| 00:05.46 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:08.14 | *** join/#asterisk fagiani (bb28e727@gateway/web/freenode/ip.187.40.231.39) |
| 00:19.43 | fagiani | hi, can anyone help pointing a way i can understand why I can't originate nor receive calls with my digium card that is connected to the E1 channel and is being detected with dahdi_scan and dahdi_hardware? is there any details I should be looking into? I appreciate any help! |
| 00:20.33 | fagiani | could that be due to the signaling and how can i debug that? |
| 00:21.13 | WIMPy | What does Asterisk say about it? |
| 00:21.25 | WIMPy | 'dahdi show status' |
| 00:24.29 | fagiani | in the asterisk -r console it says: No such command dahdi |
| 00:24.59 | WIMPy | Looks like you don't have dahdi loaded. |
| 00:25.25 | WIMPy | Either it's missing or it can't load doe to a bad config. |
| 00:25.47 | WIMPy | Try to load it manually. 'module load chan_dahdi'. |
| 00:26.02 | WIMPy | Possibly increasing debug/verbose first. |
| 00:26.35 | fagiani | bad config is possible due my trials but i can reset it and try reloading |
| 00:30.22 | fagiani | I reset the dahdi config to default and now i have this as result: |
| 00:30.29 | fagiani | Digium Wildcard TE110P T1/E1 Card 0 REC 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) |
| 00:30.46 | fagiani | but when i call to the main number i get busy signal |
| 00:30.58 | fagiani | and when i try calling out it says all circuits are busy |
| 00:31.18 | fagiani | i believe that this is due to signaling maybe? |
| 00:31.35 | WIMPy | You're not even getting that far. |
| 00:31.41 | fagiani | ok |
| 00:31.49 | WIMPy | We need an "OK" under Alarms first. |
| 00:32.00 | fagiani | can you suggest me something? |
| 00:32.17 | fagiani | what does REC mean? |
| 00:32.27 | WIMPy | So it's about line code or there is something else wrong somewhere. |
| 00:32.54 | WIMPy | I think it's RECovering. |
| 00:33.33 | WIMPy | Not something you usually see for more than a very short time. |
| 00:34.04 | fagiani | ok, now it is BLU/REC |
| 00:34.11 | fagiani | and then back to REC |
| 00:34.30 | WIMPy | Check the LEDs on the NT for alarms there. |
| 00:35.35 | fagiani | sorry, but what you mean by NT? is it the modem? the E1 card? |
| 00:35.35 | WIMPy | Maybe try a different cable. |
| 00:35.52 | WIMPy | The modem. |
| 00:37.09 | fagiani | the modem has only green leds, no one blinks all lit: PWR 104(RD) 103(TD), E1, DSL |
| 00:37.56 | fagiani | the digium card thou is alternating green and red |
| 00:38.01 | fagiani | very quickly |
| 00:38.50 | WIMPy | I haven't seen what happens in that situation, but I guess that could be a wrong line code. |
| 00:39.52 | fagiani | sorry if it sounds too basic but what is a wrong line code? |
| 00:40.14 | WIMPy | The stuff in dahdi/system.conf. |
| 00:40.23 | fagiani | if i switch the bnc's it only blinks red and all 103 and E1 modem leds turn off |
| 00:41.02 | fagiani | is that related to possibly ISDN or R2 signaling or it is nothing to do with that? |
| 00:41.07 | WIMPy | BNC? wow. |
| 00:41.33 | WIMPy | No, that's in asterisk/chan_dahdi.conf. |
| 00:41.53 | fagiani | the modem is brand new, they just installed here, but they said i need to use two coaxial cables and a balun to adapt with digium card |
| 00:42.12 | fagiani | so i got two coaxial cables and bnc adapters |
| 00:42.21 | fagiani | and the balun |
| 00:43.32 | fagiani | would the loadzone and default zone be that? |
| 00:43.46 | tzanger | whoa someone's actually doing radio? |
| 00:43.50 | WIMPy | No, they are only for tones. |
| 00:43.52 | fagiani | it reads us and i am in brazil |
| 00:44.02 | tzanger | oh e1 |
| 00:44.05 | tzanger | not as much fun :-( |
| 00:44.12 | WIMPy | It's the span= line. |
| 00:44.36 | WIMPy | Maybe it doesn't like the balun? |
| 00:44.37 | fagiani | span=1,1,0,ccs,hdb3,crc4 |
| 00:44.43 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 00:44.44 | fagiani | this is what reads there |
| 00:45.00 | WIMPy | That's the usual configuration. |
| 00:45.16 | fagiani | i have seen forums where it reads span=1,1,0,cas,hdb3 |
| 00:45.28 | fagiani | but then when i get into console dahdi module doesn't load |
| 00:46.31 | WIMPy | If the E1 LED on the NT is green, it seems to receive the signal from the card. |
| 00:46.51 | WIMPy | But the card seems to have trouble receiving something sensible. |
| 00:48.01 | fagiani | the cable i have here is like rca with BNC adapter |
| 00:48.27 | fagiani | wich gets between the modem and balun |
| 00:49.07 | WIMPy | Have you tried to awap them around? Maybe one of them has issues. |
| 00:49.56 | fagiani | yes, i had them all possibilities... |
| 00:50.01 | fagiani | only one locks the modem green |
| 00:50.08 | fagiani | other gets the leds off |
| 00:50.43 | fagiani | unfortunately the modem didn't come with the cables and i am not sure if there is a very very specific one |
| 00:51.32 | fagiani | i just got one that i think works as it has bnc adapters at the end, but they are like accessories... |
| 00:52.18 | fagiani | so i can unhook them off from the cable's end and then the cable looks like a video (signal) cable |
| 00:52.31 | fagiani | like RCA end if that makes sense to you |
| 00:52.59 | fagiani | like those we use in DVD but thicker |
| 00:56.43 | *** join/#asterisk greenwolf (4cb4d9fd@gateway/web/freenode/ip.76.180.217.253) |
| 00:58.03 | jpsharp | So you go from RJ45 on the E1 card to the balun and out of the balun comes BNC connectors? |
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| 01:16.45 | fagiani | right, the modem is brand new but the local operator says this is the standard, |
| 01:16.53 | fagiani | the modem has two bnc connectors |
| 01:17.20 | fagiani | and they say i need a cable to connect and a balun to connect to the E1 card |
| 01:17.35 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 01:18.17 | fagiani | as they say many buy cards that are BNC based, I never seen those but i guess this is related to the analog E1 on traditional PBX |
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| 01:30.27 | jpsharp | I've seen E1s delivered by coax, RJ45, and by DB-15. |
| 01:32.00 | WIMPy | There is no analog E1. |
| 01:32.34 | WIMPy | But if the cables aren't specific for that purpose, they might have the wrong impedance. |
| 01:33.08 | WIMPy | And I would have sworn, that BNC is something you might see on T1, but not on E1. |
| 01:33.20 | jpsharp | The baluns for E1s go from 75 ohm coax to 120 twisted pair. |
| 01:33.43 | jpsharp | I've never seen a T1 delivered by coax. T3, yes, but never T1. |
| 01:35.23 | WIMPy | I haven't seen BNC on telephony equipment so far, only on networking stuff. |
| 01:35.52 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 01:37.08 | jpsharp | But for fagani's problem, he'll need a G.703 balun. |
| 01:38.32 | jpsharp | And fagiani too. |
| 01:44.55 | fagiani | I can send you a picture if you want :) |
| 01:45.47 | fagiani | I am trying to build a thicker coaxial cable here as our region is not well served with shops that can do that and while the proper cable arrives. |
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| 02:33.50 | slav3_kitten | i think i may be pathetic |
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| 02:35.08 | slav3_kitten | got my hot british girl here and we just sat around chatting with each other on irc while drinking tea, writing software, and watching firefly |
| 02:35.19 | slav3_kitten | i may be going about things wrong |
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| 03:11.43 | fagiani | jpsharp i changed both balun and cable and still have the digium card green/red falternating quickly do u know what can i try else? |
| 03:20.12 | fagiani | WIMPy, I was able to get the Alarm OK |
| 03:22.10 | fagiani | but still i can't receive or originate calls, do you know what next step is? |
| 03:30.27 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:43.01 | jpsharp | Now its a matter of configuring Asterisk to talk across that circuit. Have you configured your PRI settings (if it is a PRI) or other signalling if neeed? |
| 03:43.04 | jpsharp | needed? |
| 03:56.23 | fagiani | jpsharp i believe it is a R2 signaling but i am not sure |
| 03:56.50 | fagiani | the defaults are not working and i believe they are PRI (is this ISDN signaling?) |
| 04:05.12 | fagiani | jpsharp do u know if i change from ccs to cas it needs openr2 to be installed? when i change that dahdi won't load in asterisk anymore |
| 04:22.05 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 04:25.40 | *** join/#asterisk ChrisInSydneyToo (~Chris@60-242-81-231.tpgi.com.au) |
| 04:27.06 | jpsharp | Honestly, I don't know. I haven't installed Asterisk with R2 signalling. |
| 04:28.06 | jpsharp | But before you get wild and crazy installing things, verify your signalling type. Nothing worse than spending all day getting one type of signaling to build right only to need the other. |
| 04:31.17 | *** join/#asterisk flyingbull (~Adium@cpe-065-190-148-230.nc.res.rr.com) |
| 04:42.50 | *** join/#asterisk Yourname` (~whocares@unaffiliated/yourname/x-837320) |
| 04:46.10 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 04:46.51 | PhoenixMage | Hi guys, has anyone played with the Cisco Jabber Android client on Asterisk? |
| 04:58.25 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 05:17.04 | ChrisInSydneyToo | "Not I", said the pig |
| 05:17.39 | ChrisInSydneyToo | Got a curly one...like a piggys tail |
| 05:17.46 | ChrisInSydneyToo | chan_sip.c:14239 check_auth: Correct auth, but based on stale nonce received from... |
| 05:18.14 | ChrisInSydneyToo | ...<sip@address@wherever.org>;tag=stuff |
| 05:18.34 | ChrisInSydneyToo | anyone seen something like this before ?? |
| 05:18.53 | ChrisInSydneyToo | ast 1.8.9 |
| 05:18.56 | ChrisInSydneyToo | .0 |
| 05:19.14 | ChrisInSydneyToo | maybe time for a "certified" version ?? |
| 05:20.28 | ChrisInSydneyToo | The client is a proprietory PBX |
| 05:41.37 | jpsharp | stale nonce says someone's clock is wobbling. |
| 05:41.50 | jpsharp | Or you've got some hellacious lag between sites. |
| 05:42.35 | jpsharp | I saw a similar problem trying to get some Cisco 7940s to register across a long haul satellite link |
| 05:48.04 | ChrisInSydney | the client was partying like it was August 2011 |
| 05:48.33 | ChrisInSydney | just trying to remember how to set the SNTP on win08 |
| 05:48.37 | ChrisInSydney | arghhhh |
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| 06:17.49 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 06:23.50 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
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| 06:47.37 | bulkorok | hi |
| 06:55.18 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:07.01 | ChrisInSydney | jpsharp: didnt fix. Raised a ticket with the manufacturer. Now capturing SIP |
| 07:07.10 | ChrisInSydney | to saend them |
| 07:07.26 | ChrisInSydney | bulkorok: g'da |
| 07:07.32 | ChrisInSydney | g'day |
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| 07:26.26 | bulkorok | hi |
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| 07:27.13 | pithagorians | is it possible to use in allow more than one network in sip config? |
| 07:31.43 | ChannelZ | eh? |
| 07:32.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 07:34.33 | kaldemar | pithagorians: allow is for codecs, not ACL. are you thinking about deny/permit? |
| 07:35.17 | kaldemar | pithagorians: and yes, you can use more than one permit line. |
| 07:36.25 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 07:36.41 | pithagorians | i need it in db |
| 07:36.50 | pithagorians | divided somehow |
| 07:37.18 | pithagorians | and i thought it is possible by semicolon |
| 07:41.16 | ChannelZ | Supposedly yes |
| 07:43.39 | Chainsaw | So, from SIP firmware to "UC software" on a Polycom 670... it starts out okay but the phone seems to stop responding and I get sip_xmit errors from Asterisk. |
| 07:44.58 | Chainsaw | Same handset operates correctly on the SIP firmware load, which seems to absolve the network of any blame. |
| 07:46.13 | Chainsaw | I have a choice of "TCPpreferred" or "TCPonly", for which I set up Asterisk to have a transport of "TCP,UDP" or "TCP", respectively. |
| 07:46.22 | Chainsaw | That does not appear to alleviate the problem. Any thoughts? |
| 07:47.28 | Chainsaw | (During the initial window of about 3 minutes, I am able to receive inbound calls. Forever after, I can make outbound calls but Asterisk is unable to place any calls towards it or qualify the peer) |
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| 08:43.20 | rjvvliet | Are there any DAHDI 'Experts' available? got some problems with what i think is DTMF detection on PRI Port1 to Port2 Bridged calls, PBX on port2 does not seem to detect them sometimes. |
| 08:54.31 | *** join/#asterisk Dovid (d508763e@gateway/web/freenode/ip.213.8.118.62) |
| 08:55.19 | WIMPy | rjvvliet: That's always inband, so unless your audio gets garbled (eg by EC) there isn't much that could go wrong there. |
| 08:57.18 | rjvvliet | WIMPy: Hi, yeah i was afraid someone would mention garbled audio......mmmmm, got customers that go to the 2nd PBX but sometimes get stuk in the IVR there, the DTMF gets detected by Asterisk and send over port2, but seems to never get there. |
| 08:59.05 | WIMPy | I wonder if Asterisk should be able to detect DTMF on a bridged call. Are you recording or have some features enabled that keeps Asterisk in the audio path? |
| 09:02.12 | rjvvliet | WIMPy: whel, i say asterisk 'detects' i seem to really mean chan_dahdi detects...."chan_dahdi.c: Begin DTMF digit: 0x32 '2' on DAHDI/i1/-bcb2" |
| 09:03.01 | WIMPy | Ok, might be that DAHDI informs Asterisk, no matter who does the bridging. I'm not that deep in to DAHDI. |
| 09:03.25 | *** join/#asterisk shadebob (~shadebob@41.142.254.128) |
| 09:03.30 | WIMPy | just hopes it doesn't filter them as well. |
| 09:04.27 | WIMPy | But that reminds me that I had a similar issue. Or mabye have. Checking... |
| 09:05.05 | WIMPy | Have. :-( |
| 09:08.28 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:08.34 | rjvvliet | WIMPy: Sorry my responses are delayed got a power failure at a customers BRB...... ;-( |
| 09:08.50 | WIMPy | np |
| 09:11.18 | *** join/#asterisk Dennisvj (~dennis@unaffiliated/dennisvj) |
| 09:16.21 | *** join/#asterisk Malard_Office (~Malard_Of@xbmc/staff/malard) |
| 09:19.34 | rjvvliet | WIMPy: Pfeww, i'am back again. |
| 09:20.10 | rjvvliet | WIMPy: Whell it seems that dahdi is sending the DTMF also "chan_dahdi.c: Started VLDTMF digit '2'" right after the Begin DTMF |
| 09:26.53 | Malard_Office | hi, i'm trying to setup asterisk, and want to check that i'm correctly connected to my ISDN socket, is there a way to dial a number or log an incomming call in the asterisk command line or log? |
| 09:29.07 | rjvvliet | Malard_Office: i think the "core set verbose 5" and "core set debug 5" on the CLI should tell you enough. |
| 09:35.28 | Malard_Office | if every time i try and dial either the main number or one of the DDIs the line is just cut off |
| 09:35.37 | Malard_Office | and i see no log in asterisk |
| 09:35.48 | Malard_Office | would that indicate an issue between the ISDN box and the card? |
| 09:37.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:39.28 | rjvvliet | Malard_Office: i don't think so, be sure to have the MSN's set ip correctly if its ISDN BRI. |
| 09:39.51 | WIMPy | rjvvliet: Sounds interesting, but unfortunatly I never looked in to sending DTMF with dahdi. |
| 09:40.39 | WIMPy | Malard_Office: Yes. How are you connected? Hardware? Driver? |
| 09:40.54 | rjvvliet | WIMPy: mmm, sorry, ill see what we can find, i have one option in mind that is the echocancelwhen bridge, i think its enabled, so lets disable it ;-) |
| 09:41.06 | rjvvliet | WIMPy: Thanks for you thoughts... |
| 09:41.23 | WIMPy | rjvvliet: Yes, EC might play a role. |
| 09:42.20 | rjvvliet | WIMPy: got to go, customer site PBX will not come online after power failure..... |
| 09:42.49 | WIMPy | Are you sure? |
| 09:43.08 | WIMPy | maybe it's just fscking? |
| 09:44.15 | Malard_Office | rjvvliet i dont know what you m ean |
| 09:44.16 | rjvvliet | WIMPy: Whell, that i though to but 40 mins for fsck? on 500Gb disk? |
| 09:44.20 | Malard_Office | phone systems are new to me :S |
| 09:46.06 | rjvvliet | Malard_Office: if you are using BRI, than the device needs to know what DID it needs to respond, i may be wrong but thats my experience when * does not see the call. |
| 09:46.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:46.35 | Malard_Office | its an openvox b100 card |
| 09:46.57 | rjvvliet | Malard_Office: but i;am sorry, need to go to customer, hope someone can help you further. good luck. |
| 09:47.04 | WIMPy | rjvvliet, well, probably not then :-( |
| 09:47.06 | Malard_Office | and i've just got an ISDN 2 from BT (uk provider) i plugged a network cable into one of the 2 sockets provided on the termination plate and 1 into the back of the server |
| 09:47.15 | Malard_Office | okay rjvvliet thanks |
| 09:47.25 | WIMPy | rjvvliet: No you don;t need to tell Asterisk to get a log. |
| 09:47.51 | WIMPy | Malard_Office: Hardware? Driver? |
| 09:49.25 | *** join/#asterisk Sophira (~sophie@neo.theblob.org) |
| 09:49.32 | Sophira | Hello! |
| 09:50.06 | WIMPy | Malard_Office: Do you have something that displays the cause code when you call that line? |
| 09:51.28 | Sophira | I'm using 1.8.11, and I've noticed what seems to be a bit of a disparity between the documentation and the code for the BackgroundDetect application. |
| 09:51.31 | Malard_Office | i'm just using my mobile phone |
| 09:51.57 | Malard_Office | the number is +441733233709 |
| 09:52.06 | WIMPy | Malard_Office: Some mobile can do it. |
| 09:52.06 | Malard_Office | when i call it, it just gets disconnected |
| 09:52.32 | Malard_Office | well this windows phone mobile just says disconnected |
| 09:52.43 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:52.46 | Sophira | It turns out, looking at the code, that BackgroundDetect sets a channel variable "TALK_DETECTED" if it detects non-silence during the audio playback. With this, I can have it check if something was said even if I don't have a 'talk' extension, and can do it after the playback of the file. How come this variable isn't listed on the documentation? |
| 09:53.08 | Sophira | (Note: I haven't actually tested it yet, but it seems to me like that's the way it works) |
| 09:53.13 | Chainsaw | So anyone on Polycom SoundPoint IP phones, stay away from the "UC software" and stick with SIP. UCS stops accepting transmissions from Asterisk within 10 minutes. |
| 09:53.45 | Sophira | (the variable contains the length of the detected audio in ms, I believe) |
| 09:54.29 | WIMPy | Malard_Office: "Destination out of order". That usually means there is an issue between the CO and the NT. |
| 09:54.47 | Malard_Office | the what and the what? |
| 09:55.23 | WIMPy | The Central Office and the Network Terminator (the box you got). |
| 09:55.39 | WIMPy | Does the NT have LEDs? |
| 09:56.17 | Malard_Office | it has one |
| 09:56.26 | Malard_Office | if i disconnect the cat5 cable from it |
| 09:56.27 | Malard_Office | it pulses |
| 09:56.33 | Malard_Office | if i plug it in, it stays solid |
| 09:57.06 | WIMPy | interesting. |
| 09:57.37 | WIMPy | I've seen pulsing LEDs when the NT was connected to an analog line. |
| 09:57.46 | Malard_Office | it looks like this http://www.liquidlan.com/draytek/isdn2e_box1.jpg |
| 09:58.08 | Malard_Office | the light in the middle (you can see it as clear) will flicker/pulse when nothing is connected to one of the rj45 sockets |
| 09:58.15 | Malard_Office | there is 2 dip switches underneath also |
| 09:58.18 | Malard_Office | that's all BT left me |
| 09:58.30 | WIMPy | Only two? |
| 09:58.38 | Malard_Office | let me take a photo |
| 09:59.00 | WIMPy | It's not a model I've seen, but there are so many of them... |
| 09:59.58 | Chainsaw | WIMPy: It's a Euro-ISDN BRI "NT1" device. |
| 10:00.11 | WIMPy | Err. I should have read the full output. |
| 10:00.25 | WIMPy | It says the called equipment is non-ISDN. |
| 10:00.43 | WIMPy | That again smells like you don't have ISDN on that line. |
| 10:02.36 | Chainsaw | Malard_Office: I missed this trace that you gave WIMPy. Could I see it please? |
| 10:02.58 | Malard_Office | i did'nt give a trace |
| 10:02.59 | Malard_Office | basically |
| 10:03.04 | Malard_Office | there is 2 numbers i have |
| 10:03.19 | Malard_Office | +441733 233 709 |
| 10:03.26 | Chainsaw | You're in Peterborough? |
| 10:03.26 | Malard_Office | and a ddi range +44 1733 404 940 - 949 |
| 10:03.28 | Malard_Office | yes |
| 10:03.34 | Chainsaw | I can troubleshoot in person if you need it. |
| 10:03.45 | WIMPy | did a trace |
| 10:03.50 | Chainsaw | Where are you? I'm at Priestgate :) |
| 10:04.01 | Malard_Office | sending you pm |
| 10:04.01 | Chainsaw | (Pretty much next to the phone exchange, although I don't work for BT) |
| 10:04.03 | Chainsaw | Thanks. |
| 10:24.44 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 10:24.49 | cusco | hi |
| 10:26.16 | cusco | wct4xxp 0000:02:01.0: Interrupts not detected. |
| 10:26.34 | cusco | what does that mean? |
| 10:27.52 | kaldemar | cusco: what version of DAHDI are you using? |
| 10:28.06 | cusco | 216 261 |
| 10:28.10 | cusco | sorry |
| 10:28.12 | cusco | 2.6.1 |
| 10:28.14 | cusco | both |
| 10:28.30 | cusco | sometimes when machine reboots it starts |
| 10:28.44 | cusco | but now is just tellingme red alarm |
| 10:30.29 | cusco | dmesg: http://paste.debian.net/174080/ |
| 10:30.39 | cusco | should I use some older version kaldemar ? |
| 10:32.13 | kaldemar | you could try a 2.5 version: http://lists.digium.com/pipermail/asterisk-users/2012-March/271304.html |
| 10:33.02 | kaldemar | if you can reproduce it, maybe you've found a new issue and you should file a report. |
| 10:33.31 | cusco | how todebug? |
| 10:34.05 | cusco | We don't have redundancy right now, so I just need toget it up. |
| 10:34.21 | cusco | in a few months we will have some redundancy.. but until then... |
| 10:36.33 | cusco | compiling 2.5.1 |
| 10:38.02 | cusco | 2.5.1 actually is showing OK |
| 10:38.57 | cusco | pri show spans |
| 10:39.03 | cusco | is showing down,active |
| 10:46.49 | cusco | why? |
| 10:46.50 | cusco | :( |
| 10:49.17 | cusco | kaldemar: what am I missing? |
| 10:49.19 | cusco | noalarm |
| 10:49.28 | cusco | but pri show spans shows DOWN,ACTIVE |
| 10:55.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:56.07 | Dovid | hi all |
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| 12:56.05 | eXcAliBuR | I have asteriskGUI however, the changes I make in there don't show in my sip.conf or extensions.conf, i'm not sure if I missed a step to include other conf files... where does the gui put it's changes? |
| 12:57.37 | WIMPy | Try in #asterisk-gui |
| 12:58.10 | eXcAliBuR | not many people in thre |
| 12:58.15 | eXcAliBuR | i'd rather wait here |
| 12:58.19 | eXcAliBuR | if you don't mind. |
| 12:59.45 | eXcAliBuR | i found the conf file :) |
| 12:59.52 | eXcAliBuR | now i should be ok |
| 13:01.08 | eXcAliBuR | how do i use #include properly? and where do I place it, above or below [General] ? |
| 13:02.04 | WIMPy | Wherever you want it included. But you still better ask in #asterisk-gui what you can modify at all. |
| 13:03.07 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nniriiqyhbwqirqe) |
| 13:03.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:04.53 | *** join/#asterisk bytemaster (~ewre@host81-150-217-167.in-addr.btopenworld.com) |
| 13:05.47 | [TK]D-Fender | eXcAliBuR, anywhere |
| 13:06.27 | eXcAliBuR | do you know what other files i might have to include for asterisk-gui's voicemail, and extensions? |
| 13:06.31 | eXcAliBuR | i found the users.conf one |
| 13:07.51 | WIMPy | users.conf is an asterisk config file and need/must not be included. |
| 13:08.20 | eXcAliBuR | i included it and now my sip accounts can register |
| 13:08.31 | eXcAliBuR | :/ |
| 13:08.45 | [TK]D-Fender | you don't "#include" those files ..... |
| 13:08.48 | [TK]D-Fender | you need to HAVE them |
| 13:09.04 | [TK]D-Fender | #include is an extensions.conf directive for more dialplan. |
| 13:09.15 | eXcAliBuR | ummm |
| 13:09.26 | eXcAliBuR | i put #include users.conf in my sip.conf |
| 13:09.30 | [TK]D-Fender | NO |
| 13:09.38 | eXcAliBuR | and it let my phones register |
| 13:09.44 | eXcAliBuR | it seems to work |
| 13:09.49 | [TK]D-Fender | do not do that |
| 13:09.57 | pithagorians | anybody knows how can i permit more than one network per user in sip configuration ? |
| 13:09.58 | eXcAliBuR | why not? |
| 13:10.06 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:10.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:10.06 | [TK]D-Fender | You're mixing apples and oranges |
| 13:10.24 | [TK]D-Fender | Yuo don't put diesel gas in a regular gasoline engine |
| 13:10.38 | [TK]D-Fender | the configs are NOT the same and #include is meant to merge configs. |
| 13:10.42 | [TK]D-Fender | they are not alike |
| 13:10.47 | eXcAliBuR | ok so i'll take it out, then my phones won't register |
| 13:10.52 | rjvvliet | pithagorians: You mean add IP ACL's |
| 13:11.00 | [TK]D-Fender | you've messed other things up then like modules.conf. |
| 13:11.06 | pithagorians | yes |
| 13:11.33 | pithagorians | permit = 10.1.2.0/24;192.168.88.0/24 |
| 13:11.48 | rjvvliet | pithagorians: Use multiple permit= lines |
| 13:11.55 | pithagorians | i can't |
| 13:11.58 | eXcAliBuR | haha, guess i just needed to reload the confs |
| 13:12.01 | pithagorians | i have it in database |
| 13:12.01 | rjvvliet | nope, not onle line. |
| 13:12.06 | eXcAliBuR | thats why it seems to work |
| 13:12.15 | pithagorians | and the permit field is one |
| 13:12.19 | eXcAliBuR | my bad |
| 13:12.22 | eXcAliBuR | :P |
| 13:12.32 | rjvvliet | pithagorians: put two permit= in the file either with an permit IP |
| 13:12.58 | leifmadsen | semi-colon means comment |
| 13:12.59 | rjvvliet | pithagorians: A, oke SIP realtim. |
| 13:13.06 | pithagorians | yes :( |
| 13:13.24 | leifmadsen | oh you're using realtime |
| 13:13.39 | pithagorians | i've tried , and ; and & |
| 13:13.42 | leifmadsen | I'm not sure what the separator is for realtime in that case |
| 13:13.44 | pithagorians | nothing works |
| 13:13.48 | leifmadsen | it just might not be supported |
| 13:14.03 | pithagorians | that's bad |
| 13:14.16 | pithagorians | realtime has lot of constraints |
| 13:14.17 | leifmadsen | if it's true, I suppose |
| 13:14.34 | pithagorians | any idea for workarrounds ? |
| 13:14.40 | pithagorians | like use setvar |
| 13:14.42 | pithagorians | for ex |
| 13:14.59 | eXcAliBuR | loves his new digium phone |
| 13:17.03 | rjvvliet | pithagorians: have you tryed a Space ? |
| 13:17.09 | pithagorians | yes |
| 13:18.17 | [TK]D-Fender | pithagorians, tried "|"? |
| 13:19.01 | pithagorians | no |
| 13:19.03 | pithagorians | let me |
| 13:19.06 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 13:19.53 | leifmadsen | pithagorians: semicolon should be the separator for permit/deny, allow/disallow, and setvar according to my quick google search |
| 13:20.06 | pithagorians | [2012-06-12 15:20:08] ERROR[28228]: chan_sip.c:27418 build_peer: Bad ACL entry in configuration line 0 : 10.1.2.0/24|192.168.88.0/24 |
| 13:20.18 | rjvvliet | pithagorians: found that the ; sould be the separator on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 13:20.18 | [TK]D-Fender | Guess not |
| 13:20.41 | leifmadsen | error is showing a pipe |
| 13:21.56 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.202) |
| 13:22.13 | ziz212 | Friends , I am having a elastrix box and i add custom destination in freepbx. I couldnt find it on extension_custom.conf file? Is it differ in elasteix? |
| 13:22.45 | pithagorians | when i use - 10.1.2.0/24;192.168.88.0/24 - i get Device does not match ACL |
| 13:23.39 | rjvvliet | pithagorians: weird, also this page http://forums.asterisk.org/viewtopic.php?p=172328 refers to the semicolon. |
| 13:24.32 | *** join/#asterisk stix (~stix@193.89.191.209) |
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| 13:57.07 | pithagorians | the problem is in columns order |
| 13:57.43 | pithagorians | my schema was - first permit than deny |
| 13:57.56 | pithagorians | but it must be viceversa |
| 13:58.19 | pithagorians | because in realtime mode asterisk reads in table field by field |
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| 13:59.13 | rjvvliet | pithagorians: Glad you fixed it, thanks for sharing. |
| 13:59.57 | pithagorians | np |
| 14:00.11 | volga629 | Is somebody was using fanvil android phone ? |
| 14:01.14 | volga629 | hardware, just looking to info flush the firmware |
| 14:01.24 | pithagorians | <volga629> is there such software ? |
| 14:01.37 | pithagorians | i can't find it in android market |
| 14:01.48 | pithagorians | btw fanvil is crapy |
| 14:02.03 | volga629 | http://www.fanvilphones.com/products-d800.php |
| 14:02.24 | pithagorians | ip/config.php - will show all the config |
| 14:02.28 | pithagorians | including password |
| 14:02.35 | pithagorians | in plain text |
| 14:02.44 | volga629 | yes, just looking might replace with android which support srtp or l2tp |
| 14:02.58 | pithagorians | ah, you mean android on fanvil |
| 14:03.04 | volga629 | yes |
| 14:03.13 | pithagorians | i thought you ask about software phone from fanvil |
| 14:03.18 | pithagorians | no, never used |
| 14:03.41 | volga629 | I have very old firmware version and they posted new one which support SRTP |
| 14:03.48 | pithagorians | does it have the same "feature" ? |
| 14:04.03 | pithagorians | with /config.php |
| 14:04.47 | volga629 | no I can't log in the shell or phone only web interface which don't have match futures |
| 14:05.03 | pithagorians | it is by web |
| 14:05.20 | pithagorians | like 192.168.1.10/config.php |
| 14:05.27 | *** join/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 14:05.38 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
| 14:05.39 | pithagorians | instead is ip of your phone |
| 14:05.58 | *** part/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 14:06.03 | volga629 | no didn't tried this trick need check it |
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| 14:07.05 | volga629 | very interesting If I can put debug mode as android and flush firmware |
| 14:13.34 | stix | Hi guys. I am trying to communicate with a SIP-server (swyx). I get "Found audio description format PCMU for ID 0 Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) |
| 14:13.34 | stix | Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) SIP/2.0 488 Not acceptable here. I guess I should enable/install a PCMU-codec or something? |
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| 14:15.56 | WIMPy | yes. Why did you disable it in the first place? |
| 14:16.25 | stix | I didn't and I don't see it in "core show codecs" |
| 14:16.56 | WIMPy | ulaw |
| 14:17.23 | WIMPy | Xlaw=pcmX=G.711X |
| 14:17.25 | stix | is it ulaw? |
| 14:17.31 | stix | oh okay |
| 14:17.42 | stix | I only allow alaw on that trunk |
| 14:17.49 | stix | but that clears it up :) |
| 14:17.56 | eXcAliBuR | how to dial from console? I forget |
| 14:18.12 | eXcAliBuR | console dial maybe |
| 14:18.14 | eXcAliBuR | let me go try |
| 14:18.19 | WIMPy | channel originate |
| 14:18.23 | eXcAliBuR | yaps |
| 14:18.24 | eXcAliBuR | thats it |
| 14:18.25 | eXcAliBuR | thanks |
| 14:18.26 | eXcAliBuR | :] |
| 14:19.14 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:25.19 | eXcAliBuR | anyone know how to setup the speed dial keys on a digium phone? |
| 14:25.56 | WIMPy | would think the manual does. |
| 14:26.33 | eXcAliBuR | it didn't come with a manual |
| 14:26.33 | eXcAliBuR | :[ |
| 14:26.52 | WIMPy | Err? |
| 14:27.01 | WIMPy | Not even an online one? |
| 14:27.10 | eXcAliBuR | i'm searching for it |
| 14:27.13 | eXcAliBuR | but not finding it |
| 14:45.23 | *** join/#asterisk shadebob (~shadebob@41.248.242.85) |
| 14:48.12 | stix | On the Invite sent by my Asterisk, I see "Contact: <sip:Unknown@192.168.11.45:5060>" - where does that "Unknown" come from? |
| 14:49.04 | wdoekes2 | CALLERID(num) |
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| 14:49.18 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:53.35 | stix | wdoekes2: is that on the extension or trunk? |
| 14:53.56 | [TK]D-Fender | stix, its from your caller |
| 14:54.15 | stix | hmm |
| 14:55.42 | stix | I don't get why INVITE doesn't work when I use an asterisk-trunk to our swyx-sip-server. It works with a SIP-client called Telephone (for mac). I sniffed the invite, that works: From: "test" <sip:scaa@192.168.11.63>;tag=1YqttAe89kLBInF9pqMXp2ExjqBb6NQF |
| 14:56.29 | stix | …but from my asterisk it looks like: From: "scaa" <sip:10@192.168.11.63>;tag=as0450c149 |
| 14:56.33 | wdoekes2 | sip.conf: fromuser = scaa |
| 14:56.41 | stix | it's the only difference I see |
| 14:57.52 | stix | wdoekes2: is that on the extension or trunk? |
| 14:59.24 | stix | wdoekes2: wow, it worked setting it on the trunk :) |
| 14:59.41 | stix | thanks! |
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| 17:45.47 | eXcAliBuR | http://asterisk.pastebin.ca/2160590 anyone wanna look at this and explain what is happening |
| 17:45.51 | eXcAliBuR | please and thank you lots |
| 17:46.01 | eXcAliBuR | voice mail won't let ppl leave a message |
| 17:49.19 | rjvvliet | eXcAliBuR: i think the log level sould be increesed, i past it from the messages or full log, looks like i'am missing some info. |
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| 17:55.13 | oyugik | Hey guys can anyone help me on how to configure an analog phone on an fxs port for incoming calls |
| 17:55.57 | Qwell | ~book |
| 17:55.57 | 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://www.asteriskdocs.org/ or see ~buybook |
| 17:56.32 | eXcAliBuR | oyugik i called digium and they helped me |
| 17:56.42 | eXcAliBuR | you get free support when you registe |
| 17:56.47 | eXcAliBuR | registering is free |
| 17:56.49 | eXcAliBuR | :} |
| 17:58.35 | [TK]D-Fender | It would help to know what hardware is providing that FXS interface.... |
| 18:00.01 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 18:02.24 | oyugik | it is a tdm400p |
| 18:02.32 | oyugik | 2fxo and 1fxs |
| 18:03.21 | [TK]D-Fender | What have you configured for it so far? Do you see it in the DAHDI channel listings? |
| 18:03.26 | [TK]D-Fender | ~pb |
| 18:03.26 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:03.27 | [TK]D-Fender | ^^^ |
| 18:03.49 | oyugik | the card is working well |
| 18:04.20 | oyugik | i can confirm, I can make outbound pstn calls from some sip phones |
| 18:04.25 | [TK]D-Fender | For the FXS you just point that channel it to its context. That is what they will be able to dial |
| 18:05.06 | oyugik | can I pastebin my extensions.conf so you have a look? |
| 18:05.44 | [TK]D-Fender | show us that and your chan_dahdi.conf |
| 18:06.35 | oyugik | okay ... in a minute |
| 18:11.11 | oyugik | here it is |
| 18:11.14 | oyugik | http://pastebin.com/AnVaRm1K |
| 18:14.50 | oyugik | chan_dahdi.conf has default stuff I only added #include dahdi-channels.conf |
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| 18:17.42 | oyugik | @[TK]D-Fender have you had a look at it |
| 18:20.05 | [TK]D-Fender | oyugik, You have not showed us your dahdi configs as I requested |
| 18:20.20 | [TK]D-Fender | There is also likely an entry in users.conf for your FXS port... |
| 18:22.35 | oyugik | my chan_dahdi is mostly made up of long comment |
| 18:23.44 | rjvvliet | oyugik: i think TK means also the includes of that file. like dahdi-channels.conf |
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| 18:29.15 | oyugik | dahdi-channels http://pastebin.com/FVdrzP3j |
| 18:40.16 | [TK]D-Fender | oyugik, Right now your FXS is pointed to from-internal which does not exist. |
| 18:40.21 | [TK]D-Fender | Go point it somewhere else |
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| 18:42.04 | oyugik | where is that |
| 18:42.36 | oyugik | sorry for that stupid question but I just dont understand how it worls |
| 18:43.12 | [TK]D-Fender | You have your 3rd channel pointing to from-internal. I gave yuo the test you could have SEARCHED on since that's the only thing you should have to change... |
| 18:43.22 | [TK]D-Fender | as to the context, go look where you told your other phones to look |
| 18:45.05 | oyugik | so all I have to change is "from-internal" to say another context for example "from-pstn"? |
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| 18:49.31 | oyugik | I got it |
| 18:49.34 | oyugik | thanks |
| 18:52.19 | *** part/#asterisk oyugik (~oyugik@41.84.143.52) |
| 19:17.34 | *** join/#asterisk kenshin (~kenshin@dsl-173-248-248-45.acanac.net) |
| 19:19.47 | kenshin | hi is there a open source sol for g729 ? |
| 19:20.07 | Qwell | kenshin: There are no free and legal implementations of G.729. |
| 19:20.17 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 19:20.37 | WIMPy | ... in the US. |
| 19:20.58 | kenshin | (somehow if I disable g729 on my at a nothing works, and if I keep ip enabled some calls --mostly outgoing loose audio in one way--) |
| 19:21.05 | kenshin | I'm in canada :) |
| 19:21.06 | Qwell | Hire an attorney that specializes in patents and copyright. |
| 19:21.18 | Qwell | Otherwise, pay the whopping $10/channel. |
| 19:23.38 | kenshin | well the thing I'd prefer not to use it (what;s the benefit anyway? --) |
| 19:24.07 | Qwell | get a SIP provider that supports something other than G.729 then |
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| 19:25.01 | Russ | or wait for it to expire |
| 19:25.02 | kenshin | they both do |
| 19:25.16 | kenshin | (ulaw ) |
| 19:25.35 | Qwell | Are your phones using it? |
| 19:26.49 | kenshin | my ata : http://pastebin.com/yD3HMP7u |
| 19:27.21 | kenshin | (is there a way to see what codecs in what order the other peer supports? |
| 19:28.40 | kenshin | sip show peer gives this : Codecs : (gsm|ulaw|speex|ilbc) < I think this is my list not theirs> |
| 19:31.32 | malcolmd | what's important is what's in their invite; call from the device to asterisk, and the device will invite w/ codecs it supports (or it should) |
| 19:33.16 | kenshin | how can get a glimpse of their list ??? |
| 19:33.40 | kenshin | (when a call is placed ??) |
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| 19:52.18 | kenshin | if I put a list of codecs (in some order ) in the general section of sip.conf that would be valid for all peers (except where there is a list in the def of a peer)? |
| 19:54.10 | kenshin | (like the scope of a variable in C) |
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| 20:00.54 | [TK]D-Fender | kenshin, yes |
| 20:01.10 | kenshin | cool, than |
| 20:01.11 | kenshin | x |
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| 21:42.54 | RubyRails | is it possible to count the number of rings an outbound call makes before connecting/disconnecting? |
| 21:50.50 | kenshin | RubyRails, what do you mean ? |
| 21:51.05 | RubyRails | if we make call using Dial() |
| 21:51.19 | RubyRails | is it possible to get the number of rings it took before the caller answered/hung up? |
| 21:51.42 | WIMPy | billedseconds-dialedseconds |
| 21:52.54 | RubyRails | okay, only can get seconds, not number of rings? |
| 21:55.00 | WIMPy | Divide the seconds by the ringin frequency of your country. |
| 21:56.19 | RubyRails | ah, good idea |
| 21:56.53 | RubyRails | thanks :) |
| 21:58.23 | kenshin | in what context does asterisk transcode ??? |
| 21:58.38 | kenshin | I mean what can trigger it (and how)? |
| 21:59.11 | WIMPy | Err? Bridging two channels with different codecs? |
| 22:01.02 | *** join/#asterisk Kostas (~Kostas@unaffiliated/kostas) |
| 22:01.39 | Kostas | Hello ;) I have a problem with asterisk 1.8 and gui 2.0 |
| 22:01.52 | Kostas | i get the message i dont have permisions |
| 22:01.58 | Kostas | what can i do for it ? |
| 22:01.59 | Kostas | thanks |
| 22:05.10 | Kostas | anyone ? |
| 22:06.39 | WIMPy | Which GUI? |
| 22:07.13 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nniriiqyhbwqirqe) |
| 22:07.24 | Kostas | digium i think ? im new to this |
| 22:07.42 | Kostas | but if you have to suggest anythink else im here to hear you |
| 22:07.46 | WIMPy | Then try #asterisk-gui or the documentation. |
| 22:07.53 | kenshin | how would I force that ( inside network, g711 anything that goes outside something else --gsm for ex --) |
| 22:08.19 | Kostas | WIMPy do you suggest anythink else ? |
| 22:08.25 | WIMPy | I don;t think anyone here would recommend any of the GUIs, but freePBX seems most alive. |
| 22:08.26 | Kostas | anything* |
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| 22:30.31 | citywok | anybody good at reading a PRI debug? Can you tell me if this is an error on my end, or on my carriers? (I've been waiting for 2 hours for Qwest/CenturyLink to figure out their stuff) http://pastebin.com/aCs1ficj |
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| 22:41.10 | *** mode/#asterisk [+o file] by ChanServ |
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| 23:13.56 | *** join/#asterisk fneuwald (~fneuwald@201.86.128.17.static.gvt.net.br) |
| 23:15.05 | fneuwald | I'm receiving a value with "new line" and space, in set, like this ( -- Executing [dgv_sms@from-pstn:4] Set("DGV/3-SMS-1", "mensagem=message |
| 23:15.05 | fneuwald | ") in new stack - how do I remove new line and excess spaces in dial plan? |
| 23:27.17 | fneuwald | solved using the function FILTER |
| 23:27.19 | fneuwald | tks |
| 23:27.21 | *** part/#asterisk fneuwald (~fneuwald@201.86.128.17.static.gvt.net.br) |
| 23:28.20 | sarsaeol | is this dialplan all that is need to ensure voicemail will be used correctly (unavail and busy when appropriate) |
| 23:28.38 | sarsaeol | http://pastebin.com/wVzgtFaY |
| 23:29.14 | sarsaeol | seems to work okay but just wanted some confirmation |
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| 23:39.58 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 23:40.01 | cj | hey folks |
| 23:40.19 | cj | are any of you familiar with setting up STUN servers? |
| 23:40.44 | sarsaeol | resists urge to make pun with 'stun' |
| 23:41.38 | cj | whatever was done, it was done into silence. do rhymes count? |
| 23:41.58 | cj | AND HE STUNNED THE ENTIRE #asterisk CHANNEL INTO SILENCE |
| 23:42.06 | sarsaeol | lol nice |
| 23:42.12 | cj | takes a bow |
| 23:42.16 | sarsaeol | id help but i know jack abt STUN srry |
| 23:42.18 | cj | debian? stun? anybody? |
| 23:42.24 | cj | np. |
| 23:42.56 | cj | I should read the RFC and fix the debian package's default config or its dpkg-reconfigure scripts or something |
| 23:43.26 | cj | but then I would have competition from anybody who knew how to run a debian host |
| 23:43.52 | cj | wait. CFR what? Can you give me a link? |
| 23:44.15 | cj | http://tools.ietf.org/html/rfc5389 |
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| 23:47.41 | cj | is that anything like https://www.google.com/search?aq=f&sourceid=chrome&ie=UTF-8&q=cfr+47+part+97 ? |
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