it a version to install.... try yum list asterisk\* and pick the right one :) |
| 18:49.53 | *** join/#asterisk serafie (~erin@nat/digium/x-rjzjwexfprprqkzi) |
| 18:50.49 | jo3sm1th | ok im trying... |
| 18:51.24 | TuNoD_68 | @jo2sm1th: I think they are up to v16 or something.... so yum install asterisk16 would be better... |
| 18:52.01 | TuNoD_68 | @jo2sm1th: and I would trust the howto from the source better http://www.asterisk.org/downloads/yum |
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| 18:53.33 | paulc | jo3sm1th: Is there a particular reason you don't want to install the latest version from source? |
| 18:54.39 | jo3sm1th | tried and failed |
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| 18:56.06 | wcselby | o/ |
| 18:59.02 | jo3sm1th | ok so i finally installed asterisk but the stop now command is not there |
| 18:59.17 | jo3sm1th | is there another way to stop it |
| 18:59.49 | wcselby | core stop now |
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| 19:05.06 | irroot | jo3sm1th "core stop" |
| 19:12.57 | generalhan | here is the output from my pri debug while trying to make an outbound call ... i still cannot see a reason why all the circuits are busy: http://pastebin.com/UUdN9cJm <--- if anyone has some insight i would be most greatful |
| 19:15.42 | irroot | generalhan cause 34 ... network error has it been activated ?? |
| 19:16.12 | irroot | in lusaka they require a valid CLI be provided before passing the signal |
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| 19:36.40 | ssureshot | on my current system I'm replacing when I press the hold button on the phone music plays,, (cisco 7940/7960's) the configurations have been copied over and I've updated the SetMusicOnHold() with Set(CHANNEL(musicclass)=default,, Ive set the musiconhold.conf to http://pastebin.com/5amSJbwZ |
| 19:37.12 | ssureshot | I also have format_mp3 enabled,, but I'm stumped on why the music wil not play |
| 19:37.33 | ssureshot | oh and madplay is installed,, I've tried this with and without this option |
| 20:01.38 | jaytee | I've got a discrepancy between the minute charges from Flowroute's CDR records and my server MYSQL CDR records. I have one call that supposedly lasted 6 hours according to Flowroute and was only a 200 second call according to mysql. |
| 20:01.58 | jaytee | anyone ever experienced something similar? |
| 20:03.47 | *** join/#asterisk chazzam (~chazzam@nat/digium/x-olgttzlvdwzlufcz) |
| 20:04.15 | generalhan | and out of no where ... the phone are working again ... |
| 20:04.42 | generalhan | my service provider must have finally realized that it WAS their fault and fixed it ... though they would never admit to that ! lol |
| 20:06.20 | The_REV | ok im at wits end... I cant get MOH or MeetMe to work and it seems like a timing issue cause its on a virtual system... I used to use ztdummy to fix this issue but Ive been told thats depreciated... Im on Asterisk 1.6.2.18.2 CentOS (kernel 2.6.18-194.11.4.el5xen) yes its running on a XEN server and no I cannot upgrade to 1.8 or move to real HW... any ideas or suggestions would be awesome... |
| 20:08.31 | nightrid3r | The_REV can't dahdi do the trick ? |
| 20:08.46 | beek | jaytee: the call was routed through the Bermuda Triangle. |
| 20:09.29 | The_REV | nightrid3r: I cant seem to get that to make right (im taking over this box it was built funny) |
| 20:11.06 | wcselby | The_REV- you should be able to get at least DAHDI 2.4.x to do all that for you...., but if not, you could try compiling asterisk with one of the other timer modules - that should at least help you with MOH. MeetMe requires DAHDI for mixing (not for the timer) |
| 20:11.37 | wcselby | starting in I think DAHDI 2.3, they removed dahdi_dummy and just made it part of the core of dahdi |
| 20:12.59 | The_REV | ok let me go poke it for a sec... ill be back if it doesnt work... |
| 20:16.26 | wcselby | jaytee- check with flowroute - I know they sent me a notice recently about some network issues they were having. maybe the call happened around that time? |
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| 20:28.39 | The_REV | OK i got dahdi installed and it seems to be running but my meetme are still exiting non-zero... |
| 20:31.15 | The_REV | ok there might be an issue.... asterisk console cannot pull a status or show version on dahdi |
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| 20:46.33 | wcselby | The_REV- did you recompile asterisk after you installed dahdi |
| 20:47.01 | The_REV | no... but I think its not reging a /dev |
| 20:47.20 | wcselby | you need to recompile asterisk for it to recognize that you've installed dahdi |
| 20:47.31 | wcselby | go to your asterisk source directory |
| 20:47.34 | wcselby | make clean |
| 20:47.36 | wcselby | ./configure |
| 20:47.45 | wcselby | and then make menuselect and so on |
| 20:48.00 | wcselby | otherwise the dahdi stuff won't be compiled into asterisk |
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| 20:49.39 | The_REV | ok |
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| 21:06.40 | The_REV | great built from repository... *head -> desk* |
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| 21:14.50 | godmachine-x6 | im working on getting google voice to working with asterisk so that i can use a SIP phone to make/receive calls through google voice. now i have everything configured as stated in the guides, but i don't know what to do now with my softphone. im using ekiga. can someone help? thanks |
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| 21:19.48 | RickB17 | anyone here want a google + invite? |
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| 21:30.38 | nightrid3r | RickB17 yeah |
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| 22:07.11 | godmachine-x6 | can someone here please help me get this asterisk + google voice to working? i have followed the steps online, just can't get it to dialout with my SIP phone. |
| 22:11.09 | wcselby | godmachine-x6- what's it doing? |
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| 22:19.56 | godmachine-x6 | when i try to dial with ekiga it says "user is offline" |
| 22:20.04 | godmachine-x6 | or not available |
| 22:20.29 | godmachine-x6 | i have to go mow my yard right now, but when i get done if you don't care i'll have you help me look at it |
| 22:21.37 | wcselby | if i'm still here |
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