| 00:10.06 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-135-045.pools.arcor-ip.net) |
| 00:15.15 | Mezevenf | WIMPy, What should chan_dahdi contain in this scenario? |
| 00:15.19 | Mezevenf | by default its empty. |
| 00:15.45 | WIMPy | There's an example in the cards manual. |
| 00:16.08 | Mezevenf | Entered those 6 lines in to suit, so nothing else really required? |
| 00:16.54 | Mezevenf | using bri_net for singalling |
| 00:16.58 | Mezevenf | signalling* |
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| 00:17.50 | WIMPy | That should be enough, yes. |
| 00:28.52 | Mezevenf | Any ideas why the D-Channel is going up and down constantly when trying to callout? |
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| 00:43.18 | WIMPy | I'm not sure ATM how chan_dahdi handes this. It might just be an inactivity timeout. |
| 00:43.30 | WIMPy | Does it cause any issues? |
| 00:45.49 | raden_work | I never thought VOIP over a 1 Mile wifi hop could be better than land line internet :) |
| 00:47.37 | Mezevenf | WIMPy we dont seem to be able to get a dial tone/make calls. As soon as the PBX attempts to dial out the line just goes up and down constantly until its hung up |
| 00:47.51 | Mezevenf | The D-Channel message that is. |
| 00:49.25 | WIMPy | Do yu have the correct switchtype? |
| 00:52.20 | Mezevenf | using euroISDN (I'm in Australia) |
| 00:52.40 | WIMPy | ok. Any messages? |
| 00:53.02 | Mezevenf | Just the D-Channel going up and down on span 4 when trying to place a call |
| 00:53.30 | WIMPy | ok, time for 'pri set debug 2 span 4' then. |
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| 00:57.16 | Mezevenf | pri meaning primary channel or should I use bri for that? |
| 00:58.19 | tonsofpcs | isn't an ISDN usually a BRI (or multiples, bonded)? |
| 00:58.57 | Mezevenf | yea |
| 00:58.59 | Mezevenf | it is BRI |
| 00:59.12 | Mezevenf | just wondering if that pri in that command was related or not, I dont think it is |
| 00:59.14 | tonsofpcs | (BRI = Basic Rate Interface. PRI = Primary Rate Interface) |
| 00:59.21 | tonsofpcs | oh, i have no clue, I don't actually use asterisk |
| 00:59.43 | WIMPy | There is pnly a pri command. can_dahdi uses libpri. |
| 00:59.50 | Mezevenf | ok cool, thought so |
| 00:59.54 | WIMPy | bri support came later. |
| 00:59.58 | Mezevenf | lots of information, what should I be looking for? |
| 01:00.38 | WIMPy | Any real messages? Or just layer 1 stuff? |
| 01:00.52 | WIMPy | Or put it up on a pastebin. |
| 01:01.23 | WIMPy | tonsofpcs: "usual" depands heavily on where you are. |
| 01:01.57 | Mezevenf | http://pastebin.com/A2btgSp6 |
| 01:03.15 | tonsofpcs | WIMPy: While I understand that Ts and Es are technically ISDNs, most places that offer them seem to use "ISDN" to refer to BRI-based stuff and "Tx" or "Ex" or "frame-relay" or similar to refer to PRI-based stuff |
| 01:04.15 | WIMPy | Mezevenf: I only see messages going in one direction. That looks like a very low level issue. Do you still get gree light? |
| 01:04.18 | WIMPy | green |
| 01:04.39 | Mezevenf | yup green lights |
| 01:05.16 | WIMPy | tonsofpcs: Tx and Ex can be anything, not just ISDN, and especially in the case of Tx I think it's rather unlikely to be ISDN. And frame relay is something different. |
| 01:05.16 | Mezevenf | I'm attempting to have the PBX make calls out via the SIP trunk, the ISDN card is only really there to interface with the old PBX. |
| 01:06.57 | WIMPy | wonders why it does to TEI assigned. That's fishy. |
| 01:07.23 | WIMPy | 1. because there only are messages going one way and 2. bebause it's ptp. |
| 01:07.30 | tonsofpcs | WIMPy: honestly, at this point, "ISDNs" don't have to be on an ISDN. |
| 01:07.46 | Mezevenf | whats TEI? |
| 01:08.04 | WIMPy | tonsofpcs: What does that mean? |
| 01:08.18 | WIMPy | Mezevenf: A Terminal Endpoint Identifier. |
| 01:09.39 | tonsofpcs | WIMPy: I can have a telecom company demarc an ISDN subscriber line that does not integrate directly to the 'telephone' network but instead hops a few different places and ends up getting bridged in as a different signaling scheme entirely. |
| 01:10.39 | WIMPy | tonsofpcs: Sure. That's becoming the norm, but in theory the user shouldn't (have to) care what the physical setup looks like. |
| 01:11.10 | tonsofpcs | WIMPy: in practice, stuff like that makes 'ISDN' connections _very_ unhappy. |
| 01:11.25 | WIMPy | Mezevenf: What versions of Asterisk and libpri are you using? |
| 01:11.43 | tonsofpcs | luckily, I don't have to deal with that, we have copper coming in from the RBOC |
| 01:12.30 | WIMPy | has ISDN via SIP, which as you can imagine is nearer to doesn't work that to works. |
| 01:12.52 | tonsofpcs | wonders wwhy one would "ISDN via SIP".... |
| 01:13.02 | WIMPy | Because it's cheap. |
| 01:13.15 | Mezevenf | 1.6.2.13 for Asterisk |
| 01:13.28 | tonsofpcs | but what does it give you over an IP connection (for data) or a SIP connection (for voice)? |
| 01:13.34 | WIMPy | No local switches any more. Just DSLAMs and a softswitch somewhere. |
| 01:13.55 | tonsofpcs | so the telco provides you that? |
| 01:14.05 | WIMPy | Yes. |
| 01:14.26 | tonsofpcs | wow. For once, I'm glad that I'm in an area that doesn't have much technology refresh |
| 01:14.44 | tonsofpcs | (it's ok, we have tons of glass for other fun things :) |
| 01:14.56 | Mezevenf | 1.4.11.4-0 |
| 01:14.56 | WIMPy | Yes, that NGN stuff is really horrible. |
| 01:15.01 | Mezevenf | for libpri |
| 01:15.18 | tonsofpcs | although it is kinda annoying that it would cost me almost 2x as much for SIP trunking than it would for PRI(s) for voice..... I'd rather not go through g.711 10 times per phone call... |
| 01:16.13 | WIMPy | Mezevenf: I'd try more recent versions of both. libpri 1.4.11 wasn't that ideal. With NT-pmtp it was pretty bad, actually. |
| 01:17.13 | WIMPy | tonsofpcs: Yes, that's the fun part here as well, PRIs are a lot cheaper than the same bandwidth with IP. |
| 01:18.05 | tonsofpcs | really, the only reason that it will cost me 2x for SIP trunking is that the service provder thinks they should be charging me for an IP connection over their glass (already in the building, for us only, mind you) and the data (SIP) traveling over that glass but only when the data is carried over IP.... |
| 01:18.24 | tonsofpcs | it's not like the IP connection they provide the SIP over is useful for anything else.... |
| 01:18.50 | WIMPy | Pricing can be very interesting. |
| 01:19.34 | tonsofpcs | we currently have two T1s over glass being demarced to POTS by a battery bank that they own and maintain in a box bolted to the wall in our building.... |
| 01:21.07 | WIMPy | VOIP termination for more than home use is quite expensive here, as well. |
| 01:21.57 | WIMPy | As long as you're happe with what sipgate provides, you're fine, but if you want more, it gets expensive. |
| 01:22.01 | tonsofpcs | like, if I wanted to do SIP over an IP connection that I'm paying for seperately from the SIP connection, I'd just pay for 'lines' from a SIP provider on the internet and trunk over the net. |
| 01:22.32 | tonsofpcs | if I'm going to a phone company for SIP 'lines', they shouldn't bill me for the data twice. |
| 01:23.10 | WIMPy | No. |
| 01:23.14 | Mezevenf | So for my current setup, should I be utilizing trunks or custom extensions to interface with the ISDN-SIP? |
| 01:23.17 | WIMPy | Luckily I haven't seen that. |
| 01:23.40 | tonsofpcs | because really, with SIP, you're paying for the data and handoff. |
| 01:23.46 | WIMPy | Mezevenf: What are you talking about? |
| 01:24.06 | tonsofpcs | WIMPy: it gets better, but not anything that i'd disclose to an open channel.... |
| 01:24.24 | WIMPy | Getting youtr numbers hosted is the expensive part. |
| 01:24.26 | Mezevenf | I'm not really sure how the old PBX should be routing to the SIP trunk |
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| 01:25.03 | WIMPy | Mezevenf: If you're only using one SIP account, you only need one Dial(). |
| 01:25.19 | WIMPy | Anyway, it's standard dialplan stuff. |
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| 01:32.32 | trelane | is anyone using their PBX with a managed SIP E911 service? |
| 01:32.55 | WIMPy | Managed? |
| 01:33.07 | trelane | IE there's another end that's taking the call and handling E911 |
| 01:33.38 | WIMPy | How else could that work? |
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| 02:17.36 | ghostmediapro1 | http://pastebin.com/i1SjSZL9 |
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| 02:27.54 | elb | Hi all ... I'm trying to get calls to/from Google Talk/Google Voice working, and it's only sort of going. Incoming calls come into asterisk fine, but the other end seems to get the Answer() ... the calling party rings until it goes to Google Voicemail. Outgoing calls ring for a few seconds, then report "Everyone is busy/congested at this time". I can send XMPP messages using JabberSend(). |
| 02:28.13 | elb | never seemes to get the Answer(), that is |
| 02:28.20 | elb | any ideas what I might have flubbed up? |
| 02:28.58 | elb | asterisk 1.8.3.2 |
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| 02:52.08 | Mezevenf | My ISDN PBX is connected to my Asterix box, however when you pickup a handset and dial 0 to make a call, the system answers it like an incoming call. Whats the best way to route this 'incoming' call to the sip trunk to enable it to dial out? |
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| 05:52.37 | p3nguin | Holy crap, Asterisk 1.8 uses 4x the RAM as 1.4. |
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| 06:05.50 | coppice | p3nguin: the Korean economy needs your support |
| 06:06.11 | p3nguin | :( |
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| 08:02.29 | benngard | [Mar 19 09:05:53] ERROR[22932]: res_fax.c:1728 receivefax_exec: error initializing channel 'OOH323/Avaya2-1622' in T.38 mode <- that doesnt look good :( |
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| 08:07.22 | coppice | is there even code present to negotiate T.38 on am H.323 call? |
| 08:08.02 | benngard | yes it is |
| 08:08.26 | benngard | if i send the call to an spa2103 with a fax it works |
| 08:08.57 | benngard | may has applied alot of patches for t38 over ooh323 |
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| 08:15.15 | drcode | hi all |
| 08:15.16 | coppice | I've never tried T.28 with H.323. I would guess its a mess, as T.38 over SIP was a mess 5 years ago, and most of the shaking out since then has been focussed on SIP |
| 08:15.34 | drcode | I am trying to config simple iax.conf |
| 08:15.36 | drcode | I got this: |
| 08:15.43 | drcode | chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.1.103 in the calltokenoptional list or setting user anatoliy requirecalltoken=no |
| 08:15.44 | drcode | any idea? |
| 08:17.56 | benngard | coppice: t.38 and ooh323 works well, we have about 20 faxes at our company that receives like 100 calls per day using t.38 and ooh323 |
| 08:18.24 | kerframil | drcode: http://downloads.asterisk.org/pub/security/IAX2-security.pdf |
| 08:19.29 | drcode | thanx |
| 08:23.05 | coppice | benngard: you need to be in the 100k FAXes a day range before you start to really shake these things out :-) |
| 08:26.19 | benngard | coppice: i am happy wthat my 100 faxes per day is working :) |
| 08:27.41 | drcode | i am trying to setup guest call |
| 08:27.48 | drcode | with user it work |
| 08:28.04 | drcode | any idea? |
| 08:31.13 | kerframil | drcode: you haven't read the document |
| 08:31.35 | kerframil | drcode: specifically, section 2.1.3.3. which concerns guest access |
| 08:35.32 | drcode | I will check again |
| 08:36.07 | drcode | wht I use in iaxclient? |
| 08:36.14 | drcode | user guest |
| 08:38.31 | drcode | kerframil: sorry , but it dosn't work |
| 08:38.36 | drcode | I did this : |
| 08:39.04 | drcode | [guest] |
| 08:39.06 | drcode | type=user |
| 08:39.08 | drcode | context=test |
| 08:39.09 | drcode | callerid="Guest IAX User" |
| 08:39.10 | drcode | requirecalltoken=no |
| 08:39.13 | kerframil | drcode: I don't know what the exact problem you're having now is, nor the nature of your test case. anyway, I'm going to sleep now. maybe someone else can help. also, here's a link which explains exactly how asterisk behaves during iax authentication: http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication |
| 08:39.36 | drcode | ok |
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| 08:48.57 | benngard | yay, may did fix my ooh323 -> receivefax error, it works now :) |
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| 09:35.30 | benngard | yay, fax -> pstn - avaya -> ooh323 (with t38) -> asterisk -> ReceiveFAX (covert to pdf and mail to me) woks :) |
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| 09:47.11 | coppice | covert pdfs sound really secure |
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| 12:13.03 | roxdragon | hi all |
| 12:13.52 | roxdragon | where is the download sip voip setting for nokia n95? |
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| 12:17.31 | kaldemar | roxdragon: on asterisk side, nothing special is needed. |
| 12:19.31 | roxdragon | kaldemar, don't work the voice |
| 12:19.37 | roxdragon | on my nokia |
| 12:25.54 | kaldemar | roxdragon: set directmedia=no for the device |
| 12:26.34 | roxdragon | on asterisk? |
| 12:27.55 | roxdragon | http://paste.ubuntu.com/582446/ kaldemar |
| 12:28.00 | roxdragon | trun for nokia |
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| 12:37.55 | roxdragon | kaldemar, Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729) |
| 12:42.27 | roxdragon | kaldemar, solved but don't work MOH |
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| 13:17.28 | benngard | hmmm, when u send fax to an extension that have "faxdetect=yes" shouldnt that call jump to fax,1,,... |
| 13:19.33 | neurosys | Does * 1.4 support 802.1p? |
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| 13:25.57 | iprouteth0 | benngard: There might be some other T.38 setting you need to set |
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| 14:07.46 | JerJer | iprouteth0: faxdetect=yes has nothing to do with T.38 |
| 14:08.18 | JerJer | faxdetect turns on a DSP listener for the 'fax tone' |
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| 14:49.22 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 14:49.40 | Aut0Exec | hi guys... quick question... the first place asterisk looks for configs when it loads is asterisk.conf yes? |
| 14:51.53 | Aut0Exec | I have asterick installed in /opt and its not finding any of the config files... even tho I changed the location in "directories" inside of asterisk.conf |
| 14:52.08 | Aut0Exec | is there another place I need to change this as well? |
| 14:54.59 | *** join/#asterisk BeeBuu (777f3755@gateway/web/freenode/ip.119.127.55.85) |
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| 14:56.24 | BeeBuu | anyone help me please? |
| 14:57.13 | BeeBuu | how can i set the NAT with SIP realtime? |
| 14:59.01 | elb | BeeBuu: what problem are you having? |
| 14:59.08 | *** join/#asterisk fnordus (~dnall@S01060023693bfad4.va.shawcable.net) |
| 14:59.32 | elb | you probably want to set rtpstart and rtpend in rtp.conf [general] to establish an expected range for RTP ports |
| 14:59.49 | elb | then forward that range of ports, plus 5060, to your asterisk server |
| 15:00.18 | elb | set externip to your true external IP, and/or configure STUN |
| 15:00.20 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 15:00.20 | *** mode/#asterisk [+o file] by ChanServ |
| 15:00.25 | elb | (externip in sip.conf, that is) |
| 15:02.13 | BeeBuu | elb: i want to know which conf file need to set the external IP? |
| 15:02.22 | BeeBuu | btw: i'm using realtime SIP |
| 15:03.33 | elb | I just told you |
| 15:03.35 | elb | sip.conf |
| 15:03.59 | elb | if you use stun, in res_stun_monitor.conf |
| 15:04.05 | elb | but I have this in my sip.conf: |
| 15:04.13 | BeeBuu | i had tried,but does not work |
| 15:04.14 | elb | [general] |
| 15:04.21 | elb | externip=<myhostname> |
| 15:04.53 | elb | "does not work" leaves a lot of room for error, what about it doesn't work? |
| 15:04.56 | BeeBuu | elb: are you using realtime SIP? |
| 15:05.05 | elb | have you looked at the SIP data, is it stuffing with the wrong IP? |
| 15:05.24 | elb | I haven't, but that shouldn't change wht's in the packets |
| 15:05.47 | elb | realtime SIP is account management, not IP management :-P |
| 15:06.10 | BeeBuu | when i register the sip user, "chan_sip.c:23861 build_peer: Invalid peer port configuration at line 0 : 0" show in CLI> |
| 15:06.22 | elb | so what makes you think this is an externip/NAT problem? |
| 15:06.26 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 15:06.42 | Aut0Exec | can someone please help with configs for asterisk |
| 15:06.52 | Aut0Exec | I installed in opt and now I need to make asterisk find them |
| 15:07.01 | Aut0Exec | even when I use a -C param it still complains |
| 15:07.44 | BeeBuu | what's up? |
| 15:08.13 | Aut0Exec | BeeBuu, hey man |
| 15:08.18 | BeeBuu | elb: i had done with rtp.conf,and forward the 5060 to asterisk server,and add the externip in sip.conf,but still does not work |
| 15:08.23 | Aut0Exec | BeeBuu, i installed asterisk on openwrt.. in a /opt dir |
| 15:08.33 | BeeBuu | hi,Aut0Exec |
| 15:13.37 | Aut0Exec | BeeBuu, how can I make asterisk find my config file? :( |
| 15:14.11 | WIMPy | Aut0Exec: Did you tell it where you want them? |
| 15:15.33 | Aut0Exec | WIMPy, yeah i specified the -C param |
| 15:15.56 | Aut0Exec | WIMPy, when I check ps -aux... I see it load the correct command with the -C but still doesnt find it |
| 15:16.41 | WIMPy | I never trie anything other that what I configure-ed. |
| 15:17.47 | WIMPy | Have you looked with strace? |
| 15:18.27 | Aut0Exec | nah didnt go that far |
| 15:18.33 | Aut0Exec | i installed on openwrt |
| 15:18.34 | Aut0Exec | my router |
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| 15:43.59 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 15:48.08 | raden | out of the blue for no reason I have one way audio and cannot call out when someone calls in i can hear them but they cannot hear me |
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| 15:52.16 | wdoekes2 | no reason? or did something actually change between your asterisk and the other end? |
| 15:53.12 | raden | Just the service provider this week |
| 15:53.19 | raden | from DSL to Cable |
| 15:53.25 | raden | everything else stayed the same |
| 15:53.28 | raden | IPs changed |
| 15:54.10 | raden | brb |
| 15:54.19 | elb | haha |
| 15:54.22 | raden | ? |
| 15:54.29 | elb | "out of the blue for no reason" ... "I got a new service provider" |
| 15:54.30 | elb | nice |
| 15:54.39 | raden | yea but everything was working |
| 15:54.54 | raden | and we have many other phones working just fine on same setup |
| 15:55.08 | raden | this morning my phone won t dial out |
| 15:55.22 | elb | so there are multiple phones on this asterisk server, all behind the same Internet connection, and they all work but one? |
| 15:56.05 | raden | everyone phone behind the internet connection works |
| 15:56.18 | raden | the ones external are all fine but mine |
| 15:56.27 | raden | going to restart my POS router in a min |
| 15:56.33 | raden | Need to get a cisco here |
| 15:57.14 | elb | ok, so the asterisk server is remote to the phone in question |
| 15:57.23 | elb | only one phone is affected, and it is the only phone on its Internet connection? |
| 15:57.30 | elb | is the connection NATted and/or firewalled? |
| 15:57.48 | raden | nated |
| 15:57.53 | raden | no firewall |
| 15:57.57 | raden | ports forwarded |
| 15:58.00 | raden | been working for years |
| 16:00.08 | raden | we have a WIFI bridge tooo <modem><wifi bridge><router><asterisk> QOS was enabled on the radios could that cause a issue ? |
| 16:00.09 | wdoekes2 | .. until you switched provider |
| 16:00.20 | raden | wdoekes2, actually it worked after the switch |
| 16:00.52 | raden | switch was TUE i placed my last call yesterday morning that i knew it was working |
| 16:01.45 | wdoekes2 | and your outbound calls from your phone to the asterisk don't work at all? |
| 16:01.59 | raden | they go through cant hear anything |
| 16:02.01 | wdoekes2 | and for inbound calls you miss the outbound audio? |
| 16:02.09 | raden | and inbound i can hear the person but they cant hear me |
| 16:02.25 | wdoekes2 | ok.. outbound calls work, but no audio |
| 16:02.40 | raden | correct |
| 16:02.52 | drmessano | externhost/externip changed? |
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| 16:03.31 | raden | drmessano, shit we did restart asterisk yesterday due to a weird network issue that downed are internal network |
| 16:03.35 | raden | drmessano, lemee check |
| 16:04.06 | drmessano | If you're natted, I would think you have externhost or externip defined in sip.conf |
| 16:04.14 | drmessano | You need to check there |
| 16:05.12 | raden | drmessano, I have a feeling that was the issue restarting phone now |
| 16:05.36 | raden | I bet they shut off our other internet last night to complete the change over |
| 16:05.44 | raden | would make sense |
| 16:06.25 | raden | drmessano, thank you sir :) |
| 16:06.29 | raden | all working again |
| 16:06.43 | raden | i just checked I cant ping our old IPS |
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| 16:14.53 | root52 | Hey All, I am having a reorder tone and SIP 503 sent back from my carrier when trying to dial certain area codes. Most calls complete just fine. But this 613 seems to give them trouble. I already logged a support ticket with them but I wanted to see if there was something I was missing on my end. I have the SIP trace of the failed call at http://pastebin.com/srYxFGCz Am I missing something on my end or is this their problem? Thanks!! |
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| 16:31.06 | roxdragon | hi all |
| 16:32.25 | roxdragon | i've a problem with asterisk... when registry my nokia n95 from external LAN, i don't call all internal |
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| 16:37.17 | root52 | roxdragon: More info. Like pasetbin of Asterisk CLI on failed call... |
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| 16:40.02 | roxdragon | root52, no error for call |
| 16:40.26 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 16:40.50 | roxdragon | output nokia: services telephone available |
| 16:41.29 | roxdragon | in the LAN works all |
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| 16:48.04 | root52 | roxgragon: NAT problem? Really need more information about what Asterisk is doing when you try to call and "internal" number. |
| 16:48.41 | raden | anyone figurew out the aastra music on hold issue in 1.8.x |
| 16:48.47 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
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| 16:50.10 | roxdragon | root52, NAT from 10000 to 20000 UDP 5060, 5061.... use DMZ |
| 16:50.17 | roxdragon | asterisk 1.8 root52 |
| 16:50.45 | roxdragon | Asterisk does not do anything |
| 16:50.55 | roxdragon | nokia is registred |
| 16:53.43 | root52 | brb, need food |
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| 17:32.44 | Shaaan | !ipphone |
| 17:33.31 | *** join/#asterisk drcode (~chatzilla@bzq-84-111-89-77.red.bezeqint.net) |
| 17:33.44 | drcode | hi al |
| 17:34.14 | drcode | I understand that IAX have user and password then I build dialplan to each user |
| 17:35.22 | drcode | I want to let users register with there own chosen numbder and they will be able to call each other , any idea? |
| 17:35.48 | WIMPy | What is the question? |
| 17:37.14 | drcode | is it possible? |
| 17:37.31 | drcode | can I let user register there own number and call each other in iax? |
| 17:37.32 | WIMPy | yes |
| 17:37.56 | WIMPy | Is you don't set callerid in your config, the one sent from the user is used. |
| 17:39.16 | WIMPy | But to call a number you have to configure a destination. |
| 17:39.26 | drcode | can U please show me an example , iax.conf and ext.conf? |
| 17:39.49 | drcode | lets say user will put name and password as number |
| 17:39.53 | drcode | 12345/12345 |
| 17:40.03 | drcode | in iax client |
| 17:40.37 | drcode | other user will put 12346/12346, then user 12345 will be able to call 12346 |
| 17:40.56 | WIMPy | How are you going to autorize users? |
| 17:41.13 | drcode | they will register there self |
| 17:41.23 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 17:41.24 | WIMPy | Anyone? |
| 17:41.25 | drcode | I use it in local network |
| 17:41.27 | drcode | yes |
| 17:42.30 | WIMPy | You can only register configured peers. |
| 17:43.53 | drcode | I see |
| 17:44.14 | drcode | so I will have to create each users my own? |
| 17:44.37 | drcode | I can connect same users twice ? |
| 17:44.43 | WIMPy | yes |
| 17:44.47 | drcode | and use id for number? |
| 17:44.48 | WIMPy | no |
| 17:45.12 | drcode | so its not possible |
| 17:45.28 | drcode | I will have to create each user and build dialplan? |
| 17:45.40 | WIMPy | You can configure an ID for the peer. If you don't the peer chooses itself. |
| 17:46.12 | WIMPy | I don't see a way round that. |
| 17:46.32 | drcode | ok |
| 17:47.05 | drcode | what it mean guest? |
| 17:47.43 | WIMPy | Acept unauthenticated calls. |
| 17:49.16 | drcode | so it is possible to use guest ? |
| 17:49.25 | WIMPy | It might be possible to register users via an unauthenticated call and save their location in a databse. But that will require some custon scripting. |
| 17:49.36 | WIMPy | Sure |
| 17:49.45 | WIMPy | But you can't call guests. |
| 17:49.54 | drcode | I see |
| 17:50.03 | drcode | how do I setup guest? |
| 17:50.19 | WIMPy | Just enable it. |
| 17:50.35 | drcode | ok |
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| 19:11.17 | roxdragon | i've a problem with asterisk... when registry my nokia n95 from external LAN, i don't call all internal |
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| 19:36.00 | roxdragon | toc toc |
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| 19:37.58 | gpled | can anyone recommend a wifi sip phone? |
| 19:38.24 | roxdragon | help.. nokia not work on externel LAN.. debug is: http://paste.ubuntu.com/582606/ |
| 19:38.28 | roxdragon | help please :'( |
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| 19:46.02 | wdoekes2 | roxdragon: you're trying to call a domain |
| 19:47.14 | wdoekes2 | R-URI should be sip:403@79.43.6.215, not sip:79.43.6.215 |
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| 21:14.16 | guifort | Hello all |
| 21:15.15 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 21:15.21 | guifort | I have a little question , I have a FXO Card X100P "Clone" with it i have some echo (in my side) and some time the card is not reconized "Unknow DAA chipset" |
| 21:16.01 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 21:16.10 | guifort | I have plan to replace it with a Cisco SPA3102 , what do you think about this product ? |
| 21:22.06 | Kobaz | hmm |
| 21:22.10 | Kobaz | p3nguin: poke |
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| 21:45.29 | Kobaz | is 3 way conferencing available in skinny? |
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| 21:58.07 | tzafrir_laptop | guifort, have you tried using OSLEC as the echo canceller? |
| 21:58.27 | guifort | humm, no |
| 21:59.15 | guifort | I must recompile dahdi drivers for that no ? |
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| 23:11.43 | Freeaqingme | I'm trying to set up asterisk realtime. In my mysql logs I see asterisk connecting to mysql, but sip show peers and sip show users give zero results (nor does asterisk perform a SELECT query to retrieve users/peers) |
| 23:17.46 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 23:19.39 | raden | is there a way if a call is missed in asterisk like no one picks it up and the person does not leave a voicemail it can notify someone somehow |
| 23:28.40 | *** join/#asterisk faisndfasda (~asdcz@95-42-105-83.btc-net.bg) |
| 23:28.42 | faisndfasda | hello |
| 23:29.03 | faisndfasda | anyone ever tried callerid spoofing on gsm networks ? |
| 23:30.35 | KNERD | I hear a possible spammer |
| 23:32.26 | faisndfasda | ? |
| 23:47.14 | elb | I'm trying to get calls to/from Google Talk/Voice working with Asterisk 1.8.3.2, and it's only sort of going. Incoming calls come in fine, but emit 'Unable to handle indication 3', and the other end keeps ringing even when the local phone picks up. Outgoing calls ring for a few seconds, then report 'Everyone is busy/congested at this time'. I can send XMPP messages using JabberSend(). Any ideas what I've flubbed? |
| 23:47.58 | elb | raden: I have xmpp notifications set up for that, actually |
| 23:49.12 | elb | raden: for any DIALSTATUS other than ANSWER, I send "Missed call from ${CALLERID(name)} at ${CALLERID(num)}" to an XMPP ID registered to the extension |
| 23:49.43 | elb | you could scope to specific circumstances as you need |
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