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| 01:02.44 | lucasb | Hey all... I have a question and need some info or advice on where to begin looking ..... I am running Asterisk 1.4.24.1 and I have set a custom, non-standard SIP port in sip.conf and when I run Asterisk it doesn't show that it's listening on that port when I do a netstat -ant |
| 01:03.12 | lucasb | Also, any incoming call (specifying the non-standard SIP port) doesn't connect, obviously because that port isn't open... |
| 01:04.00 | lucasb | Asterisk seems to be partially working since I can make outgoing calls just fine; though that doesn't rely on the incoming SIP port :) |
| 01:06.28 | lucasb | Is anyone here? |
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| 02:34.27 | PhunTelTek | need help getting a remote extension to register. All other extensions work properly. |
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| 02:41.11 | PhunTelTek | Someone must have had the same problem... |
| 02:43.06 | PhunTelTek | I added nat=yes and externip=77.274.23.187 and localnet=10.1.0.0/255.255.255.0 |
| 02:43.20 | PhunTelTek | to the sip_nat.conf file |
| 02:51.38 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 02:51.53 | PhunTelTek | I have UDP ports 5060:5082 10000:20000 forwarded to my * |
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| 02:58.14 | PhunTelTek | yawns |
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| 03:01.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 03:11.04 | darkmadda | anyone using google voice with their asterisk server? |
| 03:11.17 | joat | yes |
| 03:12.05 | darkmadda | is there a way to get incoming calls to my voice account to forward into an extension on my server? |
| 03:12.15 | joat | sure... |
| 03:12.23 | joat | you're piping it through gizmo? |
| 03:13.17 | *** join/#asterisk toughmarketing (n=toughmar@ip72-199-181-246.sd.sd.cox.net) |
| 03:13.50 | darkmadda | i tried to setup gizmo to forward to my sip .... but calls into the voice account don't make it to me. If i call my gizmo account (via sip) from my asterisk server the calls forward to back to my asterisk extension. |
| 03:14.17 | darkmadda | so it's like it half works. |
| 03:15.21 | darkmadda | if i turn off forwarding the calls come into my gizmo soft client just fine |
| 03:15.53 | joat | gizmo presents the call recipient with an ivr that you have to fake out... add "Wait(3)" and "SendDTMF(1)" to your extension |
| 03:16.26 | toughmarketing | I was wondering what you guys do in the case of redundancy. I would like to setup say 2 or 3 servers all running asterisk and if say server1 dies server2 and server3 still work smoothly. I want them to all have the same dial plans and settings. Also maybe even balance the load between the servers. Is this possible? I am running 1.6 with no gui. I am just looking for ideas of what others are doing. I was even thinking of se |
| 03:16.39 | joat | if you set the extension to autoanswer in a conference room, and dial into the conf room with a separate phone, you'll hear the ivr |
| 03:17.45 | joat | the ivr goes something like "press one to accept the call, press two to send the call to voicemail", etc. |
| 03:19.01 | kmem | I thought the IVR "press one to accept"... ie call presentation was done through GV/GC, not from gizmo |
| 03:19.46 | joat | kmem, it may be |
| 03:20.05 | joat | never tried to separate the two |
| 03:20.27 | joat | got that part wrong then |
| 03:20.31 | joat | still works tho |
| 03:20.33 | joat | :) |
| 03:20.46 | kmem | I've got a GV and GC account, and you can turn this off on GV but there is no option to disable it in grandcentral |
| 03:21.08 | Alfio | toughmarketing you can use asterisk real time Asterisk RealTime Architecture. ARA |
| 03:21.22 | toughmarketing | Alfio: cool going to read up on it now :) thank you |
| 03:21.25 | kmem | I wish GV/GC would allow you to change the number of rings when it answers |
| 03:22.05 | joat | kmem, did they ever fix the outbound SIP issues in GV? |
| 03:22.30 | kmem | u mean outbound via sip from GV? |
| 03:22.46 | joat | dialing to PSTN via GV |
| 03:22.52 | joat | from SIP |
| 03:22.59 | kmem | I dont think it works anymore, I couldnt make it work |
| 03:23.13 | darkmadda | can't you do it via Gizmo5? |
| 03:23.28 | kmem | no outbound, unless you pay |
| 03:23.42 | joat | it's still usuable via the click-to-call interface, i wonder if some create web scraping would solve it |
| 03:23.55 | darkmadda | yeah but calls to gv should be free (you just need to put $1.00 in your dial out) |
| 03:24.01 | kmem | there are a few programs that allow you to call via GV/GC without going to the webpage |
| 03:24.05 | joat | s/create/creative/ |
| 03:24.15 | darkmadda | joat. you link is bad. |
| 03:24.32 | darkmadda | s/you/your/ |
| 03:24.43 | kmem | right now im using GC > GV > GIZMO > * |
| 03:25.03 | kmem | just cause I still want to be able to answer my GC # |
| 03:25.40 | darkmadda | i'm looking for PTSN > GV > SIP or PTSN > GV > GIZMO > sip |
| 03:25.41 | kmem | its a pain though because the GC VM comes on too quick (before GV can pick up) |
| 03:27.08 | kmem | dark i dont think that works anymore because (from what I read) GV uses a new proxy to talk to gizmo |
| 03:27.42 | darkmadda | but shouldn't gizmo be able to forward to my sip ? |
| 03:29.17 | toughmarketing | Alfio: Am I going in the right direction with this? Say three servers all pulling from the same database? |
| 03:29.50 | kmem | i dont think u can forward giz to GV sip anymore, but I'm no expert |
| 03:30.15 | Alfio | yes |
| 03:30.28 | Alfio | sharing the same dial plan |
| 03:30.35 | toughmarketing | Alfio: easy enough :) |
| 03:30.38 | toughmarketing | Alfio: thanks buddy |
| 03:30.57 | kmem | anyone in here using a dlink VTA |
| 03:31.05 | darkmadda | no i'm trying to forward gv to giz.... which works.... and then from giz to my asterisk box |
| 03:31.39 | PhunTelTek | need help getting a remote extension to register. All other extensions work properly. |
| 03:31.41 | Alfio | no prob |
| 03:31.51 | joat | darkmada: new link sent |
| 03:32.08 | darkmadda | joat: you rocks |
| 03:32.14 | darkmadda | s/rocks/rock/ |
| 03:32.28 | darkmadda | (i love regex) |
| 03:32.53 | kmem | I don't see why that wouldnt work. |
| 03:33.18 | Alfio | PhunTelTek |
| 03:33.22 | Alfio | ~ nat |
| 03:33.23 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 03:33.46 | Alfio | ~ sipnat |
| 03:33.47 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:35.32 | PhunTelTek | assuming all that was done, what may be another reason for a remote phone to not register? |
| 03:36.40 | joat | PhunTelTek: stun? |
| 03:37.35 | PhunTelTek | not using a stun server. |
| 03:38.14 | joat | might need to set it in the phone if it's behind NAT |
| 03:39.35 | PhunTelTek | it's behind the workplace firewall. |
| 03:40.06 | joat | odd, just plugged in an old grandstream phone. it booted and promptly said that i had 280 messages, though none in voicemail. <sarcasm>I just love GS phones!</sarcasm> |
| 03:46.30 | [TK]D-Fender | You don't need STUN, and if you want to know why it isn't registering, look at * SIP DEBUG |
| 03:47.51 | PhunTelTek | the extension is never mentioned in the sip debug. it's almost like it's being ignored. |
| 03:49.16 | [TK]D-Fender | PhunTelTek: You sure you actually enabled it? |
| 03:50.09 | PhunTelTek | wireshark on the work side shows the registrations going out. It shows SYN packets going out and SYN, RST packets being received from the *. |
| 03:50.56 | [TK]D-Fender | PhunTelTek: those are TCP, not UDP |
| 03:51.06 | [TK]D-Fender | PhunTelTek: * SIP is UDP and therefor stateless |
| 03:51.09 | PhunTelTek | enabled debug yes. SIP SET DEBUG |
| 03:51.56 | [TK]D-Fender | PhunTelTek: Then packets aren't reaching * at all. Could be the phone or any piece of networking in the way |
| 03:53.43 | PhunTelTek | I have the ports forwarded on my firewall , so it must be the work firewall. |
| 03:54.50 | PhunTelTek | but the SYN RST packets appear to come from my IP for *. |
| 03:56.20 | [TK]D-Fender | PhunTelTek: Well if the wrok firewall is blocking the phone then its DOA |
| 03:58.22 | PhunTelTek | they shouldn't be blocking outgoing. |
| 03:59.35 | [TK]D-Fender | PhunTelTek: Indeed. Doesn't mean they aren't. Then again who knows if the phone is configured right. Or whats on your * side |
| 04:01.22 | PhunTelTek | It's an x-lite softphone, configured similar to those i'm using at home. On the * side I have a smoothwall with all the ports i know of forwarded to *. |
| 04:02.08 | [TK]D-Fender | PhunTelTek: What are what precisely? |
| 04:02.26 | PhunTelTek | ports? |
| 04:02.43 | [TK]D-Fender | PhunTelTek: Yes, what exactly have you forwarded on the * side? |
| 04:03.08 | PhunTelTek | 5060-5082, 3478, 8000-8001, 10000-20000 all UDP |
| 04:07.35 | PhunTelTek | i just did a sheilds up scan of those ports, they all show stealth. but they are forwarded, so * is blocking them. |
| 04:08.33 | [TK]D-Fender | PhunTelTek: No, the scan that shows "steath" is a ***TCP*** test |
| 04:08.46 | [TK]D-Fender | PhunTelTek: UDP is STATELESS. It can't be "steath" |
| 04:09.14 | PhunTelTek | oh yeah, hmphh |
| 04:09.27 | [TK]D-Fender | PhunTelTek: So its all client-dise |
| 04:09.30 | [TK]D-Fender | side* |
| 04:12.55 | PhunTelTek | is there an test extension on the web somewhere? |
| 04:14.20 | [TK]D-Fender | PhunTelTek: To test what? |
| 04:14.37 | PhunTelTek | my * setup |
| 04:14.47 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:15.15 | [TK]D-Fender | PhunTelTek: set up a dummy SIP account and PM me the info and I'll test it for you |
| 04:19.52 | toughmarketing | I have compiled odbc into my asterisk 1.6 and setup odbc.ini odbcinst.ini cdr_odbc.conf and res_odbc.conf and I am getting this on an incoming call: [Jul 18 21:17:56] ERROR[29933]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. |
| 04:23.29 | [TK]D-Fender | PhunTelTek: So its your remote end. Phone, netowrking, or both |
| 04:24.22 | PhunTelTek | yeah the work firewall must be killing it. we have cisco 7940 phones, but they use skinny protocol. |
| 04:24.59 | [TK]D-Fender | PhunTelTek: Skinny + NAT = PAIN |
| 04:25.48 | PhunTelTek | I'm trying to use the x-lite phone. I think I'll have to ssh tunnel some ports through. |
| 04:25.56 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 04:27.51 | PhunTelTek | thanx for the help. it has been driving me nuts trying to figure out which end the problem was on. |
| 04:31.12 | toughmarketing | I fixed it... |
| 04:35.19 | PhunTelTek | would a stun server help me get through a corporate firewall? |
| 04:35.31 | [TK]D-Fender | PhunTelTek: No. |
| 04:36.09 | PhunTelTek | Is there a way to tunnel though port 80 HTTP? |
| 04:36.16 | [TK]D-Fender | PhunTelTek: STUN only helps the phone side have a better idea how to keep a return path open. Your problem is the first packets aren't even getting out. |
| 04:36.45 | [TK]D-Fender | PhunTelTek: You could setup an SSH tunnel against that... |
| 04:37.37 | PhunTelTek | will do, thanx a bunch |
| 04:44.43 | toughmarketing | How do I customize the cdr using odbc and mysql? |
| 04:45.50 | darkmadda | ok so i've been chatting with kmem (trying to get gizmo incoming trunk setup) things look good i get the calls incoming but voice doesn |
| 04:46.05 | darkmadda | 't come in... it goes out but not in |
| 04:46.14 | darkmadda | any one familiar with this problem |
| 04:47.01 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 04:47.30 | [TK]D-Fender | darkmadda: Typically NAT issues : |
| 04:47.30 | kmem | the joys of sip/nat :P |
| 04:47.32 | [TK]D-Fender | ~sipnat |
| 04:47.33 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:47.34 | [TK]D-Fender | ^^^ read |
| 04:49.20 | kmem | thanks fender |
| 04:54.32 | toughmarketing | I am using an agi script to do a call from my asterisk machine and it is not storing in the cdr after switching to odbc. Anyone else have this issue? |
| 04:57.05 | darkmadda | i thought it might be that but i've even put the * server on the dmz no help |
| 04:58.29 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 05:00.51 | [TK]D-Fender | toughmarketing: What about other calls? |
| 05:01.11 | [TK]D-Fender | DarkDYou need to do other settings, read the GUIDE, and DMZ is overkill |
| 05:01.20 | [TK]D-Fender | darkmadda: You need to do other settings, read the GUIDE, and DMZ is overkill |
| 05:01.26 | toughmarketing | [TK]D-Fender: It is logging all incoming calls perfectly. Outgoing on the other hand are not being locked. |
| 05:01.42 | [TK]D-Fender | toughmarketing: Are they being answered? |
| 05:01.48 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 05:01.50 | toughmarketing | [TK]D-Fender: yep |
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| 05:05.18 | toughmarketing | [TK]D-Fender: I just noticed if I dial out with a soft phone it logs, but i guess its something with the phpagi script. What is wierd is it is making the call so I would only assume it should log it regardless of how it is started. |
| 05:05.55 | [TK]D-Fender | toughmarketing: Would have to see.. |
| 05:06.32 | *** join/#asterisk psykon (n=psilikon@140-1.35-65.tampabay.res.rr.com) |
| 05:06.33 | toughmarketing | [TK]D-Fender: which part, I can paste it on pastebin or something. I am completely lost on this. |
| 05:10.32 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 05:13.13 | kmem | ~sipnat |
| 05:13.14 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:13.18 | [TK]D-Fender | touchComplete call attempt at * CLI with AGI and core debug |
| 05:13.48 | PhunTelTek | If someone else has broadvoice, let me know how often * should register with them. It seems the default is too frequent. |
| 05:14.39 | [TK]D-Fender | PhunTelTek: Too frequent for whom? |
| 05:14.40 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 05:16.44 | kmem | fender, why would an ATA lose registration and not respond to http or ssh? Anything besides a hardware problem? |
| 05:18.08 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 05:21.08 | [TK]D-Fender | kmem: What ATA supports SSH? What do EITHER have to do with SIP? |
| 05:21.32 | kmem | i can ssh into my dlink |
| 05:22.05 | [TK]D-Fender | kmem: We know as much as you've shown for this : guess how much..... |
| 05:22.07 | kmem | it appears my ata just freezes up every 12 hours or so |
| 05:22.30 | PhunTelTek | too frequent for me trying to watch the logfiles . |
| 05:22.41 | [TK]D-Fender | kmem: And common concensus says "D-Link sucks... no not just sucks... BLOWS..... **AND** swallows |
| 05:22.54 | kmem | yep I will have to agree with you |
| 05:23.01 | kmem | but it's all ive got right now... |
| 05:23.26 | kmem | funny thing is before I used asterisk I didn't notice this problem, it may just be a coincidence |
| 05:24.58 | [TK]D-Fender | kmem: Well if your device lock up which is what you've said and completel lack of connectivity indicates, then its just junk and not much to be said for it |
| 05:25.19 | *** join/#asterisk jcape (n=jcape@adsl-99-132-248-246.dsl.chcgil.sbcglobal.net) |
| 05:25.38 | kmem | yeah I suppose I just answered my own question |
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| 05:32.15 | *** part/#asterisk toughmarketing (n=toughmar@ip72-199-181-246.sd.sd.cox.net) |
| 05:32.55 | joako | Anyone knows a softphone that can dial a SIP URI without having to register to anything, just install it and start to dial? |
| 05:33.14 | joako | My live is all VoIP so you can dial (e.g.) 5551212@joako.com and reach me, I want someone overseas to do that but with minimal hassle |
| 05:33.32 | andrewn | ekiga? |
| 05:36.09 | [TK]D-Fender | yup |
| 05:36.59 | joako | Hmm I tried to use it on my laptop normal just to register to my asterisk server. I happened to be at a datacenter and my cell phone battery was drained. It registered fine but when I dialed a call it would just ring on my end forever when asterisk indicated the call had been answered |
| 05:37.44 | [TK]D-Fender | joako: Well you've got no evidence to show us to help you with |
| 05:38.51 | joako | Yes, I know. But being on a public IP with no NAT and the same thing happening at my home where there is NAT but at least 5 or 6 SIP registrations to the same server never with any issues leades me to rule out things without looking into it... I just haven't touched Ekiga since then |
| 05:41.18 | andrewn | i haven't used ekiga much, but it worked perfectly when i used it for testing |
| 05:43.31 | joako | Well it's not working for me seems Asterisk is confusing the unregistered client from the same IP as my Nokia wifi phone.... But the facility to dial a SIP URI directly is there. I guess I will pass it on and go from there |
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| 07:03.23 | elitecoder | hey guys, I made an auto dialer and I have 7 lines from bandwidth.com. I'm getting about 50% congestion. It's really affecting performance. Does anyone know why this is happening? |
| 07:09.24 | elitecoder | I really need to sort this out. If anyone has a clue feel free to send me a PM. |
| 07:09.29 | elitecoder | I'll check my messages in the morning. |
| 07:10.31 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
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| 07:25.33 | *** join/#asterisk aurman (n=aurman@ip68-3-218-92.ph.ph.cox.net) |
| 07:25.47 | aurman | hi |
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| 07:36.34 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 07:42.24 | aurman | how do i make a condition based on callerid num? |
| 07:42.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 07:43.08 | aurman | i want to make a 'from' context so that if a call comes from a certain #, do Bridge(etc), otherwise Dial(${PHONE}) |
| 07:51.06 | morko757 | how do digiums cards with FOX ports detect the call been picked up? |
| 07:51.13 | morko757 | *FXO |
| 08:11.51 | morko757 | I mean, what kind of answer supersvision does the system have? |
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| 08:26.15 | psykon | i am looking for a guide on how to use a Linksys ATA with asterisk. |
| 08:32.47 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 08:35.13 | aurman | psykon, there are tons |
| 08:35.25 | aurman | did you google for "linksys ata with asterisk"? |
| 08:36.01 | aurman | http://lmgtfy.com/?q=linksys+ata+with+asterisk |
| 08:38.16 | aurman | http://voxilla.com/tools/device-configuration-wizard |
| 08:38.23 | aurman | that looks promising |
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| 08:42.04 | psykon | Followed a couple but they didn't seem to work |
| 08:42.12 | psykon | i'll try that one tho |
| 08:45.21 | psykon | I am getting the ATA registered to my asterisk box and the asterisk box registered to my sip provider but I am having no luck establishing an extension. |
| 08:49.02 | aurman | do you know that it's registered? |
| 08:49.10 | aurman | like when you do a 'sip show registry'? |
| 08:49.23 | aurman | and sip show peers |
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| 09:00.10 | psykon | yes registered to callcentric.com |
| 09:01.24 | psykon | 2 peers. One is my ATA and the other is the outside global addr of the sip provider |
| 09:01.57 | aurman | so what happens? |
| 09:03.06 | psykon | i try to dial and it tells me it Call from 'jshas' to extension '727xxxxxxx' rejected because extension not found. |
| 09:04.59 | aurman | that means you don't have anything that matches 727xxxxxxx in your dialplan |
| 09:06.14 | psykon | i have "exten => 727xxxxxxx,1,Dial(SIP/jshas) |
| 09:08.39 | aurman | it's not seeing it |
| 09:09.43 | aurman | shouldn't it be exten => _727XXXXXXX,1,Dial(SIP/jshas/${EXTEN}) |
| 09:10.47 | psykon | Atleast I know I am registered... don't know why I am using port 5080 to callcentric.com isntead of the 5060 I expected but I'll figure that out later |
| 09:11.10 | aurman | ^^^ |
| 09:11.30 | psykon | I think I am going to read the dial plan basics in The Future of Telephony\ |
| 09:12.36 | aurman | *shrug* |
| 09:15.59 | psykon | i appreciate the help! I gotta go to sleep |
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| 12:03.55 | stix | Which action from the AMI should I use to transfer a call? |
| 12:06.29 | kaldemar | stix: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect |
| 12:06.46 | stix | aha that's what they call it |
| 12:06.51 | stix | thanks :) |
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| 13:02.19 | Strale | <PROTECTED> |
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| 13:08.42 | Zhad | module unload chan_zap :-p |
| 13:09.12 | Zhad | (Disclaimer: The previous comment was not meant to be taken seriously). |
| 13:11.31 | Strale | Zhad: but I like everything to work normal after hangup |
| 13:12.04 | Dovid | soft hangup <tab> |
| 13:12.10 | Dovid | and hang up one at a time |
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| 13:26.17 | stix | Which action from the AMI should I use to put a call on hold? |
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| 14:02.07 | maikols | hi all |
| 14:02.59 | maikols | any can give me same answer on zaptel chan???? |
| 14:03.56 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 14:03.56 | hesco | What does this mean, in response to a dialplan reload: 'pbx.c:2055 pbx_find_extension: Maximum PBX stack exceeded', and what can I do about it? |
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| 14:15.01 | maikols | is it good to use bristuff and zaptel or i need use of misdn and dahdi????? |
| 14:15.05 | maikols | what is the best |
| 14:15.06 | maikols | ?? |
| 14:15.13 | maikols | zaptel channel seems not working good |
| 14:15.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:15.34 | maikols | zaphfc set the d channel down every hour and i must restart zaptel to get it up |
| 14:15.37 | maikols | any help? |
| 14:17.33 | maikols | any who have experience with hfc pci card? |
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| 14:55.13 | Orbixx | Are Asterisknow and Trixbox similar things? |
| 14:55.21 | Orbixx | Or are there significant differences between them? |
| 14:55.26 | Kobaz | i have a 'call forwarding' type of problem in 1.6.0.10. i have my desk phone ring for 20 seconds, if there is no pickup, the dialplan will dial my cell phone |
| 14:55.35 | Kobaz | when my cell phone picks up, no audio is passed |
| 14:55.44 | Kobaz | same exact dialplan worked fine in 1.4.x |
| 14:56.31 | aurman | Orbixx: From what I understand, AsteriskNow/AsteriskGUI was written from scratch, and is not another FreePBX copy |
| 14:57.03 | Kobaz | http://pastebin.com/m374da933 |
| 14:57.12 | [TK]D-Fender | Orbixx: Trixbox uses a forked FreePBX and custom changes to * to suppotr this meaning attempting to upgrade either outside of their support won't get you any here. Also Trixbox comes loaded with a ton of other crap |
| 14:58.15 | [TK]D-Fender | Kobaz: Then you shold be showing us the failed call with SIP debug, not your dialplan. |
| 14:58.34 | Orbixx | For somebody who just wants IVR, voicemail and an "out of office hours" thing... |
| 14:58.43 | Orbixx | What would be recommended? |
| 14:58.44 | [TK]D-Fender | Orbixx: They' |
| 14:58.49 | [TK]D-Fender | Orbixx: They'll both do it |
| 14:59.09 | Kobaz | http://pastebin.com/m60adf3e2 |
| 14:59.12 | Kobaz | [TK]D-Fender: yeah, working on it |
| 14:59.15 | [TK]D-Fender | Orbixx: I would avoid forked versions. |
| 14:59.22 | Kobaz | that's the verbose log... i don't see any problems with that part of it |
| 14:59.42 | aurman | [TK]D-Fender: How can I make it so calls coming into a context with a particular CID do one thing while the rest dial my extensions as normal? |
| 14:59.51 | Orbixx | [TK]D-Fender: So AsteriskNOW or FreePBX? |
| 15:00.09 | [TK]D-Fender | Orbixx: FreePBX is PART of AsteriskNOW and Trixbox. |
| 15:00.12 | *** join/#asterisk x86 (n=porteb1@p3m/member/x86) |
| 15:00.23 | [TK]D-Fender | Orbixx: However Trixbox uses a FORKED Asterisk and FreePBX. |
| 15:00.47 | [TK]D-Fender | kob SIP DEBUG |
| 15:00.51 | [TK]D-Fender | Kobaz: SIP DEBUG |
| 15:00.56 | Kobaz | yes yes |
| 15:00.57 | Orbixx | Right. |
| 15:00.58 | Kobaz | getting it |
| 15:01.07 | aurman | oh I thought AsteriskNow/AsteriskGui was started from scratch :) Shows what I remember |
| 15:01.18 | Orbixx | And AsteriskNOW just uses FreePBX and Asterisk. |
| 15:01.34 | aurman | it used to do some crazy XML config stuff |
| 15:01.45 | [TK]D-Fender | aurman: AsteriskNOW used to come with AsteriskGUI (by Digium) included. This changed not too long ago |
| 15:02.02 | aurman | Ah, cool. |
| 15:02.14 | [TK]D-Fender | Orbixx: Both use both. Trix uses FORKED versions however |
| 15:02.31 | Orbixx | Yeah. |
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| 15:03.08 | [TK]D-Fender | [11:01]<aurman>it used to do some crazy XML config stuff <- news to me... |
| 15:03.29 | [TK]D-Fender | aurman: Unless you aren't talking about CE here and their commercial version instead which I do not really know |
| 15:04.13 | Orbixx | Is the asterisknow website down? |
| 15:04.14 | aurman | When I first tried AsteriskGUI by itself back when it was in beta, the config files it created were XML it looked like |
| 15:04.51 | [TK]D-Fender | aurman: That sounds more like FreeSWITCH... |
| 15:05.09 | [TK]D-Fender | aurman: *GUI uses std configs (though primarily users.conf) |
| 15:05.43 | Kobaz | [TK]D-Fender: http://pastebin.com/m40193615 |
| 15:05.49 | Kobaz | [TK]D-Fender: there it be |
| 15:05.58 | aurman | Hm. I suppose I did try a whole bunch of GUIs so I could be remembering the wrong one :) |
| 15:06.50 | Kobaz | [TK]D-Fender: there's zero rtp traffic on the second dial (if the cell phone picks up) |
| 15:06.58 | Kobaz | [TK]D-Fender: i do get rtp if the first dial picks up |
| 15:07.10 | aurman | So checkit.. this script says to put "exten => s/6502650000,1,Bridge(${DB_DELETE(gv_dialout/channel)}, p)" in my extensions.conf |
| 15:07.14 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:115:8a39:2cf:8b07) |
| 15:07.20 | aurman | but I know that s/<number> part isn't going to pick anything up |
| 15:07.55 | aurman | when calls come in, they get handled by <username>,1,etc, <username>,2,etc, and so on |
| 15:08.06 | *** part/#asterisk ryguillian (n=ryguilli@addr-arpa.in) |
| 15:08.08 | aurman | how do I pick out just the call that comes in from a certain CID ? |
| 15:08.31 | Kobaz | [TK]D-Fender: thanks for taking a look |
| 15:08.48 | [TK]D-Fender | Kobaz: Contact: <sip:7177240000@192.168.5.1> <--- ANOTHER person who never seems to "get it". BAD NAT CONFIG. |
| 15:08.51 | [TK]D-Fender | ~sipnat |
| 15:08.51 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:08.53 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 15:09.21 | Kobaz | [TK]D-Fender: i don't have nat |
| 15:09.30 | Kobaz | the asterisk box is directly on the net |
| 15:10.03 | aurman | hehe why's it trying to use that internal IP? |
| 15:10.24 | Kobaz | i don't know |
| 15:10.32 | Kobaz | that's my eth1 ip |
| 15:10.33 | Kobaz | which is lan |
| 15:10.43 | Kobaz | eth0 has a 207... public ip |
| 15:10.45 | [TK]D-Fender | Kobaz: Look at that Contact. Sorry... I call BSkobMy bad. PB your sip.conf for your ITSP |
| 15:10.53 | [TK]D-Fender | aurman: nvm on that |
| 15:11.01 | [TK]D-Fender | akjshdlkjafglsd |
| 15:11.06 | [TK]D-Fender | gah |
| 15:12.16 | Kobaz | http://pastebin.com/m782ec3bf |
| 15:13.28 | Kobaz | should i define localnet |
| 15:14.21 | aurman | I do, just in case |
| 15:14.29 | aurman | and I have externhost= defined |
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| 15:14.42 | *** mode/#asterisk [+o angler] by ChanServ |
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| 15:14.50 | aurman | since my ip could change |
| 15:14.57 | aurman | (instead of externip=) |
| 15:15.02 | Kobaz | yeah |
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| 15:16.17 | [TK]D-Fender | Kobaz: Right now I don't see the problem..... still looking. Be sure to use "nat=no" for your peer though |
| 15:16.27 | [TK]D-Fender | aurman: externhost + externrefresh |
| 15:16.35 | aurman | yes |
| 15:16.43 | Kobaz | k |
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| 15:19.03 | Kobaz | okay i did nat=no |
| 15:19.07 | Kobaz | but in sip show peer... it says: Nat : RFC3581 |
| 15:19.17 | Kobaz | which is the same as what it said before |
| 15:21.04 | [TK]D-Fender | Kobaz PB everything. |
| 15:22.20 | aurman | D-Fender: did my question earlier make sense? |
| 15:23.02 | Kobaz | http://pastebin.com/m2832165a |
| 15:25.25 | *** join/#asterisk Caesar (n=apollock@debian/developer/apollock) |
| 15:25.40 | Kobaz | i added externhost and localnet |
| 15:25.42 | Kobaz | same problem |
| 15:25.55 | Caesar | Hi, I'm trying to understand my sip.conf a bit better... |
| 15:26.17 | Caesar | The register directive in the [general] section, is that for inbound or outbound calls? |
| 15:27.05 | Kobaz | registration is for telling a remote system where you are... so you can recieve calls |
| 15:27.24 | Caesar | Kobaz: cool, thanks |
| 15:28.12 | Caesar | So that means the other related section in the [authentication] section is for outbound calls... |
| 15:30.10 | *** join/#asterisk MrNaz (n=mrnaz@118.208.209.120) |
| 15:31.00 | Kobaz | for outbound generally you configure sip peers, each sip peer is a different 'account' or 'trunk' |
| 15:31.44 | Kobaz | ~sippeers |
| 15:32.07 | Kobaz | isn't there a bot entry for that |
| 15:32.14 | Kobaz | how do you search |
| 15:32.21 | Kobaz | ~sip |
| 15:32.22 | infobot | rumour has it, sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 15:32.32 | maikols | kobaz do you think is good to use zaptel and bristuff or misdn and dahdi ???? |
| 15:32.44 | Kobaz | dahdi is the new stuff |
| 15:32.59 | Kobaz | so if you have problems, and file bugs on zaptel... it won't help much |
| 15:33.03 | maikols | but i only need to add the misdn driver or i need same other patch? |
| 15:33.30 | maikols | the problem is that the dchannel go down every hour and don't go up |
| 15:33.43 | maikols | thisi is my problem |
| 15:33.49 | Kobaz | maikols: ask your telco to check their end also |
| 15:33.53 | maikols | i use the zaphfc modle |
| 15:34.13 | maikols | the telco is not the problem |
| 15:34.16 | Kobaz | though, i have had nasty problems with bad drivers on rhino cards |
| 15:34.21 | maikols | if i restart zaptel it come up agian |
| 15:34.26 | Kobaz | and the rhino would randomly start dropping the d channel |
| 15:34.27 | Kobaz | k |
| 15:34.46 | maikols | if i use the dahdi stuf |
| 15:34.54 | Kobaz | apparently it was a dsp problem... took me two months of time working with rhino support... and they still didn't fix it... so i switched to sangoma |
| 15:35.12 | maikols | in the outbound extention how i can call the channel |
| 15:35.24 | maikols | now i use Dial(ZAP/1/EXTEN) |
| 15:35.32 | maikols | with dahdi what i must use????? |
| 15:35.36 | Kobaz | [TK]D-Fender: no ideas? |
| 15:35.41 | Kobaz | yeah |
| 15:35.47 | Kobaz | well that's zap |
| 15:35.52 | Kobaz | for dahdi it's similar |
| 15:35.56 | *** join/#asterisk eliyahud (n=eliyahu@77.127.163.130) |
| 15:35.57 | Kobaz | Dial/DAHDI/... |
| 15:36.04 | maikols | ok |
| 15:36.06 | Kobaz | Dial(DAHDI/.. |
| 15:36.26 | maikols | so you are telling me that misdn modules are more stable than zaptel one? |
| 15:36.54 | [TK]D-Fender | [11:28]<Caesar>So that means the other related section in the [authentication] section is for outbound calls... <- no, and it is not required |
| 15:37.01 | [TK]D-Fender | Kobaz: I was expecting your sip.conf... |
| 15:37.04 | Kobaz | maikols: i've never used misdn |
| 15:37.06 | Kobaz | [TK]D-Fender: oh |
| 15:37.12 | Kobaz | [TK]D-Fender: i pasted it before. lemme get it |
| 15:37.53 | [TK]D-Fender | Kobaz: ALL OF IT |
| 15:37.58 | Kobaz | http://pastebin.com/m6b28a399 |
| 15:37.58 | Kobaz | oh |
| 15:38.01 | Kobaz | well |
| 15:38.04 | Kobaz | it's comming from a database |
| 15:38.14 | Kobaz | lemme do the whole query with the general section |
| 15:43.21 | *** join/#asterisk dispy (n=werbung_@p4FDF3243.dip0.t-ipconnect.de) |
| 15:43.42 | Kobaz | http://pastebin.com/m4af2c514 |
| 15:43.45 | Kobaz | thar it be |
| 15:44.51 | *** join/#asterisk laggo (n=user@80.175.142.209) |
| 15:45.08 | dispy | http://np.gfx-dose.de/1666/na/ <<== does anybody know where the fault is ? |
| 15:45.41 | dispy | if I call with an external telephone that config works correctly |
| 15:45.47 | laggo | can somebody explain to me the difference between FXS and PSTN. don't both of these lead back to the provider? |
| 15:46.16 | dispy | only if I try to access this callthrough internally with asterisk, for exaple with sip/2000, it fails with the error-message given above |
| 15:46.43 | [TK]D-Fender | laggo: No. FXO is for plugging in a LINE. FXS is for plugging in a PHON |
| 15:46.56 | laggo | [TK]D-Fender: i never said FXO |
| 15:47.08 | [TK]D-Fender | laggo: OOps.. |
| 15:47.19 | laggo | i do know the diff between FXO/FXS i think |
| 15:47.27 | [TK]D-Fender | FXO is analog signalling. It is a tech that get you to the PSTN. The PSTN is : |
| 15:47.28 | [TK]D-Fender | ~pstn |
| 15:47.29 | infobot | it has been said that pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 15:47.52 | [TK]D-Fender | laggo: There are many protocols that can get you to the PSTN. ISDN / SS&, etc |
| 15:48.24 | laggo | ahh so the pstn is kind of a term to describe the infrastructure, which may or may not be FXS |
| 15:48.36 | dispy | 2000 is a sip-client with type=friend ^^ |
| 15:48.43 | [TK]D-Fender | laggo: FXO/FXS is boring analog coppr lines. |
| 15:48.47 | laggo | right |
| 15:48.50 | laggo | twisted pairs |
| 15:49.28 | Kobaz | they don't necessarily have to be twisted |
| 15:49.37 | [TK]D-Fender | dispy: This is not a dialplan issue so far. * has no way to contact whoever you are dialing. |
| 15:50.35 | [TK]D-Fender | Kobaz: So * is NAT'd? |
| 15:50.58 | Kobaz | nope |
| 15:51.08 | dispy | Fender what do you mean ? |
| 15:51.18 | [TK]D-Fender | Kobaz: Just runs a private behnd it then? |
| 15:51.19 | dispy | look down at the context provider, that's the context for incoming calls |
| 15:51.28 | Kobaz | [TK]D-Fender: yeah |
| 15:51.35 | dispy | if it's an internal call, exten=>5,1 will be executed |
| 15:51.45 | [TK]D-Fender | dispy: Forget the dialplan. We can see that * does not know how to contact whatever you were dialing. |
| 15:51.48 | dispy | and that uses a Goto-command to jump to the provider-handling |
| 15:52.23 | dispy | sorry I don'T understand you - perhaps caused by my bad english :( |
| 15:53.19 | [TK]D-Fender | dispy: You are DIAL-ing a SIP device. * does not know where to reach them |
| 15:54.22 | laggo | [TK]D-Fender: am i then right in assuming that a copper line labelled "PSTN" is actually an FXS port then? |
| 15:54.59 | Kobaz | [TK]D-Fender: the weird thing... is this same thing worked fine in 1.4 |
| 15:55.04 | Kobaz | [TK]D-Fender: i'm thinking it's a bug? |
| 15:55.07 | [TK]D-Fender | laggo: largely yes, depends on whose POV you are looking at the signalling. |
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| 15:55.40 | dispy | * = the external/internal caller ? I thought so far, Dial() would behave like a bridge between both callers |
| 15:56.06 | [TK]D-Fender | laggo: Typically telcom cards use FXO ports to connect to the telco. THEY are the FXO, therefor the card acts as an FXS |
| 15:56.21 | [TK]D-Fender | dispy: * = ASTERISK |
| 15:56.41 | dispy | the client 2002 is defined as a sip-client in sip.conf ? |
| 15:56.42 | Kobaz | FXS = provides dialtone (ie: the wall jack)... FXO 'recieves it' (ie: a modem) |
| 15:56.42 | [TK]D-Fender | dispy: ASTERISK does not know where the device you want to call is. |
| 15:57.07 | dispy | hm |
| 15:57.07 | [TK]D-Fender | dispy: We don't see what you are dialing. You pasted the error, and not the line that CAUSED it |
| 15:57.13 | laggo | [TK]D-Fender: of course. this port isnt on a telephone, it's ambiguously placed on a voip gateway |
| 15:57.51 | [TK]D-Fender | laggo: That would normally be to connect to the telco. |
| 15:58.20 | dispy | fender that's no line, I just call "5" as the sip-client 2000 and than I press 3 |
| 15:58.21 | laggo | [TK]D-Fender: i.e. an FXS port |
| 15:59.06 | [TK]D-Fender | dispy: exten=>2002,1,Dial(SIP/2002) <- well Asterisk doesn't know where to call to reach SIP/2002 |
| 15:59.20 | dispy | ok |
| 15:59.24 | dispy | how can I tell him ? ;D |
| 15:59.53 | [TK]D-Fender | dispy: You either specify the IP or normally you have that device RESIGTER to * |
| 15:59.59 | [TK]D-Fender | REGISTER |
| 16:00.16 | [TK]D-Fender | Kobaz: I'm a little dry on ideas for your case. |
| 16:00.23 | Kobaz | me too |
| 16:00.53 | dispy | Fender that means, I have to call an asterisk-own client via IP 127.0.0.1 ? |
| 16:01.21 | [TK]D-Fender | dispy: Where is SIP/2002? |
| 16:01.46 | dispy | that#s an internal client defined at sip.conf, in this case logged in via "Phoner", a SIP-Client |
| 16:01.55 | [TK]D-Fender | dispy: WHERE |
| 16:02.10 | [TK]D-Fender | dispy: and "logged in" certainly isn't registered |
| 16:04.50 | dispy | ok "registered" not "logged in " |
| 16:05.38 | dispy | what do you exactly mean by "where" ? I#ve a register-line to register at a sip-provider and for each internal number, 2002 too, I 've [2002] and so on with their settings in sip.conf |
| 16:07.59 | [TK]D-Fender | dispy: * has not REGISTERED! |
| 16:08.09 | [TK]D-Fender | dispy: And * has no idea WHERE they are |
| 16:08.39 | dispy | ok |
| 16:08.47 | dispy | how can I register * itself ? |
| 16:09.05 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 16:09.20 | [TK]D-Fender | dispy: ..... not to itself... |
| 16:09.42 | dispy | to what then ? |
| 16:09.44 | [TK]D-Fender | dispy: SIP/2002. WHAT is it? |
| 16:10.28 | dispy | Protocol/resource |
| 16:10.48 | [TK]D-Fender | dispy: ..... |
| 16:10.59 | dispy | don't know what you're asking sorry :( |
| 16:11.06 | [TK]D-Fender | you said its a SIP Client. * does not have a ADDRESS TO CALL THEM AT |
| 16:12.16 | dispy | If I enter "sip show peers" in the asterisk-console it shows the peers with their IP-address and their(high)ports |
| 16:12.56 | [TK]D-Fender | dispy: PASTEBIN the output of SIP show peer. and pastebin another failed call with SIP DEBUG enabled. "sip set debug" |
| 16:12.59 | [TK]D-Fender | ~pb |
| 16:13.00 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 16:13.01 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 16:13.29 | dispy | thanks :P |
| 16:16.14 | dispy | whow |
| 16:16.17 | dispy | large debug |
| 16:17.32 | dispy | http://np.gfx-dose.de/1667/na/ |
| 16:19.58 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:20.53 | [TK]D-Fender | dispy: 2002/2002 (Unspecified) D N 0 UNKNOWN <--- 2002 has NOT REGISTERED |
| 16:20.59 | [TK]D-Fender | dispy: There is no IP. * has nowhere to call |
| 16:21.46 | dispy | lol |
| 16:21.50 | dispy | öh |
| 16:21.57 | dispy | it's registered |
| 16:22.01 | dispy | phoner shows "registered" |
| 16:22.03 | [TK]D-Fender | DiNo, it ISN'T |
| 16:22.09 | dispy | I cann call internal and external numbers via phoner |
| 16:22.19 | dispy | if the client wouldn't be registered, that won't work ? :) |
| 16:22.29 | [TK]D-Fender | dispy: being able to call * has NOTHING to do with * knowing where to call BACK |
| 16:22.47 | [TK]D-Fender | dispy: You don't need MY phone number for me to call YOU |
| 16:22.56 | dispy | clera |
| 16:22.58 | dispy | *clear |
| 16:23.01 | [TK]D-Fender | dispy: your device knows ASTERISK's IP. |
| 16:23.13 | [TK]D-Fender | dispy: Now go fix the registration |
| 16:23.56 | dispy | aaaaah |
| 16:23.58 | dispy | I'm 2000 |
| 16:24.00 | dispy | the other is 2002 |
| 16:24.02 | dispy | worried ;D |
| 16:24.13 | dispy | haircut, back in ten minutes ;) |
| 16:24.30 | aurman | d-fender: is it possible to have one command in an incoming context when a call comes in from a certain number, and another command for all other calls? |
| 16:30.07 | *** join/#asterisk Sajam (n=chatzill@94.187.17.195) |
| 16:30.24 | [TK]D-Fender | aurYou can do whatever you want |
| 16:30.57 | [TK]D-Fender | aurman: You can do XYZ if its raining in Los Angeles, the Mets won their last home game, and its tuesday night |
| 16:32.11 | aurman | hehe what would the syntax be if i want it to run Dial(SIP/101) only when a call comes in from xxx number? |
| 16:32.42 | Sajam | Hello, need to install asterisk 1.4 on debian, any recommendation reading on how to install it, pdf or website links ? |
| 16:33.15 | aurman | the readme and the docs on the site are pretty good |
| 16:33.41 | Sajam | aurman: asterisk website?? |
| 16:35.55 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 16:37.54 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 16:39.26 | [TK]D-Fender | Sajam: "apt-get install asterisk" |
| 16:39.27 | [TK]D-Fender | DONE |
| 16:39.41 | [TK]D-Fender | aurman: Go read the BOOK... this is in the dialplan basics |
| 16:40.16 | [TK]D-Fender | aurman: This is part of basic extension matching. Or you can check in the exten itself with GotoIF |
| 16:42.01 | *** join/#asterisk eliyahud (n=eliyahu@77.127.163.130) |
| 16:42.31 | PhunTelTek | do i need nat-yes for extensions on the same localnet as *? |
| 16:45.29 | [TK]D-Fender | PhunTelTek: no |
| 16:46.09 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 16:46.51 | Sajam | [TK]D-Fender: i have the asterisk1.4.26.tar.gz on my desktop, apt-get will work?? |
| 16:48.42 | aurman | who'da thought that something that could take two seconds would be refused and something that takes an hour is suggested instead. |
| 16:50.52 | eppigy | calm down |
| 16:51.23 | *** join/#asterisk suma (n=root@c-98-245-185-125.hsd1.co.comcast.net) |
| 16:52.30 | [TK]D-Fender | Sajam: Why would apt-get care what you have on your desktop? |
| 16:52.30 | dispy | Fender: works. thanks :) |
| 16:52.59 | [TK]D-Fender | aurman: Namely? |
| 16:53.47 | Sajam | [TK]D-Fender: ok, it looks on the CD room, what i suggest do i need apt-get to install it since i have it on my desktop, or you prefer to install it using apt-get? |
| 16:55.46 | PhunTelTek | [TK]D-Fender: thanx |
| 16:56.29 | afink | Hello everyone. I am having a little trouble getting DAHDI setup with asterisk. I think I have dahdi configured correctly but * isn't seeing it. * doesn't see any dahdi channels. When I do dahdi show status everything is ok but dahdi show channels returns nothing and pri show span 1 says no pri running on span 1 |
| 16:57.30 | [TK]D-Fender | Sajam: IT? You have a TARBALL on your desktop, not a DEB |
| 16:57.41 | [TK]D-Fender | Sajam: And I suggest youpick a distro you know how to manage |
| 16:58.42 | [TK]D-Fender | afink: And I don't see your configs anywhre. What about the output of "dahdi_cfg -vvvv"? |
| 16:59.16 | eppigy | lol |
| 16:59.18 | Sajam | [TK]D-Fender: Sorry for disturbing, i am trying to get involved in this since i am new, if i want to use apt-get how can i tell it to use the web for installation? your help is high appreciated. |
| 16:59.57 | suma | Is it ok to add a source of GPL V3 to add it to asterisk and release ? |
| 17:00.13 | [TK]D-Fender | Sajam: Then forget the packaged version and go follow the install instructions in the tarball you already downlaoded |
| 17:00.27 | tzafrir_laptop | suma, I suppose that no, unless you have an explicit permission from someone |
| 17:00.51 | tzafrir_laptop | suma, any specific GPLv3 component? |
| 17:00.58 | suma | tzafrir_laptop: libs3 |
| 17:01.03 | [TK]D-Fender | suma : You can do whatever you want to YOURS, but you cannot redistribute them together, nor will * include them |
| 17:01.23 | tzafrir_laptop | suma, what is it? |
| 17:01.42 | eppigy | suma has root on the svn box |
| 17:02.08 | suma | having amazon s3 storage as realtime resource for asterisk |
| 17:07.54 | aurman | GotoIf doesn't seem to be working for me as per docs |
| 17:08.32 | afink | [TK]D-Fender: here is dahdi_cfg -vvv: http://pastebin.com/m33b3d7d7 configs on the way: http://pastebin.com/m33b3d7d7 |
| 17:08.43 | afink | oops |
| 17:09.34 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 17:12.50 | [TK]D-Fender | aurman: It works just fine |
| 17:13.19 | [TK]D-Fender | aurman: You either do not understand * expressions or the syntax for GotoIF. |
| 17:14.32 | afink | [TK]D-Fender: system.conf: http://pastebin.com/m72949061 |
| 17:14.56 | [TK]D-Fender | ... |
| 17:15.54 | afink | I'm getting the other one too |
| 17:17.53 | Sajam | any help on how to install a asterisk tarball on debian OS |
| 17:18.31 | suma | Sajam: Is it any different from other linux OS ? |
| 17:18.37 | suma | i mean other linux flavours |
| 17:19.12 | aurman | i just follow the readme, mostly with the ./configure;make;make install with each of the requirements and then finally asterisk |
| 17:19.18 | Sajam | suma: ok great, i just want to know how can i install it, i am newbie to both of them |
| 17:19.19 | *** join/#asterisk stope (n=nobody@sud-cable-cmts3-69-60-242-213.vianet.ca) |
| 17:19.21 | [TK]D-Fender | Sajam: Follow the INSTRUCTIONS in the tarball |
| 17:19.28 | afink | chan_dahdi.conf: http://pastebin.com/m49c20705 |
| 17:19.53 | Sajam | where can i find INSTRUCTIONS in the tarball, |
| 17:19.58 | *** join/#asterisk S2AnGeL (n=S2AnGeL@CPE001839f225cc-CM0011aea1c1ca.cpe.net.cable.rogers.com) |
| 17:20.00 | [TK]D-Fender | afink: Permanently remove all that commented garbage and re-PB |
| 17:20.27 | [TK]D-Fender | afink: 1300 lines of unadulterated TRASH |
| 17:20.35 | [TK]D-Fender | afink: Shouldn't be more than 30 lines |
| 17:20.59 | afink | ok, will do thought about doing that. Its working now |
| 17:21.25 | aurman | waits for the phone to be free again for testing. |
| 17:21.57 | [TK]D-Fender | Sajam: How about starting with the blatant "README" |
| 17:22.17 | [TK]D-Fender | afink: Whats working? DAHDI? |
| 17:23.42 | afink | yes and the channels. output is good now |
| 17:23.50 | stope | leifmadsen: setting TDMV_HW_DTMF = NO on wanpipe config gets rid of the one way audio and the following message on the cli... WARNING[9581]: rtp.c:2248 ast_rtp_senddigit_begin: Don't know how to represent 'f' |
| 17:24.22 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:24.33 | [TK]D-Fender | afink: Probably failed because you didn't do DAHDI-CFG before starting * and it didn't initialize the channels |
| 17:24.51 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 17:25.10 | afink | a good possibility I restarted * and dahdi so many times I can't remember |
| 17:26.34 | S2AnGeL | I am trying to do a zaptel flash bilind transfer although it just bridges with anouther zap channel and calls out is there a way to take advantage of the BELL blind transfer from my provider |
| 17:26.51 | [TK]D-Fender | S2AnGeL: You can almost forget about this via FreePBX |
| 17:27.09 | [TK]D-Fender | S2AnGeL: Your Flash is on the local end, it doesn't get passed on to the other side of a bridged call. |
| 17:27.33 | [TK]D-Fender | S2AnGeL: You ned a dynamic feature configured to call Flash() which is WAY outside the scope of the GUI |
| 17:28.53 | S2AnGeL | I figured as much. But where to start where to read on it. |
| 17:29.24 | [TK]D-Fender | S2AnGeL: features.conf + "core show application flash" |
| 17:29.45 | [TK]D-Fender | S2AnGeL: this will be triggered by DTMF to send the flash to the remote end |
| 17:29.53 | S2AnGeL | I want to avoid the bridge part and just take a call and flash send it off using BELLs network |
| 17:30.04 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 17:30.21 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 17:31.16 | S2AnGeL | sendDTMF |
| 17:31.52 | [TK]D-Fender | S2AnGeL: Flash + SendDTMF |
| 17:32.06 | S2AnGeL | so I should be reading up on making a feature code that will do a SendDTMF |
| 17:32.20 | S2AnGeL | hey thanks I am looking into it now any links you have would be great |
| 17:33.25 | [TK]D-Fender | S2AnGeL: Read the sample configs |
| 17:34.14 | S2AnGeL | ok |
| 17:36.06 | *** join/#asterisk psykon (n=psilikon@140-1.35-65.tampabay.res.rr.com) |
| 17:40.02 | [TK]D-Fender | out for a while, BBIAB |
| 17:45.18 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 18:05.10 | psykon | I'm not having much luck googling up a howto on Sipura ATA + Asterisk + Voip SIP provider |
| 18:06.40 | *** join/#asterisk Sajam (n=chatzill@94.187.17.195) |
| 18:11.31 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 18:15.58 | *** join/#asterisk rhombus (n=rhombus@dsl-vlan435-66-18-218-36.nucleus.com) |
| 18:17.02 | rhombus | Does anybody have any tips for controlling AC hum? I just moved a phone to a different location and now I'm getting AC hum in it. I've tried plugging it into different outlets in the same room, different power bars, turning off lights -- no dice. |
| 18:32.29 | PhunTelTek | light dimmers create noise on the ac lines. |
| 18:33.50 | PhunTelTek | so do touch lamps. And you have to unplug them to see if they are the problem. |
| 18:37.04 | rhombus | PhunTelTek: Yeah, I tried it with all the lights off, no difference; also, hum goes away when I use the handset, as opposed to the headset. |
| 18:37.42 | PhunTelTek | headset plugged into handset? |
| 18:37.45 | rhombus | PhunTelTek: It's very faint when I use the handset, but only because the person on the other end knows to listen for it. |
| 18:37.59 | rhombus | PhunTelTek: Headset plugged into the headset port on the back of the phone. |
| 18:38.23 | PhunTelTek | expensive headset? old? |
| 18:39.17 | rhombus | PhunTelTek: Headset is a Plantronics with a noise canceller, the amplifier is a Plantronics Vista M12. This worked fine in the other office. |
| 18:39.47 | rhombus | PhunTelTek: I have no reason to think there's anything inherently wrong with the headset/amp assembly. |
| 18:40.14 | PhunTelTek | VoIP phone? |
| 18:40.44 | rhombus | PhunTelTek: Yeah, it's a Polycom IP501... that's what makes this more difficult to troubleshoot versus an analog phone :) |
| 18:41.43 | PhunTelTek | well we know the noise isn't being picked up on RJ11 flat cable. is it PoE? |
| 18:42.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:42.29 | rhombus | PhunTelTek: No. There is a PoE switch in the space, but this phone is getting juice from a brick. |
| 18:43.03 | rhombus | PhunTelTek: because it's plugged into an intermediate switch. |
| 18:43.54 | PhunTelTek | buy an outlet tester, and make sure all of your outlets are wired correctly. |
| 18:44.31 | *** join/#asterisk toughmarketing (n=toughmar@ip72-199-181-246.sd.sd.cox.net) |
| 18:44.48 | PhunTelTek | The noise is either conducted or radiated. It could be that the headset is just worse in this location because there is noise. |
| 18:45.23 | rhombus | PhunTelTek: Well, there is a UPS between this device and the wall outlet. UPSes test the line and, depending on how they are wired, can act as line conditioners. The wiring light is green on the UPS... |
| 18:46.07 | rhombus | PhunTelTek: You mean noise from AC? |
| 18:46.42 | PhunTelTek | cheap UPSs also use a modified square wave, which can cause problems with some equipment. |
| 18:47.18 | *** join/#asterisk P0RN5T4R (n=P0RN5T4R@213.63.190.66) |
| 18:50.22 | PhunTelTek | do you know if it is an online or offline UPS? |
| 18:53.33 | rhombus | PhunTelTek: I don't, no. I don't know how would I tell, either. |
| 18:53.46 | rhombus | PhunTelTek: It's a typical APC UPS. |
| 18:53.49 | PhunTelTek | an online UPS is always running, an offline UPS switches on when it detects a power issue. |
| 18:54.34 | rhombus | PhunTelTek: Well, there is definitely a relay in it -- it will click when the power goes. |
| 18:54.51 | rhombus | PhunTelTek: And it's a BackUPS, which is the cheaper line. |
| 18:55.41 | PhunTelTek | offline then, it shouldn't create noise unless it's switched over. You can remove it and try it though. probably not he cause though. |
| 18:56.19 | PhunTelTek | apartment or house? |
| 18:56.27 | rhombus | PhunTelTek: Seems doubtful. I am thinking it's something about the wiring in this space. I'm just wondering if there's a way I could filter it -- but you say it might also be radiated, in which case, filtering won't help. |
| 18:56.31 | rhombus | PhunTelTek: It's a house. |
| 18:57.54 | PhunTelTek | radiated would have to be stronger to cause you problems. conducted is more likely. check everything on that circuit. |
| 18:58.08 | PhunTelTek | does it come and go? |
| 18:58.55 | rhombus | PhunTelTek: Only in the sense that it's most audible to remote callers when I am speaking -- the varistor in the headset amplifier doing that, probably |
| 18:59.03 | rhombus | PhunTelTek: But for me, it's constant. |
| 19:00.00 | PhunTelTek | do you plug it into a different phone at work? |
| 19:00.09 | *** join/#asterisk FreakGuard (n=freak@unaffiliated/freakguard) |
| 19:00.57 | rhombus | PhunTelTek: No. Both the phone and the headset assembly were moved together, and they were in the same house before. |
| 19:01.35 | FreakGuard | I'm looking for a sample script in lua for call forwarding... anyone got something in handy? |
| 19:02.57 | PhunTelTek | rhombus: in the other office they were fine workiing together? |
| 19:03.18 | rhombus | PhunTelTek: Yeah. |
| 19:03.39 | rhombus | PhunTelTek: This is now in the basement, and on a different circuit. |
| 19:03.47 | rhombus | PhunTelTek: Beyond that, everything is the same. |
| 19:03.48 | PhunTelTek | other office in same house or different location? |
| 19:03.57 | rhombus | PhunTelTek: Other office in the same house. |
| 19:04.11 | PhunTelTek | got fluorescent lights anywhere? |
| 19:04.42 | elitecoder | hey guys, I made an auto dialer and I have 7 lines from bandwidth.com. I'm getting about 50% congestion. It's really affecting performance. Does anyone know why this is happening? |
| 19:05.57 | elitecoder | I've watched the channels, it never goes above 7, which is the number of "sip trunks" from bw.com we have. |
| 19:08.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:09.48 | rhombus | PhunTelTek: There are compact fluorescents, but when I turned off all the lights, the hum was still present. |
| 19:10.06 | rhombus | PhunTelTek: There is a motor that runs constantly, for a fan in the attic. |
| 19:11.02 | PhunTelTek | You'd have to be closer to that, unless it's on the same circuit/leg. |
| 19:12.03 | PhunTelTek | get an extension cord and plug the brick into the old outlet, see if it gets better. |
| 19:13.22 | rhombus | PhunTelTek: might be hard, it's way upstairs :) |
| 19:13.34 | rhombus | PhunTelTek: I'll need a super long extension cord, but I can try that. |
| 19:13.51 | rhombus | PhunTelTek: What about if the circuits are proximal in the panel? |
| 19:14.29 | PhunTelTek | there could be some crossover |
| 19:15.30 | rhombus | rhombus: I could try tripping the breaker for that circuit, too. |
| 19:15.39 | rhombus | likes talking to himself. |
| 19:15.45 | rhombus | PhunTelTek: I could try tripping the breaker for that circuit, too. |
| 19:16.13 | PhunTelTek | yep |
| 19:16.57 | rhombus | PhunTelTek: Okay -- I have some things I can try. Thanks for your help! |
| 19:17.38 | PhunTelTek | yep, have fun |
| 19:19.53 | Orbixx | Any known issues with Asterisk and OpenVZ? |
| 19:22.21 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:22.27 | elitecoder | :o woo |
| 19:22.53 | elitecoder | Orbixx: I think a better idea would be to go check their respective bug report tools |
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| 19:23.08 | elitecoder | in any large project there are usually a lot of open issues |
| 19:26.20 | toughmarketing | Hey guys I am using odbc with mysql and storing the extensions in my database in the format of: id,context,exten,priority,app,appdata and this is working great! The only issue is at the top of my context for my ivr I have include => ivr1-day,09:00-16:59,mon-fri,*,* and if it is during those hours and days it goes to ivr1-day context... Is there a way I can include this in the database as well? |
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| 20:41.46 | PhunTelTek | when i try to transfer a call to an outside number, i get the "this number does not accept blocked calls" messgage. Is there a way to pass the CID? it works when i dial the number directly. Just not for transfers. |
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| 20:46.40 | leifmadsen | stope: thanks! |
| 20:46.44 | leifmadsen | I'll update the blog |
| 20:47.35 | leifmadsen | (and done) |
| 20:50.59 | *** join/#asterisk Alfio (n=Amunoz@adsl-51-21.tricom.net) |
| 20:54.43 | *** join/#asterisk S2AnGeL (n=S2AnGeL@74.12.50.212) |
| 21:06.13 | S2AnGeL | http://pastebin.com/d6d797a30 I am trying to make this work |
| 21:06.52 | S2AnGeL | it just hangs up the call does not even play the pls-wait-connect-call |
| 21:07.21 | S2AnGeL | I need some sort of tip or push in the right direction |
| 21:09.52 | carrar | isn't the value of ${EXTEN} 's' |
| 21:10.11 | carrar | since thats the extension you are in |
| 21:10.59 | carrar | change to ${ARG1} |
| 21:11.04 | kaldemar | "-- Executing [s@macro-dial:7] Dial("Zap/5-1", "custom-call_cell,theNumberIamTryingToGetItToDial,1||tr") in new stack" also looks wrong. Dial(Tech/...) is the syntax, see core show application Dial. |
| 21:12.07 | carrar | I'll assume they editted it out |
| 21:12.10 | carrar | heh |
| 21:14.24 | kaldemar | sure, the number. but dial doesn't take a context, number and a priority. Dial(Local/theNumber@custom-call_cell,,tr) would be right if the exten in custom-call_cell is changed from s to something that matches the given number. |
| 21:15.44 | kaldemar | and changing EXTEN to ARG1 won't make it work if there is nothing passing an argument. |
| 21:22.24 | *** join/#asterisk reg (n=reg@2a01:240:fe29:1:0:0:101:dead) |
| 21:23.24 | ManxPower | In a macro EXTEN is "s" |
| 21:23.52 | ManxPower | if you want the extension that it was when the macro was called you would use MACRO_EXTEN. |
| 21:24.00 | ManxPower | This should be documented in channelvariables.txt |
| 21:24.54 | ManxPower | also you NEVER EVER want to use "r" without a very, very good reason. Asterisk will by default provide a ringing tone. |
| 21:26.41 | kaldemar | that is not a macro. |
| 21:27.45 | kaldemar | a mere context that he's trying to reach with a dial. |
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| 21:50.38 | zr0 | is an isdn line physically different from an analog pots line? or is it just the physical layer protocol different? |
| 21:52.55 | S2AnGeL | I am not sure if it should be s or what ever I just put s in there I had it at _X. |
| 21:54.22 | ManxPower | zr0: ISDN BRI and POTS both run on two wires. That's about the only thing they have in common. |
| 21:55.30 | S2AnGeL | its to take a incoming call that someone hits this extension and they get passed that |
| 21:56.09 | S2AnGeL | it should flash the zap and hang up sending the call through the BELL transfer command |
| 21:56.17 | S2AnGeL | freeing up the zap |
| 21:57.01 | ManxPower | S2AnGeL: does the line have Conference/Drop/Transfer feature or just 3-way calling? |
| 21:57.14 | zr0 | ManxPower: cool, thx |
| 21:57.41 | zr0 | ManxPower: do you use ISDN BRI? |
| 21:58.27 | ManxPower | zr0: not in years. Historically it's not been well supported in Asterisk. What country are you in? |
| 21:58.52 | S2AnGeL | Its a Bell business line with the ability to recieve a call flash hook dial anouther number and hang up (transfering the call on the BELL CO keeping my line free.. I am unsure of the exact terminolighy |
| 22:00.47 | S2AnGeL | I changed the EXTEN to ARG1 but I fear I am not passing any argument |
| 22:01.30 | S2AnGeL | it just hangs up when anyone calls in and types in 921 extention |
| 22:02.50 | S2AnGeL | Its a recent install of trixbox. works great for incoming calls and all just has to be a way to hookflash and send a call off |
| 22:04.54 | ManxPower | try asking on a channel where people use Trixbox, like #trixbox |
| 22:07.08 | S2AnGeL | http://pastebin.com/d89d873d |
| 22:07.44 | S2AnGeL | really seems to ignor what I entered .. I think I have no idea how to pass the numberIamTryIngToGetItToDial |
| 22:09.28 | S2AnGeL | should I use _X! instead of s |
| 22:10.22 | S2AnGeL | maybe I should use _. |
| 22:10.59 | kaldemar | never use _., it matches to special extensions such as i, t, and h. |
| 22:11.52 | S2AnGeL | Good point |
| 22:11.54 | jblack | s,h,i, and t |
| 22:12.09 | jblack | coincidence? I think not! |
| 22:12.29 | S2AnGeL | or _XXX since its a 3 digit extention I am using |
| 22:12.45 | S2AnGeL | lol |
| 22:13.07 | zr0 | ManxPower: bummer. i was considering it as the most reliable landline i could get for doing my own voicemail. |
| 22:13.22 | zr0 | ManxPower: guess i'll just get a regular landline. |
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| 22:19.35 | [TK]D-Fender | jblack: Actually its "O SHIT A FAX"! |
| 22:20.40 | [TK]D-Fender | S2AnGeL: What have you done with features.conf? |
| 22:21.28 | S2AnGeL | nothing yet I just trying to make when I dial a extention that it dials my cell and it does not tie up the phone |
| 22:22.15 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 22:22.29 | shmaltz | why am I getting errors compiling zaptel? |
| 22:22.44 | S2AnGeL | HOw the heck do i make it dial the number.. I mean I hear it flash.. then well I hear a dial tone becuase it hung up on me.. did'nt pass nothing so I need to know how to pass the number to the custom-call_cell |
| 22:23.18 | S2AnGeL | features.conf is what ever it comes stock with for trixbox what ever is in there |
| 22:23.19 | shmaltz | here is the error: |
| 22:23.20 | shmaltz | http://pastebin.ca/1500494 |
| 22:27.12 | [TK]D-Fender | S2AnGeL: SendDTMF is it... just put some "w"'s before the number to delay the dialing |
| 22:28.00 | ManxPower | shmaltz: is that the latest 1.2.x zaptel? |
| 22:28.36 | shmaltz | ManxPower, yes |
| 22:28.48 | ManxPower | weird |
| 22:29.28 | ManxPower | shmaltz: try doing a test compile on the linux kernel, then try compiling zaptel again. |
| 22:29.39 | shmaltz | zap 1.2.27 |
| 22:29.42 | ManxPower | (nod need to install the kernel) |
| 22:29.48 | ManxPower | s/nod/no/ |
| 22:30.32 | shmaltz | how do i do that? |
| 22:30.55 | ManxPower | shmaltz: There are a billion documents on how to do that |
| 22:31.21 | shmaltz | doesnt even know about one of those millions |
| 22:31.31 | shmaltz | or billions |
| 22:33.42 | ManxPower | http://www.cyberciti.biz/tips/compiling-linux-kernel-26.html |
| 22:33.47 | ManxPower | first damn google hit. |
| 22:34.25 | ManxPower | if you are able to compile the linux kernel on your machine then you will know that everything is installed that zaptel requires (at least for the kernel) |
| 22:34.52 | ManxPower | you can assume you already have the linux kernel installed. |
| 22:35.19 | ManxPower | also remember compile zaptel against the same version of the kernel source as the kernel version you are using. |
| 22:35.54 | ManxPower | just take the defaults from "make menuconfig" and don't actually install the compiled kernel. |
| 22:36.01 | *** part/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 22:38.06 | shmaltz | ManxPower, this is a custom compiled kernel |
| 22:38.27 | shmaltz | I'm running this off a custom compiled kernel, system boots fine, everything else works |
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| 22:48.47 | ChUbB | hi guys i am looking to setup a asterisk server for abit of fun whats the cheapest hand sets i can get ? |
| 22:49.39 | shmaltz | ChUbB, have you tried a soft phone? |
| 22:49.51 | shmaltz | ~softphone |
| 22:49.52 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 22:50.07 | shmaltz | ~xlite |
| 22:50.07 | infobot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 22:52.18 | ChUbB | safe |
| 22:59.28 | zr0 | anybody use voip.ms? |
| 23:02.37 | afink | how can I get rid of this? chan_dahdi.c:1774 dahdi_enable_ec: Unable to enable echo cancellation on channel 23 or is it a problem? I have hardware echo cancellation |
| 23:07.35 | shmaltz | ManxPower, what could be the problem? |
| 23:11.40 | *** part/#asterisk sfire (n=sfire@businessservers.info) |
| 23:12.32 | shmaltz | can someone help me? I'm tryint to complie Zaptel and I"m getting thsese errors: |
| 23:12.34 | shmaltz | http://pastebin.ca/1500527 |
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| 23:57.32 | rhombus | shmaltz: Is that a supported kernel? |