| 00:02.01 | p3nguin | Adding that localnet didn't help. Here's sip debug and rtp debug: http://pastebin.com/DbRj4K8V |
| 00:04.34 | p3nguin | Peer audio RTP is at port 172.16.255.21:16428 <-- this seems like a problem. |
| 00:05.01 | *** join/#asterisk kazaa_lite (~eddie@212.183.140.2) |
| 00:06.27 | kazaa_lite | hi all |
| 00:07.37 | lepine | I remember reading about playing with goto and contexts recursively ... and that page mentionned being able to refer to the parent context (eg, goto) ... |
| 00:07.59 | lepine | Can someone point me in the right direction? (eg, wiki page name or topic) |
| 00:09.14 | lepine | <PROTECTED> |
| 00:10.13 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
| 00:11.43 | p3nguin | [tk]d-fender: Is that enough debug to see what's going on? |
| 00:21.43 | *** join/#asterisk cidu (~PISSSSSSS@whthyt253-29.northwestel.net) |
| 00:23.14 | cidu | so, quick question here, having trouble with CALLERID(ANI) over PRI, cant seem to get asterisk to recieve, or transmit CALLERIS(ANI) properly, its always populated with the CALLERID(NUM) Value and not the charge number data |
| 00:23.27 | cidu | is this a known bug? or am i overlooking something simple |
| 00:29.42 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 00:30.15 | QbY | with a realtime voicemail configuration, how does one force * to re-load the voicemail table? |
| 00:30.44 | cidu | no clue, never used realtime, and the channel is really quiet |
| 00:37.27 | *** join/#asterisk coppice (~chatzilla@m121-202-34-88.smartone-vodafone.com) |
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| 00:46.18 | LemensTS | anyway to remove a user when you do sip show peers? |
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| 01:21.29 | prgmrchris | can someone help me figure out why my dialplan doesnt work? im really new to this and im sure its a simple mistake: http://pastebin.com/wWFshffM |
| 01:24.06 | *** join/#asterisk mpd (~chatzilla@bas1-malton22-1167904914.dsl.bell.ca) |
| 01:24.43 | *** part/#asterisk mpd (~chatzilla@bas1-malton22-1167904914.dsl.bell.ca) |
| 01:25.16 | carrar | ~book |
| 01:25.16 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:25.40 | prgmrchris | ? |
| 01:26.41 | carrar | exten matches on DNIS not DID |
| 01:27.02 | carrar | or ANI |
| 01:27.26 | prgmrchris | so how do i write a rule for a certain DID? |
| 01:27.50 | prgmrchris | what do i use instead of exten |
| 01:27.56 | TJNII | "from-trunk" freepbx? |
| 01:28.02 | carrar | You are matching on DNIS or ANI? |
| 01:28.09 | prgmrchris | carrar: yeah thats what im trying to do |
| 01:28.37 | prgmrchris | im looking at http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns for general info, if you know any other good links that might help it would be appreciated :) |
| 01:29.08 | TJNII | prgmrchris: Did you install from source or are you using some packages Asterisk? |
| 01:29.19 | prgmrchris | TJNII: does that matter? |
| 01:29.59 | prgmrchris | the question is pretty generic i dont think the packaging matters in this case |
| 01:30.07 | TJNII | Yes, because if you used FreePBX or TrixBox your dialplan is wrapped in their configs and your call could be going off into GUI la-la land. |
| 01:30.21 | carrar | prgmrchris, install Asterisk from Source |
| 01:30.25 | carrar | then come back |
| 01:30.33 | prgmrchris | the distribution has freepbx but there is no incoming routes or anything in freepbx |
| 01:30.40 | TJNII | ~freepbx |
| 01:30.40 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 01:30.45 | carrar | or try #Freepbx |
| 01:31.05 | prgmrchris | i have another box which is from source |
| 01:31.09 | TJNII | Not trying to be rude, but it is damn hard to know what that dialplan is doing, due to the complexity. |
| 01:31.10 | prgmrchris | and it doesnt work there either |
| 01:31.18 | prgmrchris | i dont think its an issue with that |
| 01:31.24 | TJNII | pastebin from the source box. |
| 01:31.30 | prgmrchris | ok |
| 01:31.30 | TJNII | We can help you a lot better. |
| 01:31.32 | carrar | prgmrchris, install Asterisk from Source |
| 01:31.36 | carrar | then we can help |
| 01:31.37 | prgmrchris | carrar: i did, read above |
| 01:32.34 | carrar | also check out that book |
| 01:32.39 | carrar | that goes over basic dialplan |
| 01:33.35 | carrar | Why did you paste the context [test]? |
| 01:34.51 | prgmrchris | i thought it needed to be included to work |
| 01:34.56 | prgmrchris | im checking out the book now, thanks |
| 01:35.20 | carrar | plus you're using freepbx |
| 01:35.24 | carrar | good luck |
| 01:35.46 | *** join/#asterisk encinoman (~matthew@pool-74-102-71-73.nwrknj.fios.verizon.net) |
| 01:35.53 | prgmrchris | carrar: im not right now. no |
| 01:36.02 | carrar | That paste is |
| 01:36.26 | prgmrchris | i know it is, if you read what i wrote earlier you will see that i said the first box had freepbx and the second box doesnt |
| 01:36.34 | prgmrchris | ive been pretty clear about that |
| 01:36.49 | carrar | don't paste irrelevant stuff for us to fix |
| 01:37.09 | encinoman | Anyone have any advice on implementing call conferencing? |
| 01:37.12 | prgmrchris | its not irrelevant, it was a honest question, to which TJNII advised to compile from source which i have |
| 01:37.15 | TJNII | Well pastebin the output of the second box. Then we can try and help you spot the error without the freePBX junk. |
| 01:37.46 | prgmrchris | TJNII: i will, im going to read the book some first and see if i can figure it out myself first, more rewarding :) if i get stuck i will ask again |
| 01:37.49 | TJNII | I don't think you have a dialplan error, but I need some output to back that up. |
| 01:37.51 | prgmrchris | thanks for the help |
| 01:38.28 | TJNII | Okay, make sure to verify your contexts in sip.conf. |
| 01:39.33 | *** join/#asterisk halkun (~chatzilla@mke-66-97-112-72.milwpc.com) |
| 01:39.44 | p3nguin | I've encountered a NAT issue where the remote device's RTP packets have the PRIVATE IP address on them instead of the PUBLIC address. If anyone cares to help, here's the sip debug and rtp debug: http://pastebin.com/DbRj4K8V |
| 01:40.13 | p3nguin | The problem is related to this line: Peer audio RTP is at port 172.16.255.21:16428 |
| 01:41.04 | p3nguin | The peer's config includes nat=yes, and the localnet setting is including all RFC 1918 ranges. |
| 01:42.07 | p3nguin | Using a softphone rather than the SPA-3102 on that LAN works fine. The RTP packets have the public address on them when using zoiper. |
| 01:44.56 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 01:49.55 | LemensTS | MoOo |
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| 02:10.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:19.14 | *** join/#asterisk geneticx (~geneticx@adsl-2-214-190.mia.bellsouth.net) |
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| 02:36.39 | [TK]D-Fender | p3nguin: pastebi your sip.conf masking only passwords ([general] and [203]) |
| 02:38.20 | *** join/#asterisk geneticx (~geneticx@adsl-10-113-95.mia.bellsouth.net) |
| 02:40.58 | [TK]D-Fender | prgmrchris: Your dialplan looks fine. It is however completely unused. |
| 02:41.49 | coppice | you mean "as new condition"? :-) |
| 02:42.51 | [TK]D-Fender | coppice: "new code smell" |
| 02:44.44 | *** join/#asterisk gospch_ (~gospch@unaffiliated/gospch) |
| 02:47.43 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 02:48.33 | p3nguin | sip.conf: http://pastebin.com/F6biEM9y |
| 02:49.43 | p3nguin | Erm, wait... something was cut off the bottom. |
| 02:52.54 | [TK]D-Fender | p3nguin: Um, is this SPA just completely remote? No VPN or anything, right? |
| 02:53.59 | p3nguin | That's correct, simply a remote LAN using regular old UDP for SIP/RTP. |
| 02:54.24 | [TK]D-Fender | p3nguin: If you don't have those subnets we talked about as actually local, trash them |
| 02:54.31 | [TK]D-Fender | (localnet) |
| 02:54.37 | [TK]D-Fender | p3nguin: Just keep the real one. |
| 02:55.10 | p3nguin | A few of the settings for the peer got cut off in the first paste, not sure what happened. This is the re-paste if you need it: http://pastebin.com/uxppq0mp |
| 02:55.22 | [TK]D-Fender | p3nguin: Waitasec.... were you working on SIP AGL settings on a Cisco for this earlier with MmanxPower? |
| 02:55.43 | p3nguin | He insisted I'm using it, but I'm not. |
| 02:55.59 | [TK]D-Fender | p3nguin: what routers are on each end? |
| 02:56.23 | p3nguin | Linux box/iptables on the client side, Cisco 831 on the server side. |
| 02:57.14 | coppice | [TK]D-Fender: new code usually stinks |
| 02:57.22 | p3nguin | For what it's worth, I started out with only my real localnet in the conf, then added the rest earlier. |
| 02:59.08 | [TK]D-Fender | p3nguin: Which end doesn't get audio? |
| 03:00.12 | p3nguin | neither way, two-way no audio |
| 03:00.41 | p3nguin | placing a call to that device or calling from that device, same result, no audio at all. |
| 03:00.42 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 03:01.46 | p3nguin | It puzzles me that zoiper on the computer on that remote LAN puts the public IP address in the RTP packets, but this SPA device is putting the private on the packets. |
| 03:02.02 | p3nguin | and, of course, zoiper works. |
| 03:02.08 | p3nguin | good audio with it. |
| 03:04.45 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:05.28 | [TK]D-Fender | p3nguin: ensure that there is NO NAT awareness on the SPA |
| 03:05.39 | [TK]D-Fender | p3nguin: Also, what port is it connected via? |
| 03:05.53 | p3nguin | I checked it, all boxes are either cleared or marked NO. |
| 03:06.16 | p3nguin | Hmm, you mean either the WAN/Internet or the LAN port on the device? |
| 03:06.53 | p3nguin | If so, I'm actually not sure about that. |
| 03:07.15 | p3nguin | Which way is ideal? |
| 03:09.28 | p3nguin | It is connected via WAN port. |
| 03:09.34 | p3nguin | Nothing is on the LAN port. |
| 03:10.23 | [TK]D-Fender | p3nguin: OK so far.... wondering if something else is off... |
| 03:27.22 | p3nguin | It's a puzzler. |
| 03:34.51 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-147-211.dsl.stlsmo.sbcglobal.net) |
| 03:45.24 | *** join/#asterisk kartik (~koolkarti@117.199.112.58) |
| 03:46.54 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 03:47.57 | sawgood | I get now ... I really do ... I understand and appreciate using Asterisk CLI only (even Asterisk 1.4.x) ... but the learning curve for voicemail and IVR and ring groups gets steep |
| 03:48.09 | sawgood | having FreePBX to fall back on for these things is so tempting |
| 03:48.20 | sawgood | I guess I'll take a break for a while |
| 03:48.52 | [TK]D-Fender | sawgood: Learning curve.... for voicemail? |
| 03:49.06 | sawgood | well, just needed a place to vent ... so to speak |
| 03:49.21 | sawgood | having to build the entire Asterisk dialplan from code is time consuming and hard |
| 03:49.23 | coppice | [TK]D-Fender: maybe the voice mail is in a foreign language |
| 03:49.37 | carrar | let it out |
| 03:49.38 | [TK]D-Fender | sawgood: Whats to learn? a VM box is 1 line of config. Sending a call to leve a voicemail is also 1 line of dialplan. Picking up voicemail is also 1 single line of dialplan... |
| 03:49.48 | sawgood | in general I meant not having a set of GUI tools |
| 03:50.05 | carrar | haha' |
| 03:50.10 | carrar | seriously? |
| 03:50.12 | sawgood | maybe voicemail was a stretch .... I really meant IVR ... and what not |
| 03:50.32 | carrar | GUI is restrictive |
| 03:50.39 | sawgood | you can say that again ...! |
| 03:50.58 | [TK]D-Fender | sawgood: Dial(SIP/100@SIP/110@SIP/120,20) <-- congrats.. a "ring-group" .... in again... 1 LINE OF DIALPLAN |
| 03:51.00 | sawgood | is there an 'easy' way to have grep print out to the screen only lines which do NOT start with ; |
| 03:51.34 | carrar | grep -v "^;" poop.txt |
| 03:51.42 | sawgood | thank you |
| 03:52.18 | carrar | But you should remove all the lines with ; anyways |
| 03:52.29 | carrar | unless you added them for a reason |
| 03:53.35 | sawgood | what does the ^ sign do for the syntax? |
| 03:53.46 | carrar | please visit regex |
| 03:53.55 | sawgood | don't get me wrong it worked just as advertised ... |
| 03:53.58 | sawgood | just trying to learn |
| 03:54.03 | carrar | beginign of line |
| 03:54.46 | sawgood | I love that O'Reilly book, "Understanding Regular Expressions" .... I might should read it again soon |
| 03:55.04 | carrar | or just google |
| 03:55.45 | carrar | they are all in google books |
| 03:58.14 | carrar | I hsould visit this |
| 03:58.14 | carrar | http://en.akihabaranews.com/44454/misc/bandai-open-its-first-gundam-cafe-in-akihabara%E2%80%A6-and-we-tested-it |
| 03:58.22 | carrar | no idea what Gundam is though |
| 03:59.20 | carrar | Barista loks cute |
| 04:00.29 | carrar | sawgood, you remove your comment lines yet? |
| 04:02.36 | sawgood | I did ... thank you |
| 04:03.06 | carrar | You'll be graded on a clean and orderly extensions.conf |
| 04:03.42 | sawgood | I can easily make the SIP trunk work if I set the box up on a FreePBX build, but Asterisk only (which you would think should be easier) is not working |
| 04:03.50 | sawgood | I know I have something incorrect in sip.conf |
| 04:03.55 | sawgood | I am getting closer and closer |
| 04:06.53 | *** join/#asterisk adam_g (~adam@c-67-189-34-196.hsd1.or.comcast.net) |
| 04:07.36 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 04:09.45 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 04:15.19 | halkun | Ok, I have an account in asterisk, I and the server talks to my xlite. I have the 3102 on the network and can see it's config page when I go it it's IP address. I need some instruction on how to link the 3102 to asterisk so I can make calls on my land line. |
| 04:15.41 | halkun | (Almost there :)) |
| 04:16.44 | [TK]D-Fender | halkun: Dial(SIP/peertothe3102fxoport/1234567890) |
| 04:18.53 | adam_g | hmm. strange issue where we've switched from nat = route to nat = no, all but one SIP peers is working as it should. the other still shows Addr->IP = its old source, as if it were still being natted. also with a weird port ( 1029 as opposed to 5060), any ideas? |
| 04:19.26 | adam_g | do such address get cached for some period? rtcachefriends is not set to yes (though not explicitly set no) |
| 04:25.18 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 04:26.27 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 04:30.54 | halkun | I found this as a quick setup |
| 04:30.59 | halkun | http://forum.voxilla.com/cisco-linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html |
| 04:32.28 | halkun | however, it's a little different than "default" as I don't need 10 digit dialing and asterisk is the internet gateway while the 3102 is behind that. It appears to be the reverse on this setup. |
| 04:34.48 | halkun | the dial plan from that setup is Dial Plan: (*xx|[3469]11|0|00|[2-9]xxxxxxxxxS0|1xxx[2-9]xxxxxxS0) |
| 04:35.04 | halkun | is there a place I can go to decipher that? |
| 04:36.18 | [TK]D-Fender | halkun: thats from the FXS portion, right? |
| 04:36.57 | [TK]D-Fender | halkun: Do yourself a favor : (*x.T|#x.T|x.T) |
| 04:38.54 | halkun | you see, I don't know what that means/ |
| 04:39.32 | halkun | (Embarrassingly, I'm getting FSX and FXO confused now) |
| 04:39.55 | [TK]D-Fender | hlLINE vs PSTN tabs in the GUI admn for it |
| 04:40.04 | [TK]D-Fender | halkun: LINE vs PSTN tabs in the GUI admn for it |
| 04:40.23 | [TK]D-Fender | halkun: LINE = FXS = Phone. PSTN = FXO = your phone line |
| 04:41.10 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 04:41.12 | halkun | FXO = the thingy that makes the dial tone |
| 04:41.27 | p3nguin | no |
| 04:42.00 | halkun | FXS = walljack |
| 04:42.20 | p3nguin | yes |
| 04:42.52 | halkun | FXO = jack at the end of the phone cord |
| 04:43.20 | [TK]D-Fender | ~fxsfxo |
| 04:43.20 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 04:43.38 | [TK]D-Fender | ~fxs |
| 04:43.39 | infobot | [fxs] foreign exchange station - type of port you need to connect an analog device (phone, fax machine) to a pbx. This is the type of port found in your wall jack. |
| 04:43.40 | [TK]D-Fender | ~fxo |
| 04:43.40 | infobot | [fxo] foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx. This is the type of port found on phone or fax machine devices. See also: http://www.digium.com/index.php?menu=fxsvfxo |
| 04:44.18 | halkun | you man need to update that last URL |
| 04:44.20 | halkun | it 404s :) |
| 04:45.27 | halkun | ~dialplan |
| 04:45.27 | infobot | methinks dialplan is the thing configured in extensions.conf |
| 04:45.44 | halkun | hmm |
| 04:46.51 | [TK]D-Fender | halkun: just use what I gave you for the LINE page dialpan |
| 04:47.27 | *** join/#asterisk sourcode (~code@ppp-58-11-75-89.revip2.asianet.co.th) |
| 04:47.34 | halkun | can you explain it to me before I pop it in? |
| 04:49.35 | [TK]D-Fender | halkun: the dialplan on the device determines when it thinks you're finished dialing your number before passing it on (in whole) to * |
| 04:50.02 | *** join/#asterisk Gopal (~Miranda@61.12.17.170) |
| 04:50.24 | [TK]D-Fender | halkun: You could set it so 1000-1999 are taken as "as soon as the 4th digit in the pattern matches, consider it complete and send immediately". |
| 04:50.35 | [TK]D-Fender | halkun: To speed up calling local phones, etc. |
| 04:51.34 | [TK]D-Fender | halkun: This requires you to program all the kinds of patterns you'll actually be coding in your ASTERISK dialplan anyway. So better off just telling it "take what I dial and just wait till I stop for 3s before considering that I'm done". |
| 04:51.39 | [TK]D-Fender | halkun: the STFU dialplan. |
| 04:52.50 | ChannelZ | the Shitty Tit Fuck Up dialplan! |
| 04:53.51 | Jumpie | ChannelZ is that even possible? |
| 04:53.59 | Jumpie | to shitty tit fuck |
| 04:54.05 | halkun | so putting (*x.T|#x.T|x.T) in the 3102 will allow me to set up the dialplan in * and all I'm doing here is telling the 3012 to take what it is given |
| 04:54.34 | ChannelZ | Jumpie: well if you were doing it to Nancy Pelosi.. that'd be pretty shitty |
| 04:54.41 | Jumpie | haha |
| 04:54.43 | Jumpie | ugh |
| 04:54.50 | Jumpie | just hearing that skanks name makes me wanna vomit |
| 04:55.35 | ChannelZ | See! |
| 04:56.00 | [TK]D-Fender | halkun: Yes |
| 04:56.15 | halkun | http://craphound.com/images/5jplr.jpg <---- This needs to be the official asterisk t-shirt (funnier if you have read "Breakfast of Champions" |
| 04:59.12 | halkun | Ok, reading ahead... This goes into the "Line 1" tab |
| 04:59.38 | halkun | because there is another dailplan that goes into the PSTN line |
| 05:00.00 | [TK]D-Fender | halkun: that one is very separate and different |
| 05:00.17 | halkun | Ok |
| 05:02.24 | ChannelZ | damnit what's the app that plays signal tones? I though it was PlayTones or something |
| 05:03.57 | halkun | Ok, so now I'm configuring PSTN now.... |
| 05:04.24 | halkun | proxy is the * server (192.168.0.1) |
| 05:05.25 | halkun | under dialplan 8 it says to put S0<:123@192.168.0.2> |
| 05:05.36 | halkun | oops, wrong I'm at .1 |
| 05:06.30 | halkun | so in this example, the incoming calls are coming on extention 123 |
| 05:06.57 | *** join/#asterisk `Sauron (~sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 05:10.07 | ChannelZ | hmm darn there is no indication sound for 'left your phone offhook' |
| 05:11.58 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 05:16.38 | halkun | hmm |
| 05:17.14 | halkun | Ok, I have a test config in my sip.conf that works with a usrename ans secret |
| 05:17.56 | halkun | oh, never mind... found it |
| 05:19.49 | ChannelZ | Hmm. I need a really cheesy version of Girl From Ipanema, like Musak version with xylophones and crap |
| 05:20.59 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 05:23.37 | carrar | http://www.youtube.com/watch?v=_ZmQr78Otv4&feature=fvw |
| 05:24.13 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 05:24.44 | [TK]D-Fender | checkout time. Later all |
| 05:24.50 | carrar | WAIT |
| 05:24.53 | carrar | Don't GO |
| 05:25.18 | carrar | I've giving away $1,000,000 buy only to you in 5 seconds if you are here |
| 05:25.22 | carrar | but |
| 05:25.31 | carrar | doh, missed out |
| 05:26.05 | halkun | hmmm |
| 05:26.27 | halkun | I put the new setting in there and now xlite is giving me a 403 - forbidden error |
| 05:27.21 | halkun | I guess it's finding the account |
| 05:27.36 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 05:28.25 | ChannelZ | carrar: that's pretty good, but instrumental. I think I found a sufficiently shitty one on amazon |
| 05:28.26 | Gopal | halkun: if it is 403 may be some authentication prob with username and password or auth name |
| 05:28.53 | ChannelZ | It's actually not quite the song I was thinking of though, I guess I don't know the name of the one I'm thinking |
| 05:29.29 | halkun | The sip.conf I'm using is from here ---> http://forum.voxilla.com/cisco-linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html |
| 05:29.52 | halkun | I'm logging in with with the username of "line1" and my passowrd |
| 05:30.07 | halkun | That's the only lie I see with a username |
| 05:30.17 | halkun | line I mean |
| 05:31.21 | halkun | under the [xlite] section I don't see a username up there |
| 05:31.49 | halkun | only under the [line1] section |
| 05:33.21 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 05:34.14 | Gopal | username will is the context name [xlite] is the uername |
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| 05:37.46 | halkun | ERROR[9913]: chan_sip.c:8671 register_verify: Peer 'line1' is trying to register, but not configured as host=dynamic |
| 05:38.26 | halkun | you know, you can get a much of really awesome errors when you actually look at what * is doing... |
| 05:39.05 | halkun | aha :) |
| 05:39.46 | halkun | helps when you log in with the right username... thanks |
| 05:40.40 | Gopal | halkun: you have to give host=dynamic in your context [xlite] |
| 05:41.15 | halkun | that's there.... |
| 05:41.49 | halkun | it's saying I need host=dynamic for "line1" |
| 05:42.43 | halkun | but that's not right. I have "host=192.168.0.13" (The address of the 3102) |
| 05:45.49 | Gopal | halkun: try giving host = dynamic |
| 05:47.01 | Gopal | halkun: you have to give your sip provider IP as host |
| 05:47.04 | p3nguin | If your device is trying to register, you can't use host=ipaddress, you have to use host=dynamic. |
| 05:47.37 | halkun | http://scsys.co.uk:8002/43469 |
| 05:47.59 | halkun | Ok, I'll make id dynamic ans see |
| 05:48.29 | halkun | the URL above is the error and the sip.conf I coypasta-ed |
| 05:50.53 | halkun | well, it's dynamic, and I'm not getting and more errors in CLI |
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| 05:54.52 | halkun | <PROTECTED> |
| 05:55.18 | halkun | looks like I'm off to configure extensions.conf |
| 05:57.02 | halkun | hhhm |
| 05:57.12 | halkun | the example here uses the 10-digit numbers again |
| 05:57.36 | halkun | I don't think that's going to work as I changed the dialplan in the 3102 |
| 05:58.04 | p3nguin | The dialplan in the 3102 just tells it where to send what calls. |
| 05:58.15 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-oitdrbvjmjeijkmk) |
| 05:58.15 | p3nguin | Extensions.conf is where you have to match dialed numbers for processing. |
| 05:58.26 | p3nguin | extensions.conf, rather. |
| 05:58.40 | carrar | ~book |
| 05:58.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:59.52 | Gopal | halkun: yes you need to have a dialplan to route the outgoing calls |
| 06:01.09 | halkun | in that case I'm going to bed for the night. |
| 06:01.23 | halkun | I'll deal with dialplans as another project. |
| 06:01.33 | carrar | after you read the book |
| 06:01.49 | p3nguin | Don't confuse asterisk's dialplan with the dialplan on the device. |
| 06:02.05 | halkun | I'm just happy that xlite is logging in and (theoretically) the 3102 is connected |
| 06:02.10 | carrar | dialplan or digit map? |
| 06:02.33 | p3nguin | Polycom calls it s digit map, but Cisco/Linksys call it dialplan. |
| 06:03.05 | halkun | thanks for the help guys :) |
| 06:10.19 | ChannelZ | carrar: call SIP/spark.idolum.com |
| 06:10.47 | p3nguin | When I dial out via future nine, the counter on the phone never starts, indicating that the call is still in progress instead of ever being answered (even though the far end has answered). Is there anything I can do on my end to make the line go up? |
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| 06:13.52 | p3nguin | I guess there was... switching to ideasip instead of future nine solved it. :/ |
| 06:14.17 | p3nguin | Dialed the same number through the other provider and it showed Connected and the counter started. |
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| 06:16.16 | ChannelZ | sounds like their system isn't sending back progress |
| 06:16.49 | p3nguin | Yeah, and that prevents me from being able to pass DTMF to the other end of my calls. |
| 06:17.45 | p3nguin | But, since I wasn't paying them for any services, they probably don't really care that their system is broken. |
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| 06:48.11 | b14ck | Is there a way in dialplan to convert a string variable to an int? |
| 06:48.22 | b14ck | like INT(${my_string_var}) or something? |
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| 06:52.57 | ChannelZ | variables are sort of typeless. They're all kind of strings. Certain operators will treat them like numbers if they contain only numbers |
| 06:53.05 | b14ck | gotcha |
| 06:53.06 | b14ck | thanks |
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| 07:33.47 | Gothicmaster86 | good morning |
| 07:34.20 | xheliox | good morning |
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| 07:35.58 | Gothicmaster86 | has anyone here experience with AsteriskNow and ISDN? |
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| 07:40.29 | zamba | Gothicmaster86: ask your question |
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| 07:43.38 | Gothicmaster86 | zamba: i search a Version of AsteriskNow what works with ISDN-Cards. Without GUI. |
| 07:44.40 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:45.30 | kaldemar | the whole point of AsteriskNow is to be a complete distro that gives you a subset of features and is configured via a GUI. maybe AsteriskNow is not what you're looking for? |
| 07:46.26 | kaldemar | choose your favorite linux distro (if you have one) and install asterisk on it. what kind of ISDN card do you mean? |
| 07:48.57 | Gothicmaster86 | kaldemar: Maybe i've installed Asterisk 1.4.29.1. on a Debian Etch. The Problem is, i didn't found any Driver for an ISDN HFC-S PCI-Card (1 Port) what works without problems! |
| 07:50.54 | Gothicmaster86 | kaldemar: Hardware-bridging is terrible (my Distro kills hisself at a Call) and Software-bridging or echocanceling is with the actual drivers impossible |
| 07:52.47 | kaldemar | you tried wcb4xxp on the latest dahdi? |
| 07:56.26 | kaldemar | there might be lack fo support for BRI in 1.4 chan_dahdi anyway. i'd try 1.6 and if it doesn't work for you, buy a card that is known to work. i've seen more issues than success with generic HFC-S based cards. |
| 07:57.09 | Gothicmaster86 | kaldemar: i've tried, but it doesn't work. DAHDI doesn't acknowledges my cards. In the moment it works halfway with mISDN-1.1.7.2 |
| 07:58.31 | Gothicmaster86 | kaldemar: have you experience with the new vISDN? or mISDN v2? |
| 07:59.51 | kaldemar | neither. it's been years since i've used any kind of BRI. i used to use bristuff with HFC-S cards on asterisk 1.2. |
| 08:01.03 | Gothicmaster86 | kaldemar: i heared bristuff didn't work with asterisk 1.4.* and newer... |
| 08:01.59 | kaldemar | it does, 0.4.0 versions are meant for 1.4: http://www.junghanns.net/downloads |
| 08:03.33 | kaldemar | 0.4.0-RC3k is meant to build against asterisk 1.4.29.1. 0.4.0-RC4 is the newest, updated to asterisk 1.4.31 |
| 08:04.01 | Gothicmaster86 | kaldemar: and what you mean is better? Capi or Bristuff? |
| 08:05.16 | kaldemar | i've never tried capi. |
| 08:06.21 | Gothicmaster86 | hmm then i would check out bristuff. thanks ;) |
| 08:08.20 | Gothicmaster86 | i must anyway make new my Distro. Then i could try the new driver. (i think i've killed my Kernel with a bad kernel-update) |
| 08:09.17 | Gothicmaster86 | exit |
| 08:11.04 | xheliox | hmms |
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| 09:41.33 | xheliox | *tumbleweeds* |
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| 09:59.11 | AndroFlex | hello Can anybody give me some fast advice? I'm trying to develope a CTI with Flex, but I don't want to use Java + Flex or .Net + Flex, I want to use flex alone? any ideas? |
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| 10:06.21 | BrokenNoze | Hi all. I have a very serious problem with a production on asterisk 1.4.20.1, after about 2 weeks of successful operation my IAX channel reports "maxium retries succeeded" between 2 systems and then hangs. Asterisk will not restart, stop and I am unable to even use kill to kill the pid. The only solution is a full reboot. Upon the reboot the system comes back up without any problem and continues working again |
| 10:06.58 | BrokenNoze | Is there any known issue within the dialplans that could cause this? is there any work around? I would post on forums but i am desperate for a soultion! |
| 10:07.39 | BrokenNoze | I should say that it had been working for 3 years until very recently and the only change i can think of is one within the dialplan itself.. |
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| 11:40.11 | wonderworld | hi. can i run shell commands from the asterisk CLI? |
| 11:40.39 | kaldemar | wonderworld: !<command> |
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| 11:43.05 | wonderworld | tnx |
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| 11:51.52 | Gopal | hi steve123 |
| 11:51.57 | Steve123 | hi |
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| 12:25.56 | chripher | is there any soft phone for linux that support auto provisioning |
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| 12:28.30 | [TK]D-Fender | chripher: zoiper |
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| 13:13.40 | hluesea | hello channel |
| 13:14.29 | Baylink-lastday | hears the channel say "Hi", and wonders why |
| 13:15.37 | hluesea | is anybody know that : i have 16 channels and 4 channels 2 different trunks and i want to outcalls using that but they have a channel limitation. How can i limitation each channel and whole outbound calls ? |
| 13:16.46 | [TK]D-Fender | hluesea: GROUP() |
| 13:17.22 | hluesea | but that is not dahdi channel, each one is the different brands sip gw mean sip trunks |
| 13:17.31 | [TK]D-Fender | hluesea: GROUP() <- DIALPLAN FUNCTION |
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| 13:19.02 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:19.23 | hluesea | thanks [TK]D-Fender i am looking on |
| 13:22.01 | BarthezZ | hmm, is it normal for a sip provider to deliver 1 account per possible active account? I'm used to get one (or one account per number) which can hold multiple calls |
| 13:22.52 | [TK]D-Fender | BarthezZ: One account per account? Unheard of... |
| 13:23.06 | BarthezZ | uhmm excuse me, one account per possible outgoing call |
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| 13:23.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:23.30 | *** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl) |
| 13:23.36 | pabelanger | BarthezZ: Anything is possible, their network, their rules. |
| 13:24.38 | sigius | Q: What is a popular asterisk load testing tool ? |
| 13:25.00 | BarthezZ | hmm, than I have to think of a way to check for which account is idle (no call on it) and use that one for an outgoing call |
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| 13:27.19 | [TK]D-Fender | BarthezZ: "core show application ChanIsAvail" |
| 13:27.22 | [TK]D-Fender | sigius: SIPP |
| 13:28.08 | BarthezZ | yeah [TK]D-Fender, just my outgoing dialplan is going to be a bit messy in that case |
| 13:28.30 | [TK]D-Fender | BarthezZ: Pick another provider then |
| 13:28.33 | BarthezZ | ah well, sometimes you need to adapt to the customers wishes :( just wish they would have consulted me before selecting a sip provider |
| 13:29.46 | Gido-E | BarthezZ do they want technology or a sip account? |
| 13:30.11 | BarthezZ | technology? what do you mean? |
| 13:30.37 | Gido-E | BarthezZ you can always say, buy this or do that. |
| 13:31.05 | BarthezZ | yes... but before consulting me they already registered with a sip provider, got a block of numbers, etc... and signed a 2 years contract |
| 13:31.32 | BarthezZ | the guy who hired me does know something, but misses some important parts of knowledge :p |
| 13:33.00 | [TK]D-Fender | BarthezZ: Time for some messy dialplan. |
| 13:33.09 | Gido-E | yep :-) |
| 13:33.13 | BarthezZ | yup :( |
| 13:33.34 | BarthezZ | ah well, first get that crappy ISDN hardware running for the lifeline |
| 13:33.53 | BarthezZ | openvox is a pain in the ass |
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| 13:42.53 | ruben23 | hi guys |
| 13:44.26 | Baylink-lastday | Morning (he said, falling manfully on the grenade) |
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| 13:47.03 | *** join/#asterisk Wolfeyes (~Wolfeyes@41.124.150.186) |
| 13:47.41 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 13:48.23 | Wolfeyes | Good day everyone. |
| 13:48.41 | Wolfeyes | Anyone here from South Africa? |
| 13:50.00 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 13:50.26 | *** join/#asterisk kazaa_lite (~eddie@212.183.140.16) |
| 13:57.10 | Wolfeyes | Anyone willing to help a beginner in here? |
| 13:58.23 | Baylink-lastday | Sure. |
| 13:58.25 | *** join/#asterisk werty1st (~werty1st@p5B20B27E.dip.t-dialin.net) |
| 13:58.48 | Baylink-lastday | said, falling manfully on another grenade. :-) |
| 13:58.52 | *** join/#asterisk af_ (~getsmart@88-149-241-148.dynamic.ngi.it) |
| 13:59.41 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 14:01.31 | werty1st | hello can somebody help with an asterisk sending a wrong/unknown ip address |
| 14:01.50 | werty1st | i dont know how asterisk detects its wan IP |
| 14:02.46 | werty1st | on debug i can read this: We're at 79.212.110.117 port 10036 |
| 14:03.10 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:03.18 | werty1st | this is definitely not my wan address |
| 14:03.44 | Gido-E | werty1st dont use nat |
| 14:04.35 | werty1st | i have pfsense with siproxd |
| 14:04.43 | werty1st | i disable nat now |
| 14:05.38 | *** join/#asterisk ManxPower (~manxpower@61.sub-75-254-195.myvzw.com) |
| 14:06.58 | werty1st | shouldi disable STUN too? |
| 14:10.13 | [TK]D-Fender | ~sipnat |
| 14:10.13 | 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 |
| 14:10.16 | [TK]D-Fender | werty1st: ^^^^ |
| 14:12.06 | werty1st | i have SoftPhone -> wanIP -> pfsense -> asterisk |
| 14:12.24 | werty1st | pfsense has siproxd and NAT configured |
| 14:12.29 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:12.55 | werty1st | port 5060/tcp and 10000-10100/udp for rtp |
| 14:13.28 | werty1st | if i disable nat on asterisk i cant register my softphone any lnger |
| 14:15.54 | ManxPower | Yay wireless! I get an external antenna for my EVDO card and I lose 3 DB of signal when I plug it in. Maybe they send me an anti-antenna |
| 14:16.15 | ManxPower | werdan7, you realize SIP generally uses UDP, right? |
| 14:16.41 | Nivex | ManxPower: did it look like this? http://en.wikipedia.org/wiki/File:Cantenna.JPG |
| 14:17.00 | ManxPower | Nivex, no |
| 14:17.26 | [TK]D-Fender | [10:12]<werty1st>pfsense has siproxd and NAT configured <--- your firewall should GTFOP of *'s way |
| 14:17.27 | ManxPower | Can you do cantennas on 1900/800 Mhz? |
| 14:18.35 | coppice | ManxPower: sure. don't you remember those early cellphone users with big parabolic dishes on their heads? |
| 14:18.49 | ManxPower | coppice, must have been before my time. |
| 14:19.02 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 14:19.11 | Slugs_ | morning |
| 14:19.18 | ManxPower | I guess I should check to see if I can return it to VZ |
| 14:20.17 | *** join/#asterisk e3eli3h (~Chris@95.211.84.170) |
| 14:20.21 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:21.36 | e3eli3h | Hi all. Anyone in here able and willing to help me troubleshoot a g729 codec installation issue? |
| 14:22.08 | werty1st | i have also setup the coresponding FW rules for incomming sip and rtp |
| 14:22.58 | ManxPower | werty1st, NO! You do NOT need to do that when you are trying to make everything work. FW should be the LAST thing you set up |
| 14:23.19 | ManxPower | e3eli3h, contact Digium for support for that commercial product |
| 14:24.10 | werty1st | i have no problem with internal clients |
| 14:24.36 | ManxPower | Just let us know when you are ready to follow advice. |
| 14:24.55 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:25.00 | werty1st | could u explain me what to do first? |
| 14:25.05 | ManxPower | service iptables stop |
| 14:25.22 | e3eli3h | @ManxPower, tried that and I am getting no response, turning here as a last resort before I jump ship. |
| 14:25.26 | ManxPower | or whatever you do to turn off your firewall on your asterisk box |
| 14:25.51 | ManxPower | e3eli3h, not much anyone here can do for you |
| 14:25.53 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:26.07 | e3eli3h | I see. |
| 14:27.10 | ManxPower | e3eli3h, You copy the module to the asterisk modules directory, run the register application, enter your registration info and restart Asterisk, This isn't rock science and is documented |
| 14:28.44 | e3eli3h | Done all that and the asthostid output is matching the host id in the generated .lic file. Hold on. You haven't even asked me what my problem is, have you? |
| 14:29.13 | ManxPower | e3eli3h, I doubt it will matter. Digium is the only one that can support that binary non-open source codec. |
| 14:30.00 | e3eli3h | Again, I see. Thanks for all your efforts. |
| 14:31.58 | [sr] | i'm sad |
| 14:32.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:32.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:33.28 | ManxPower | Yay leifmadsen! The person that single handedly removed the requirement of TeX to read the documentation! Yay! |
| 14:34.48 | [TK]D-Fender | [10:28]<e3eli3h>Done all that and the asthostid output is matching the host id in the generated .lic file. Hold on. You haven't even asked me what my problem is, have you? <--- You're supposed to TELL us what your problem is, not make us ask for it. |
| 14:36.01 | *** join/#asterisk werty1st (~werty1st@p5B20FE46.dip.t-dialin.net) |
| 14:36.26 | leifmadsen | ManxPower: yay! :) |
| 14:36.44 | leifmadsen | ManxPower: that reminds me -- in the latest releases did it auto-generate the text version? |
| 14:36.49 | leifmadsen | I forgot to look for that |
| 14:37.02 | ManxPower | leifmadsen, No idea. 8-) I'm stuck on 1.4.x for a while more. |
| 14:37.05 | leifmadsen | it should have included asterisk.txt or something like that in the doc/ or doc/tex/ directory |
| 14:37.15 | leifmadsen | is there tex in 1.4? |
| 14:37.19 | ManxPower | But thousands of current and future generations thank you. |
| 14:37.48 | leifmadsen | I'm gonna go grab coffee and then I'll check when I get back unless you take a couple minutes to grab the latest RC and check for me :) |
| 14:37.57 | e3eli3h | Anyone in here able and willing to help me troubleshoot a g729 codec installation issue? <--- is a question. If you are both willing and able, I'll ask you the question and paste my logs in a private chat. I will not waste channel real estate. |
| 14:38.10 | ManxPower | leifmadsen, I didn't pester you do to the text stuff for *me*. I already know Asterisk pretty well. I did it for all the poor n00bs out there. |
| 14:38.18 | ManxPower | ~ask |
| 14:38.18 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:38.40 | leifmadsen | ManxPower: I'm not sure where I said you did :) |
| 14:39.05 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:39.45 | ManxPower | leifmadsen, the thing is the txt versions were REMOVED in 1.6.mumble. |
| 14:41.33 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 14:41.56 | madduck | [TK]D-Fender: the provider (netvoip.ch) immedietaly responded and told me to set inband and relaxdtmf=yes, and now it works. just fyi |
| 14:43.11 | [TK]D-Fender | madduck: Ah... CRAPPY inband DTMF. If you have to do that... try to replace them ASAP |
| 14:43.33 | ManxPower | madduck, Did they also tell you to use alaw (or ulaw)? |
| 14:43.57 | madduck | [TK]D-Fender: why? now it works. |
| 14:44.06 | madduck | ManxPower: no. i told them i already tried alaw/ulaw |
| 14:44.17 | ManxPower | madduck, because any carrier that requires inband is not a provider you want. |
| 14:44.39 | ManxPower | madduck, um, inband DTMF doesn't work unless you are using ualw or alaw |
| 14:44.49 | [TK]D-Fender | madduck: And any one that not only requires it, but also isn't stable enough that you require relaxdtmf is horrible |
| 14:45.26 | ManxPower | [TK]D-Fender, it's almost like a provider has a "kick me" sign on their back,. |
| 14:45.28 | madduck | ManxPower: fair enough. inband means that the tones are transmitted, right? |
| 14:45.42 | ManxPower | madduck, it means the DTMF is transmitted as audio. |
| 14:45.59 | ManxPower | So if your audio is always perfect, I guess inband will work. |
| 14:46.01 | madduck | right |
| 14:46.16 | ManxPower | the relaxdtmf thing is just bizarre. |
| 14:47.57 | madduck | i asked them about it. |
| 14:49.05 | ManxPower | in my experience relaxdtmf causes issues with double DTMF digits being seen as a single DTMF digits. i.e. 54467 might be seen as 5467 |
| 14:49.26 | coppice | replace the DTMF decoder with a decent one |
| 14:49.46 | ManxPower | coppice, that and removing relaxdtmf |
| 14:49.46 | madduck | netvoip.ch are the only ones that give me a zurich phone number for free. :( |
| 14:50.03 | ManxPower | madduck, you. get. what. you. pay. for. |
| 14:50.15 | madduck | ;) |
| 14:50.26 | coppice | relaxdtmf seems to be necessary to cope with some systems, but its pretty harmless with a good decoder |
| 14:50.49 | ManxPower | hugs RFC2833 |
| 14:55.22 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 14:56.33 | *** join/#asterisk Tim_Toady (~moi@193.92.247.185.dsl.dyn.forthnet.gr) |
| 14:58.35 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:59.00 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:01.31 | madduck | now all i really need to figure out is how to turn a peer-to-peer phone call into a conference ;) |
| 15:01.58 | madduck | i.e. how to call someone else and merge, or even let people join. |
| 15:02.03 | werty1st | thanks for trying no luck today :-( |
| 15:02.06 | *** part/#asterisk werty1st (~werty1st@p5B20FE46.dip.t-dialin.net) |
| 15:07.45 | leifmadsen | ManxPower: no removed -- converted I'm sure |
| 15:08.32 | lepine | just deploy his first asterisk setup in production. |
| 15:08.44 | Wolfeyes | Anyone from South africa in here? |
| 15:13.04 | *** join/#asterisk MiserySoft (~LND@92.41.77.98.sub.mbb.three.co.uk) |
| 15:20.24 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:22.45 | *** join/#asterisk MAbbas (~abbas@203.215.177.194) |
| 15:22.58 | Qwell | <coppice> replace the DTMF decoder with a decent one |
| 15:23.01 | Qwell | coppice: patches welcome :p |
| 15:23.48 | coppice | Qwell: you shouldn't have broken the one you have |
| 15:25.09 | russellb | yeah, thanks a lot Qwell |
| 15:25.12 | russellb | :-p |
| 15:30.49 | leifmadsen | Qwell: sheesh! |
| 15:31.07 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:31.25 | wcselby | o/ |
| 15:32.04 | *** part/#asterisk MiserySoft (~LND@92.41.77.98.sub.mbb.three.co.uk) |
| 15:39.41 | *** part/#asterisk ruben23 (~magisx@202.137.112.11) |
| 15:43.01 | ManxPower | isn't there a DTMF detector in spanDSP? |
| 15:45.29 | *** join/#asterisk kazaa_lite (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 15:48.08 | *** join/#asterisk korcan (~kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net) |
| 15:49.29 | coppice | :-) |
| 15:51.45 | *** join/#asterisk jonny330 (~Jon@12.222.63.34) |
| 15:51.47 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:52.33 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:52.57 | coppice | does anyone know how fast Howlertech's new G.722 codec runs? |
| 15:54.01 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 15:54.15 | *** join/#asterisk jonny330 (~Jon@12.222.63.34) |
| 15:54.27 | *** part/#asterisk Wolfeyes (~Wolfeyes@41.124.150.186) |
| 15:54.30 | jonny330 | i am having problems dialing out on a dahdi line |
| 15:55.34 | *** join/#asterisk analogkid (~xchat@dialin-145-254-158-213.pools.arcor-ip.net) |
| 15:56.16 | analogkid | hi, does anyone know if there are capi drivers for gentoo ppc? |
| 15:56.22 | analogkid | so linux |
| 15:57.58 | jonny330 | i have a tdm800p and it takes 2-4 tries to call outbound, inbound works perfect |
| 15:59.43 | Qwell | jonny330: show us logs |
| 16:02.13 | TSM | is it correct that if someone hangs up from within the voicemail app it terminates the call script straight away instead of allowing the call script to continue to do other things? |
| 16:02.39 | *** join/#asterisk MmixX (mixed@unaffiliated/mmixx) |
| 16:04.53 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.239.229.dsl.dyn.forthnet.gr) |
| 16:05.34 | [TK]D-Fender | TSM: "Call script"? Pardon? |
| 16:05.38 | wcselby | TSM - yes, when a call is hung up, it's hung up |
| 16:06.23 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:06.24 | TSM | well thats poo means that no one can script anything after VM, mabey there should be an option to continue with a status of VM for if a message was left or not |
| 16:06.38 | wcselby | TSM - there's always the h extension..... |
| 16:07.13 | wcselby | plus, there's an option in voicemail.conf that will execute a user defined command once a voicemail has been left |
| 16:07.40 | wcselby | TSM - look at externnotify option in the sample voicemail.conf file |
| 16:08.36 | TSM | wcselby: yup but it would be nice if that was within the dialplan |
| 16:08.55 | wcselby | TSM - ...... try the 'h' extension in the dialplan then |
| 16:09.08 | wcselby | TSM - since we have no idea what you're trying to do (you're pretty vague), it's hard to help |
| 16:09.45 | TSM | ahh, i have a script that notifies users of missed calls, prob is that it wont if they go into Voicemail and then just hang up |
| 16:10.41 | wcselby | TSM - call the script before it goes to voicemail().... |
| 16:10.41 | Qwell | if it hits voicemail, it's already missed. so why not notify before? |
| 16:10.41 | p3nguin | Eh, the missed calls on the phone and the message waiting indicator isn't enough? |
| 16:11.06 | TSM | @Qwell: because i dont want a message that a call was missed if they do leave a message as then its two messages, one that its missed and another that they have left a message, but i do see your point |
| 16:11.34 | wcselby | TSM - then we're back to the 'h' extension |
| 16:12.34 | p3nguin | If there's voicemail, that's a pretty good indication that there's voicemail. :/ |
| 16:12.44 | p3nguin | And if the phone shows a missed call, that's a pretty good indication that the call was missed. |
| 16:12.49 | wcselby | p3nguin - he's saying if they hangup before leav ing a voicemail |
| 16:12.57 | *** join/#asterisk gelo (~gelo@mx01.quobis.com) |
| 16:12.58 | cidu | so, quick question here, having trouble with CALLERID(ANI) over PRI, cant seem to get asterisk to recieve, or transmit CALLERIS(ANI) properly, its always populated with the CALLERID(NUM) Value and not the charge number data, is this a known bug or what? |
| 16:13.00 | p3nguin | "missed calls" |
| 16:13.09 | cidu | err, CallerID(ANI) even |
| 16:16.15 | jhirley | o/ Hello Peeps |
| 16:24.11 | *** part/#asterisk gelo (~gelo@mx01.quobis.com) |
| 16:25.04 | p3nguin | I've encountered a NAT issue where the remote device's RTP packets have the PRIVATE IP address on them instead of the PUBLIC address. If anyone cares to help, here's the sip debug and rtp debug: http://pastebin.com/DbRj4K8V |
| 16:25.51 | p3nguin | Using a softphone rather than the SPA-3102 on that LAN works fine. The RTP packets have the public address on them when using zoiper. |
| 16:26.00 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 16:27.14 | p3nguin | Here's a comparison of RTP from zoiper and from the SPA-3102: http://pastebin.com/1w3T3PKt |
| 16:37.12 | *** join/#asterisk bsaxon_ (~bsaxon@12.107.149.61) |
| 16:37.44 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
| 16:38.35 | *** join/#asterisk endemic (~endemic@lynx.ipv6.onvox.net) |
| 16:42.20 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 16:42.23 | [sr] | howdy people |
| 16:42.28 | [sr] | i need to ask a dumb question :P |
| 16:42.49 | [sr] | i have a 2 port ISDN card: 00:10.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) |
| 16:43.15 | [sr] | my question is, with this card, i can only connect it to my telephone network NTBA's |
| 16:43.22 | [sr] | and then route all in asterisk |
| 16:43.35 | [sr] | am i correct? |
| 16:43.55 | [sr] | if yes, i defined a incoming route for any DID/CID |
| 16:44.14 | WIMPy | No you can also connect Telephones to it. |
| 16:44.16 | [sr] | to a SIP extension, but nothing happened and the number of the line i called, doesn't ring |
| 16:44.27 | [sr] | hum... Wimme |
| 16:44.30 | [sr] | WIMPy: |
| 16:44.41 | *** part/#asterisk Baylink-lastday (~jra@cerberus.vicimarketing.com) |
| 16:44.46 | [sr] | i tried to connect a isdn extension from the central |
| 16:44.57 | WIMPy | What driver are you using? |
| 16:45.11 | [sr] | but the port didn't came green, as it did when i connect it to the PSTN NTBA |
| 16:45.45 | WIMPy | You need a crossover cable to connect telephones. |
| 16:46.04 | [sr] | ahhhhh |
| 16:46.23 | [sr] | for the NTBA connection i was using a direct RJ45 |
| 16:46.30 | [sr] | the driver is: wct4xxp |
| 16:46.41 | WIMPy | That NOT the same as an ethernet crossover, however. |
| 16:46.46 | [sr] | is there a schema of the crossover cable? |
| 16:47.33 | WIMPy | Hmm, I didn't succceed in NT mode with the dahdi driver. |
| 16:47.55 | [sr] | hum, which one then? |
| 16:47.57 | WIMPy | 2<>4 and 5<>6 |
| 16:48.18 | p3nguin | [sr]: First things first. There is no "SIP extension." There are extensions (found in extensions.conf) and there are SIP devices (found in sip.conf). |
| 16:48.54 | [sr] | WIMPy: is there a visual schema on the net? may be beter for me.. |
| 16:48.58 | p3nguin | [sr]: So start from the top. Do you have a peer entry in sip.conf for your device? If yes, do you have an extension to Dial() to that device in extensions.conf? |
| 16:49.17 | [sr] | p3nguin: defined it on the interface |
| 16:49.20 | [sr] | freepbx |
| 16:49.24 | Qwell | [sr]: http://www.google.com/images?hl=en&q=isdn%20crossover%20cable |
| 16:49.38 | WIMPy | [sr]: With older kernels you can use misdn/can_misdn or with more recent kernels misdn2/lcr/chan_lcr. |
| 16:49.39 | [sr] | Qwell: merci, gonna check |
| 16:50.11 | *** join/#asterisk ebroad (~EB@72.11.213.195) |
| 16:50.17 | p3nguin | FreePBX, eh? |
| 16:50.51 | [sr] | :$ |
| 16:50.52 | [sr] | yes |
| 16:51.05 | [sr] | but wait, i'm going to start with the cable |
| 16:51.19 | [sr] | and make it work with the isdn extension |
| 16:51.28 | p3nguin | I wish my issue was with a cable. It would make things so much easier. |
| 16:51.45 | p3nguin | Especially since no one knows much about overcoming NAT problems. |
| 16:51.57 | [sr] | for the isdn extension i need a crossover, and for the NTBA connection i can use the direct cable, correct? |
| 16:52.09 | WIMPy | correct |
| 16:52.14 | [sr] | perfect |
| 16:52.58 | [sr] | before i go, i have another machine with a 4x FXO card, and to make it work i have to change on dahdi to : from-zaptel |
| 16:53.24 | [sr] | i have to do the same herE? |
| 16:54.00 | *** join/#asterisk dabaR (~dbernar1@24.77.24.161) |
| 16:54.28 | [sr] | it has the default: from-pstn |
| 16:55.01 | dabaR | I am trying to send voicemails as attachments and it looks like after the separator I get a blank line before the headers in the source of the email. This of course results in the attachment being interpreted as text. Any hints on where I could fix that? |
| 16:55.33 | ManxPower | ~freepbx |
| 16:55.33 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 16:55.55 | *** join/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 17:07.24 | wcselby | asterisk 1.4.26 - is there any way to get the device state? i need to know if a phone is off the hook or available to be rung... |
| 17:08.00 | hardwire | stretches |
| 17:08.25 | ManxPower | wcselby, SIP or ZAP? |
| 17:08.30 | wcselby | SIP |
| 17:08.47 | Gido-E | chanaivalable ofzo? |
| 17:08.47 | Qwell | a SIP phone won't tell Asterisk that it's offhook |
| 17:08.49 | ManxPower | wcselby, the phone does not contact the pbx when it goes off hook so you'll never get that info on sip |
| 17:09.04 | [TK]D-Fender | wcselby: "core show application ChanIsAvail" |
| 17:09.06 | ManxPower | chanisavail can check if the phone is inuse . |
| 17:09.17 | ManxPower | there are DEVSTATE backports for 1.4.x, IIRC |
| 17:09.18 | *** join/#asterisk decaffeine (~decaffein@unison.bgp.cmk.ru) |
| 17:09.19 | wcselby | [TK]D-Fender - thanks, I'll look at that |
| 17:09.29 | ManxPower | wcselby, looks like you might be using FreePBX |
| 17:09.35 | decaffeine | Hi :) |
| 17:09.43 | Gido-E | freepbx is much better dan asterisk. |
| 17:09.55 | ManxPower | *** Gido-E added to Ignore List |
| 17:09.55 | wcselby | ManxPower - no, I'm not |
| 17:09.56 | Qwell | Gido-E: cake is much better than buildings |
| 17:10.10 | [sr] | hey Qwell, with the crossover cable, connected to the central extension to the isdn card, it should become green algo right? |
| 17:10.11 | Gido-E | :) |
| 17:10.34 | p3nguin | <Gido-E> freepbx is much better dan asterisk. <-- Do you have any idea what you're even talking about? |
| 17:10.41 | Gido-E | nobody gets the joke, are you are all on to much caffeine |
| 17:10.42 | ManxPower | p3nguin, he is trolling |
| 17:11.27 | p3nguin | Might as well say something like, "CentOS is so much better than Linux." |
| 17:11.33 | decaffeine | Can somebody answer some question about linking astersik accounts with existing accounts on other sip provider? |
| 17:11.33 | wcselby | ManxPower - curious, what about what I said made you think I was using FreePBX? |
| 17:11.33 | ManxPower | wcselby, as much as I HATE to admit it, you might try the asterisk rpms from FreePBX/TrixBox. They have DEVSTATE and some other things already patched. |
| 17:11.38 | [sr] | Qwell: it is blinking red like it dopesn't have nothing connected |
| 17:11.50 | ManxPower | decaffeine, the answer is "Yes, everyone does that" |
| 17:12.06 | decaffeine | Cool :) |
| 17:12.12 | ManxPower | wcselby, 1.4.26 is the EXACT Asterisk version in many of our FreePBX installs. |
| 17:12.28 | Gido-E | decaffeine that is one of the basic things you want normally. |
| 17:12.28 | wcselby | ManxPower - freepbx / trixbox is not an option for this client |
| 17:12.34 | [TK]D-Fender | ~devstate |
| 17:12.34 | infobot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/community/russell/asterisk-1.4/func_devstate-1.4 , or http://www.asterisk.org/node/48325 |
| 17:12.34 | wcselby | ManxPower - ahhhh, gotcha |
| 17:12.36 | [TK]D-Fender | ^^^^^^ |
| 17:12.45 | decaffeine | I'm new to astersik and PBX |
| 17:12.55 | decaffeine | so i have no skilz :) |
| 17:12.56 | [TK]D-Fender | For non-device specific stuff |
| 17:12.58 | ManxPower | wcselby, FreePBX is not an option for ANYONE. People just ignore that fact and use it anyway |
| 17:13.05 | wcselby | ChanIsAvail should be what I'm looking for |
| 17:13.06 | ebroad | wasn't devstate ported to 1.6? |
| 17:13.09 | ManxPower | decaffeine, then you should be reading the asterisk book |
| 17:13.13 | [TK]D-Fender | ebroad: Yes |
| 17:13.15 | ManxPower | ~book |
| 17:13.15 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:13.36 | [sr] | WIMPy: ideas? with the crossover cable the port doesn't became green as when i connect it to the NTBA |
| 17:13.41 | decaffeine | I've started it already |
| 17:13.41 | ebroad | thought so |
| 17:14.08 | decaffeine | The main idea i want to use asterisk+adhearsion to test telephony on other sip prov |
| 17:14.16 | decaffeine | i.e IVR |
| 17:14.30 | jonny330 | so i am having a lot of problems dialing out on a tdm800p the call does not go throguh on the first try they have to make 2-4 attemps calling the number |
| 17:14.36 | *** join/#asterisk tokozedg (~toko@94.240.227.209) |
| 17:14.50 | jonny330 | here is my dial string |
| 17:14.50 | WIMPy | [sr]: Maybe it's not configured properly? i.e. not in NT Mode? |
| 17:15.03 | [sr] | WIMPy: hum, how can i know that? |
| 17:15.09 | [sr] | know/configure |
| 17:15.09 | WIMPy | That was what didn't work for me when using dahdi. |
| 17:15.19 | jonny330 | Dial(DAHDI/r0/1${EXTEN},60,r) |
| 17:15.42 | [sr] | WIMPy: switchingtype ? |
| 17:16.14 | WIMPy | switchtype must match, yes. |
| 17:16.30 | [sr] | hum ok i have euroisdn |
| 17:16.36 | [sr] | that's for NTBA for sure i guess |
| 17:17.02 | [sr] | and the other option should be? |
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| 17:18.02 | WIMPy | tries to remember... I think it's 'signalling'. There is bri_cpe and bri_net or something. |
| 17:18.15 | AndyGraybeal | what phone has the best (loudest, quality) speakerphone available? |
| 17:18.34 | WIMPy | But the hardware also needs to know. That's what failed for me. |
| 17:18.43 | WIMPy | Anyway, I'm out for lunch. |
| 17:18.44 | AndyGraybeal | i want to be able to hear the phone in a busy kitchen |
| 17:19.06 | AndyGraybeal | on speakerphone |
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| 17:20.19 | decaffeine | Thanks for the link |
| 17:21.28 | jonny330 | anyone able to help me out with dahdi? |
| 17:23.21 | *** part/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 17:23.23 | decaffeine | got no clue what it is.... |
| 17:23.24 | decaffeine | %) |
| 17:23.32 | ChannelZ | Ask an actual question |
| 17:23.35 | decaffeine | It's in FreePBX |
| 17:23.59 | p3nguin | FreePBX still isn't supported here. Still. |
| 17:24.38 | decaffeine | Thanks for the book i will be later with questions :) |
| 17:24.56 | p3nguin | FreePBX won't be supported here when you come back, either. |
| 17:25.37 | jonny330 | so i am having a lot of problems dialing out on a tdm800p the call does not go throguh on the first try they have to make 2-4 attemps calling the number |
| 17:25.41 | jonny330 | Dial(DAHDI/r0/1${EXTEN},60,r) |
| 17:26.06 | *** join/#asterisk lupino3 (~andrea_@static-217-133-45-108.clienti.tiscali.it) |
| 17:26.17 | lupino3 | hello everybody |
| 17:26.47 | ChannelZ | jonny330: do you get 'the number couldn't be completed as dialed' or somthing on a failed call? |
| 17:27.02 | TSM | foudn the info i wanted to check if someone has actualy left a VM, ${VMSTATUS} returns FAILED if they dont leave a message and SUCCESS if they do |
| 17:27.39 | jonny330 | ChannelZ: no nothing happens it takes a long time for anything to happen so users are hangning up |
| 17:28.03 | lupino3 | can anybody help me with my Astribank? I randomly get USB errors "device not accepting address" and "device descriptor read/n" with errors -71 and -10 |
| 17:28.12 | lupino3 | in those cases the astribank is not available via USB |
| 17:28.21 | lupino3 | but sometimes it Just Works (tm) |
| 17:28.30 | lupino3 | I can't reproduce the issue consistently |
| 17:28.38 | lupino3 | i tried two astribanks, two cables and two MB's |
| 17:28.40 | ChannelZ | jonny330: well one possible problem is your telco isn't providing a dialtone quick enough when the hardware picks up the line. Try putting a 'w' in before the 1 of your dial... |
| 17:28.49 | ChannelZ | Dial(DAHDI/r0/w1${EXTEN},60,r) |
| 17:28.54 | lupino3 | does anybody have a suggestion? |
| 17:29.16 | dabaR | I am trying to send voicemails as attachments and it looks like after the separator I get a blank line before the headers in the source of the email. This of course results in the attachment being interpreted as text. Any hints on where I could fix that? |
| 17:29.18 | tzafrir_laptop | lupino3, "randomly" - also same combination of astribank and system? |
| 17:29.25 | jonny330 | w1? |
| 17:29.25 | lupino3 | yes |
| 17:29.36 | ChannelZ | w means 'wait a second' |
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| 17:29.42 | *** join/#asterisk d00gster (~dt@94.98.9.227) |
| 17:29.44 | jonny330 | ahh cool |
| 17:29.45 | lupino3 | tzafrir_laptop, the same hard disk on two different MB's |
| 17:29.49 | ChannelZ | the 1 was yours, it was already there in the string you procided |
| 17:29.51 | ChannelZ | provided |
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| 17:31.00 | lupino3 | tzafrir_laptop, when it is working, it survives the reboot of the machine, the reboot of the astribank and the usb unplug/replug cycle |
| 17:31.12 | lupino3 | tzafrir_laptop, but when I get the USB errors... no way! |
| 17:31.29 | jonny330 | yeah thats seems to be making calls more reliable |
| 17:31.42 | jonny330 | but they are just taking a little longer to make it out |
| 17:31.46 | lupino3 | tzafrir_laptop, I tried different USB ports (with same results) |
| 17:32.00 | lupino3 | tzafrir_laptop, and on each port I tried an USB key (it works) |
| 17:32.15 | lupino3 | tzafrir_laptop, suggestions? while you think I just go banging my head on the nearest wall :( |
| 17:32.17 | *** join/#asterisk decaffeine (~decaffein@188.227.5.253) |
| 17:32.22 | jonny330 | ChannelZ: is there anytying else you can recommend? |
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| 17:33.01 | jonny330 | i just made a couple of test calls and 2 of them were good now these old ones are back to normal |
| 17:33.11 | minaguib | Hey folks. I've been reading a bit about VOIP security, but all of it relates to SIP. Do SIP headers map 1-1 with IAX2 headers ? |
| 17:33.22 | ChannelZ | well if the reason your calls are failing are because of that, then you can either complain to the telco and see if they will do anything, or leave the 'w' in and wait an extra couple of seconds |
| 17:34.10 | ChannelZ | You might be experiencing a different problem all together, not sure. Not enough information. |
| 17:34.31 | ChannelZ | minaguib: no IAX is totally different |
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| 17:35.26 | halkun | ok, so i've decided to start from scratch and go through the asterisk book |
| 17:35.30 | minaguib | ChannelZ: I understand that, but does that mean that an asterisk server that forwards a SIP call to another asterisk server using IAX will strip info ? |
| 17:35.42 | [TK]D-Fender | minaguib: SIP is to IAX2 as MS Word .doc is to OOo .odf. Same kind of goal, completely different contents |
| 17:35.52 | halkun | but it assumes I have something called a "Zatpel" which I don't have |
| 17:35.58 | halkun | so most of the examples are useless |
| 17:36.07 | p3nguin | minaguib: Define "forward." |
| 17:36.10 | [TK]D-Fender | minaguib: There is no "strip". There is no relationship for the information contained in one, to the other. |
| 17:36.33 | [TK]D-Fender | minaguib: And there is no "forward". With * these are 23 absolutely different calls |
| 17:36.37 | halkun | it's all "let's set up asterisk! First load the zaptel module.. |
| 17:36.42 | jonny330 | Channelz: well i am doing test calls by using dial phone#@internal from the cli and the calls are not always making it to me |
| 17:36.54 | halkun | so I'm pretty much stuck on step 2 |
| 17:36.57 | Qwell | halkun: s/zaptel/dahdi/ |
| 17:36.59 | [TK]D-Fender | minaguib: The only thing they may have in common is taht the CallerID from call A is used as the outbound for call B |
| 17:37.04 | p3nguin | halkun: replace... what qwell said. |
| 17:37.12 | halkun | I don't have one of those eather |
| 17:37.12 | AndyGraybeal | has anyone used the MCD100-M USB Speakerphone from plantronics? http://www.plantronics.com/north_america/en_US/products/computer/unified-communications-headsets/mcd100-m |
| 17:37.24 | minaguib | Ok for example, SIP has well-defined headers such as from/to, but many others in various RFCs (for example caller id presentation, ANI...). |
| 17:37.45 | halkun | I don't have any telephony equipment other then the 3102 |
| 17:38.11 | p3nguin | You'll still want dahdi at least for its dummy module. |
| 17:38.23 | [TK]D-Fender | minaguib: IAX2 doesn't have "headers". IAX2 is not a routed protocol like SIP is. There are no proxies. There are no special headers because IAX is a CLOSED protocol that isn't "extended" the way SIP gets extended |
| 17:39.27 | minaguib | [TK]D-Fender: So that means a SIP call will lose all its SIP headers except for the ones that map cleanly 1->1 with the IAX protocol ? |
| 17:39.50 | Qwell | minaguib: no, it means that they are 2 *separate* channels. |
| 17:39.51 | [TK]D-Fender | minaguib: No.. a SiP call coming in to * is a SIP call coming in to *. It doesn't LSOE anything |
| 17:40.02 | Qwell | Asterisk is not a proxy. |
| 17:40.15 | [TK]D-Fender | minaguib: the fact that * acts on that incoming call and decides to call OUT with IAX2 has absolutely no impact on the originating call |
| 17:40.27 | [TK]D-Fender | minAs qwell said... Asterisk is NOT a proxy |
| 17:40.44 | minaguib | Hmm. Perhap I'm confusing things then. |
| 17:41.01 | [TK]D-Fender | minaguib: There is not route. There are no headers. There is no proxy. THERE IS NO SPOON |
| 17:41.34 | p3nguin | minaguib: Think of it like taking a ride on a bus to the train station, where you get off the bus and get on a train. Your bus ticket is not good to ride on the train. |
| 17:41.52 | Qwell | p3nguin: but the other way usually does work |
| 17:41.55 | [TK]D-Fender | minaguib: A SIP call you pump out IAX2 is no different that acoustic coupling a CB radio to a cell phone. Yes audio passes, but each end has jack shit to do with the other |
| 17:41.59 | Qwell | for example, the bus is free if you just got off the train |
| 17:42.06 | minaguib | I'm explicitly thinking of a case where a SIP call is dropped into a context that does DIAL(IAX2/foo) |
| 17:42.19 | Qwell | ...only in your analogy, of course. that isn't the case in Asterisk :P |
| 17:42.38 | [TK]D-Fender | minaguib: If you call out using a PCI FXO interface... where do SIP Headers go? Bell does not know about "SIP headers". Your IAX2 call is no different in that respect |
| 17:42.46 | [TK]D-Fender | minaguib: * is a B2BUA. |
| 17:43.22 | [TK]D-Fender | [13:42]<minaguib>I'm explicitly thinking of a case where a SIP call is dropped into a context that does DIAL(IAX2/foo) <- IAX2 is completely different tech. Again, only things in common are CID & Audio |
| 17:44.19 | minaguib | Cool I get it |
| 17:45.51 | minaguib | Another question, security-wise, does IAX2 make provisions for the whole trusted/untrusted peers thing with regards to forwarding/stripping callerid based on presentation ? |
| 17:46.24 | *** join/#asterisk Z_God (~julius@wlan230145.mobiel.utwente.nl) |
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| 17:47.28 | [TK]D-Fender | minaguib: There is a rpesentation flag. Thats about it. |
| 17:49.31 | minaguib | [TK]D-Fender: Are there provisions for asteriskA to deem asteriskB untrusted and if that flag is set, to strip out the CID ? |
| 17:49.43 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.46.180) |
| 17:50.33 | [TK]D-Fender | minaguib: B Already knows if it trusts A or not and do whatever the hell it wants. |
| 17:50.46 | [TK]D-Fender | minaguib: All processing = dialplan |
| 17:52.28 | minaguib | [TK]D-Fender: But what about the other way around ? If providerA has to forward a call to providerB but it does not trust them to respect the flag, it must, itself, filter out the CID before forwarding |
| 17:52.42 | [TK]D-Fender | minaguib: Clearly |
| 17:53.06 | [TK]D-Fender | minaguib: Don't tell me a secret and then think about whether you should tell me. |
| 17:53.08 | minaguib | [TK]D-Fender: Does * offer dialplan tools for providerA to inspect that flag and strip out the CID ? |
| 17:53.20 | [TK]D-Fender | minaguib: Yes. |
| 17:54.28 | anonymouz666 | sharing a sipfriends table between two boxes, if I call from A to B in first box, but B is registered in the second box, will that work? |
| 17:54.57 | minaguib | [TK]D-Fender: Any tips on which commands will accomplish that ? I'd like to read up more offline instead of asking here. |
| 17:55.26 | [TK]D-Fender | minaguib: "core show application SetCallerPres" , "core show function CALLERID" |
| 17:55.44 | minaguib | [TK]D-Fender: Thank you very much. I appreciate the time and info |
| 17:56.24 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 17:56.59 | [TK]D-Fender | minaguib: You're welcome. Make it a topic for an MLUG get-together or something.... |
| 17:57.42 | minaguib | Hehehe. No time for LUGs/sleep after kids :) |
| 17:57.46 | p3nguin | Is there a comprehensive description of the presentation types? |
| 17:58.12 | p3nguin | The basic list provided in the docs assume you know what each thing means. |
| 18:02.11 | jonny330 | ChannelZ: i think i got some more info that can help us troubleshooth this problem |
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| 18:18.47 | ChannelZ | sorry had to wander away |
| 18:21.52 | iscario | hi, just a question abt sip : i heard sip had problem with NAT. Does it mean that if my sip clients are using nat i would need to set up a sip proxy ? or does sip encounter prb only when the server is behind nat ? |
| 18:26.30 | [TK]D-Fender | ~sipnat |
| 18:26.30 | 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 |
| 18:26.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
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| 18:33.18 | jdoe | hrm. Crap. Should have done the math. |
| 18:33.21 | jdoe | ulaw streams add up quick. |
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| 18:46.35 | niekvlessert | would it be possible to add options (call recording for example) to a call coming from a queue to a queue agent? |
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| 18:52.54 | jonny330 | is there anyway to get the inbound caller id from a pots line on a dahdi card? |
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| 18:53.28 | jonny330 | i would asume we would have to have caller id on the line right? |
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| 18:55.38 | jonny330 | any ideas on how to get inbound caller id from pots lines? |
| 18:55.55 | paulc | jonny330: it should support it out the box, provided you're getting it from the telco |
| 18:56.07 | paulc | some config file tweaking for format maybe (US vs UK vs Europe etc) |
| 18:56.13 | jonny330 | yeah i don't think i am getting it from telco |
| 18:56.24 | paulc | plug a regular phone or caller ID box in to verify? |
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| 18:57.37 | anonymouz666 | Using app_while (not using AEL2) is it possible once you hit a condition to BREAK the loop? |
| 18:58.25 | jonny330 | k thanks |
| 18:59.46 | anonymouz666 | ExitWhile |
| 18:59.47 | anonymouz666 | lol |
| 18:59.56 | anonymouz666 | pretty smart |
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| 19:22.03 | yonahw | I am having trouble setting up my box to receive calls from my sip provider, I have posted the configuration details at http://pastie.org/957566 |
| 19:22.59 | yonahw | i am receiving an error message of "Call from 'userid' to extension 'userid' rejected because extension not found." although I have setup a user in sip.conf which is mapped to an existing extension in extensions.conf |
| 19:23.19 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:23.22 | [TK]D-Fender | yonahw: extension 'userid' rejected because extension not found <--- as it states, you have no extension that matches that pattering in yoru dialplan in the context the call is landing in |
| 19:23.45 | yonahw | but shouldn't it match 's'? |
| 19:23.52 | ManxPower | yonahw, no! |
| 19:24.02 | yonahw | oh ok |
| 19:24.14 | ManxPower | "s" means "no extension". It does not mean "any extension" |
| 19:24.25 | yonahw | ok this makes more sense |
| 19:24.28 | [TK]D-Fender | ~stdextens |
| 19:24.28 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 19:24.43 | yonahw | thanks guys |
| 19:26.00 | wcselby | chanisavail always wants to return Status Unknown ... :( |
| 19:26.25 | wcselby | hmmmmm...... |
| 19:26.26 | [TK]D-Fender | checkout time, later all |
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| 19:30.31 | wcselby | Trying to determine the device state (in use, not in use, etc) for members of a queue before they're called. Not having much luck. Everything works as expected if the agent answers a call from the queue, they are not sent any more calls from that queue until they hang up. However, if they answer a call placed to them directly (not through the queue), or if they make a call outbound, then the queue will continue to send them calls and they will get cal |
| 19:32.20 | edwin_quijada | how can I asociate a recording call with agent and extension took the call? |
| 19:36.05 | *** join/#asterisk ACK-NAK (~Miranda@home.chicagoventure.com) |
| 19:36.07 | *** join/#asterisk voxter (~voxter@fl-67-233-9-119.dhcp.embarqhsd.net) |
| 19:36.48 | ACK-NAK | has the CLI originate command moved in 1.6.2? I know a lot of other ones have |
| 19:37.17 | *** join/#asterisk KnucKles_ (~bocao_198@189.26.255.140) |
| 19:37.25 | ACK-NAK | what's its new name? Anyone? |
| 19:37.32 | ACK-NAK | Bueller? |
| 19:38.20 | wcselby | ACK-NAK - don't think it's changed.... |
| 19:38.40 | ACK-NAK | Hmmm. |
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| 19:39.21 | ACK-NAK | I should be able to type at the CLI, originate [tech... etc] and originate a new call, right? |
| 19:39.23 | wcselby | if you're trying to originate from an AMI command, I think they added a new originate permission in manager.conf |
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| 19:40.14 | wcselby | ACK-NAK - I think so. |
| 19:40.30 | wcselby | ACK-NAK - if you type orig and then hit tab at the CLI prompt, does it auto-complete? |
| 19:40.48 | ACK-NAK | It does not. I'm trying to do something simple such as asterisk -rx originate ... |
| 19:41.01 | ACK-NAK | wcselby: I just get odbc |
| 19:41.40 | KnucKles_ | Hi All... Any of you could help me with DAHDI using Tormenta 3 E1 card(Tor 3 - from zapata telephony) ? |
| 19:42.39 | ACK-NAK | FOUND IT: Now called "Channel Originate" |
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| 19:57.40 | *** join/#asterisk KnucKles_ (~bocao_198@189.26.255.140) |
| 20:00.02 | KnucKles_ | Does anybody have the driver for tormenta 3 E1 card to run with DAHDI ? |
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| 20:07.55 | jasonjjohnsonjr | Does anyone have any recommendations for a simple CDR tool? I looked at freeside and a2billing but they seem like overkill just to give users access to CDR data. |
| 20:08.14 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 20:09.43 | KnucKles_ | jasonjjohnsonj: Look at http://www.areski.net/asterisk-stat-v2/about.php |
| 20:09.51 | KnucKles_ | It is a good tool |
| 20:10.48 | jasonjjohnsonjr | KnucKles_: It shows that the last release was 2005. Are there any issues with it working with 1.6.2? |
| 20:12.13 | KnucKles_ | Yes... I had tested into 1.6.1 |
| 20:12.19 | KnucKles_ | works ok! |
| 20:12.41 | KnucKles_ | opss.... No issues!!! |
| 20:12.55 | jasonjjohnsonjr | I will take a look. Thanks, |
| 20:13.00 | KnucKles_ | no problem! |
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| 20:18.22 | jsidhu | anyone know if we can use the Vonage Business plus account as a sip trunk? |
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| 20:34.53 | devmod | I was trying to add a local ext as a queue member, but I see invalid when doing queue show: Local/1000@agents (dynamic) (Invalid) has taken no calls yet |
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| 21:29.20 | wcselby | can hardware echo cancelation modules be bought and installed on digium boards that we already own? |
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| 21:29.36 | Qwell | wcselby: sure. find a reseller that will sell it separately |
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| 21:37.11 | d_preston215 | What does switchtype=national mean? |
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| 21:41.23 | wcselby | is _**XXXX, supposed to also match *8., ? |
| 21:42.25 | wcselby | nevermind, misread my cli |
| 21:45.32 | kazaa_lite | hi all |
| 21:45.46 | kazaa_lite | how can i configure asterisk to start at some very high priority? |
| 21:45.55 | ManxPower | d_preston215, It means the switch is provisioned for National ISDN 2 service |
| 21:46.06 | ManxPower | It is the most common PRI setup in the USA and Canada |
| 21:46.12 | d_preston215 | ok |
| 21:47.07 | ManxPower | kazaa_lite, "man asterisk" pay special attention to the "pseudo realtime" switch. Don't expect it to do anything other than cause problems. |
| 21:48.13 | d_preston215 | PRI T1 cards for the most part use pri_cpe as signaling? |
| 21:48.36 | ManxPower | d_preston215, when connecting to a telco, yes |
| 21:49.02 | d_preston215 | Would it be the same if I was using a PRI appliance such as a redFONE? |
| 21:56.47 | ManxPower | it would depend on how the redFONE was configured |
| 21:57.13 | ManxPower | you might need a crossover t-1 cable if you are connecting Astersik to a non-teleco port |
| 21:57.31 | d_preston215 | Made my own T1 crossover cables. |
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| 21:58.21 | ManxPower | you are sure they are not ethernet crossover? |
| 21:58.35 | d_preston215 | I'm sure. |
| 21:59.08 | d_preston215 | Basically switch out pins 4&5 with pins 1&2. |
| 21:59.36 | d_preston215 | 1 & 2 with 4 & 5 I mean. |
| 22:00.23 | d_preston215 | http://blog.elastixdepot.com/2009/11/19/configuring-a-redfone-fonebridge2-with-elastix-part-1.aspx |
| 22:00.40 | d_preston215 | Unless the instructions for the cable on here are wrong. |
| 22:01.43 | d_preston215 | But in any case, actual lines from the teleco would be signaled normally with pri_cpe. |
| 22:01.48 | d_preston215 | If I'm correct. |
| 22:02.27 | d_preston215 | Just that any loop back port would have to be signaled with pri_net. |
| 22:04.08 | edwin_quijada | how can I figure out in a queue what agent took the call and extension for recording the call?? |
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| 22:16.50 | rare1980_ | hi all i am trying to load modprobe ztdummy |
| 22:16.56 | rare1980_ | but i am getting msg that |
| 22:17.03 | rare1980_ | FATAL: Module ztdummy not found. |
| 22:17.10 | Qwell | Is it installed? |
| 22:17.19 | rare1980_ | please can any one guide me? |
| 22:17.41 | rare1980_ | Qwell: yes i have install zaptel |
| 22:17.49 | rare1980_ | on ubuntu kernal 2.6 |
| 22:18.02 | Qwell | How did you install it? |
| 22:18.32 | rare1980_ | i have download this ---- svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zaptel |
| 22:18.44 | rare1980_ | ./configure |
| 22:18.44 | rare1980_ | make |
| 22:18.45 | rare1980_ | sudo make install |
| 22:18.45 | rare1980_ | sudo modprobe ztdummy |
| 22:18.50 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:18.54 | Qwell | show me the output of those commands. |
| 22:18.56 | Qwell | ~pastebin |
| 22:18.56 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 22:19.00 | rare1980_ | sure |
| 22:19.09 | rare1980_ | plz 1 sec |
| 22:19.35 | rare1980_ | u mean output of all installation? |
| 22:19.38 | rare1980_ | results |
| 22:19.47 | Qwell | the entire output from all of those commands |
| 22:20.10 | rare1980_ | rite |
| 22:20.42 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:20.57 | rare1980_ | i am using putty.. i can't see all installation result on it.. is there any log file from where i can get the results? |
| 22:21.58 | rare1980_ | ? |
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| 22:23.33 | rare1980_ | qwell? please tell me is there any installtion log file |
| 22:23.38 | Qwell | no |
| 22:25.06 | rare1980_ | ok then let me get those results which i can see :) |
| 22:25.56 | rare1980_ | http://pastebin.com/zW7Ma3Jx |
| 22:26.08 | rare1980_ | here are the results on pastebin |
| 22:28.20 | Qwell | you didn't install it for the correct kernel version |
| 22:29.42 | rare1980_ | how can i install correct kernal version? |
| 22:31.21 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:31.21 | rare1980_ | Qwell: please can u guide me a bit? |
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| 22:39.09 | rare1980_ | atleast guide me which kernal ver is comptble with zaptel |
| 22:39.10 | rare1980_ | ? |
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| 22:39.37 | Deeewayne | rare1980_, did you run make config like the output told you to do? |
| 22:40.19 | Deeewayne | oh nm |
| 22:40.24 | XTeo | hi |
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| 22:41.07 | XTeo | it's necesary install Mysql with Asterisk, (without Freepbx or another web gui) ?? |
| 22:42.14 | decaffeine | XTeo, no |
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| 22:43.45 | decaffeine | good night all |
| 22:45.04 | XTeo | ok |
| 22:45.08 | XTeo | thans |
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| 22:47.59 | devmod | How can I check before dialing if a sip dev is online or not? |
| 22:48.25 | hardwire | hmm.. getting duplicate DTMF using AGI get_data.. seeing correct results with dtmf debugging |
| 22:48.37 | hardwire | juts the result from get_data is wonky if I dial too fast. |
| 22:49.00 | hardwire | so.. rfc2833 works fine.. can verify that in the pcap as well as the DTMF debugging via logger.conf |
| 22:49.08 | hardwire | I see the begins and ends.. they don't overlap |
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| 22:52.07 | nny | stupid question, is Asterisk sip notifyringing=no suppose to supress ringing notifications for hints? |
| 22:53.16 | nny | and if so, is there any reason why it wouldn't work? |
| 22:53.36 | ManxPower | hardwire, make sure you don't have rfc2833compensate=yes |
| 22:53.47 | ManxPower | as well as make sure relaxdtmf is not set |
| 22:54.16 | rare1980_ | i am unable to install kernal 2.6.26--- apt-get install linux-source-2.6.26 |
| 22:54.32 | rare1980_ | on ubuntu it says |
| 22:54.36 | rare1980_ | no package found |
| 22:57.18 | nny | right now, when I dial in with Asterisk sip notifyringing = no I get http://pastebin.org/227652 |
| 22:57.39 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 22:57.42 | hardwire | ManxPower: indeed.. since DTMF debugging is correct that appears to be a non issue (it's also not set) |
| 22:57.58 | nny | and the blfs on the phones flash... I have another system I can't test right now, but I thought it disabled ringing hints.. is it wrong or a bug with the newer version of asterisk i am running? |
| 22:58.00 | hardwire | both are not set |
| 22:59.02 | hardwire | ManxPower: I'm about to test something.. the playback file doesn't last very long.. I'm going to add a ton of silence to the end |
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| 23:06.01 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:10.51 | nny | hmm only found a little info on google, and the same query (still get ringing notification) anything I can do to diagnose further? |
| 23:12.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 23:13.05 | nny | [TK]D-Fender: can I pick your brain for a sec? |
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| 23:13.49 | [TK]D-Fender | nny: sure |
| 23:15.38 | nny | [TK]D-Fender: is Asterisk sip notifyringing = no suppose to keep asterisk from doing "Extension Changed 102[hints] new state Ringing for Notify User 101" |
| 23:17.18 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 23:17.33 | [TK]D-Fender | nny: It's supposed to suppress SIP 180 Ringing messages between the invite & answer/abort on that channel |
| 23:18.01 | sawgood | exten => _X.,1,Dial(SIP/itsp-01/${EXTEN}) |
| 23:18.12 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 23:18.19 | sawgood | how can I 'change' this statement to 'must' match a 9 in front |
| 23:18.37 | [TK]D-Fender | sawgood: exten => _9X.,1,Dial(SIP/itsp-01/${EXTEN:1}) |
| 23:18.42 | [TK]D-Fender | sawgood: Shove a 9 in front |
| 23:18.44 | sawgood | thank you |
| 23:18.49 | sawgood | perfect answer |
| 23:18.52 | sawgood | thank you so much |
| 23:19.04 | nny | [TK]D-Fender: ahh. I misuderstood. Is there a way to suppress ringing states for phones subscribed to hints? |
| 23:19.06 | [TK]D-Fender | sawgood: And as you hopefully noted I STRIPPED it off of the actual number being passed to that peer |
| 23:19.33 | [TK]D-Fender | nny: Hmm... odd request.... Nothing I can think of short of vi |
| 23:19.38 | sawgood | ty |
| 23:20.18 | [TK]D-Fender | sawgood: "Dial 9" prefixes are soooo 1980..... |
| 23:21.04 | nny | [TK]D-Fender: have a sidecar that blows up when all the phones ring at the same time, and another phone that has 2 blfs for other users. In both cases the blf flashing when the call comes in is more of a distraction than information for the users |
| 23:21.39 | nny | blows up = flashes/lights up all buttons at the same time ;) |
| 23:21.52 | nny | all though the other way would be damn interesting |
| 23:22.02 | [TK]D-Fender | nny: Sounds like An Aastra 5i series :) |
| 23:22.09 | nny | [TK]D-Fender: Cisco spa500 |
| 23:22.21 | [TK]D-Fender | nny: Shit-acular! |
| 23:22.30 | [TK]D-Fender | nny: Craptastic even... |
| 23:22.33 | ManxPower | Real Men use Polycom (tm) |
| 23:22.50 | nny | [TK]D-Fender: so I assume polycoms have a way to suppress that? |
| 23:23.06 | [TK]D-Fender | nny: No... Polycom's simply don't have a hissy fit over it :) |
| 23:23.07 | nny | [TK]D-Fender: or do they not flash blfs when a call comes in...? |
| 23:23.08 | ManxPower | nny, I doubt it, but I don't think they crash |
| 23:23.13 | nny | not hey dont crash ha |
| 23:23.15 | nny | they* |
| 23:23.33 | Naikrovek | polycom > * |
| 23:23.39 | nny | sorry miscommunicated, the fact that the blf blinks in a network where all phones/most phones ring on an incoming call is bad |
| 23:23.47 | nny | this would be the same for all phones I assume |
| 23:24.05 | nny | suprised that you can't disable it though |
| 23:24.09 | nny | at least not in * |
| 23:24.18 | [TK]D-Fender | nny: 2 solutions : vi chan_sip.c OR run it through a proxy of some kind that lets you filter. |
| 23:24.30 | WIMPy | nny: On the SAP962 you can define the visuals. |
| 23:24.41 | nny | WIMPy: i'll check the 508 thanks |
| 23:25.09 | WIMPy | If it's supported, just change the pattern to 'off'. |
| 23:25.25 | nny | WIMPy: gotcha, was gonna look over the interface again if asterisk could not suppress it |
| 23:25.28 | WIMPy | I think that was the only thing I liked about the SPA. |
| 23:25.59 | nny | I dunno, overall the phones work rather well |
| 23:26.18 | nny | do the polycoms have backlights now? |
| 23:26.23 | WIMPy | I like it so much, it's not even connected any more. |
| 23:26.56 | [TK]D-Fender | nny: Most of the newer modules, yes |
| 23:27.01 | WIMPy | I've never seen a Polycom. Would like to, but I'm not going to buy one, just out of interest. |
| 23:27.12 | [TK]D-Fender | nny: 335,450,550,560,650,660 all do |
| 23:27.38 | nny | that's better at least |
| 23:28.07 | nny | here's a question that popped up. Anyone implemented BLF state changes based on duration of a call on hold? if so what phone model? |
| 23:28.35 | nny | Some client had a Avaya or something that did that, they wanted the feature but I haven't had the chance to look around |
| 23:28.55 | Naikrovek | how would you do that? /me ponders. not being sarcastic. voip doesn't have the notion of "lines" like analog and digital systems sometiems do |
| 23:29.39 | Naikrovek | maybe you could have something that notified the phone via message or something when a parked call had been waiting too long |
| 23:29.43 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:29.44 | Naikrovek | or perhaps just call the receptionist back |
| 23:30.00 | nny | Naikrovek: apart from parking lot queue times, nothing I know of yet |
| 23:30.07 | nny | and all that does is return the call to the user |
| 23:30.13 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:30.13 | Naikrovek | yeah |
| 23:30.21 | WIMPy | hold or park? |
| 23:30.36 | nny | hmm well, the avaya was hold, but park would be more useful |
| 23:30.54 | WIMPy | Makes sense that way. |
| 23:31.12 | nny | it was more of an answer to a simple system, numerous lines on hold, which one has been there the longest kind of thing (as far as I know) |
| 23:31.24 | Naikrovek | yeah that's valid |
| 23:31.48 | nny | WIMPy: yeah the cisco SPAs show promise, although their engineers craptastically add bugs to every firmware update... |
| 23:32.02 | Naikrovek | polycom phones can kinda handle that. they can show you how many calls you have on hold and how long each has been on hold. doesn't do sorting based on hold time or anything though |
| 23:32.06 | *** join/#asterisk hfb (~hfb@pool-98-112-146-69.lsanca.dsl-w.verizon.net) |
| 23:32.20 | nny | Naikrovek: yeah the SPAs will show you how long a line has been on hold for too |
| 23:32.21 | WIMPy | nny: I don't see any promise there. |
| 23:32.38 | nny | WIMPy: well, it's not game breaking bugs, so far it |
| 23:33.09 | WIMPy | I think they're totally crappy from the users point of view. |
| 23:33.38 | nny | it's been sidetone missing, (patch other things fixed) firmware provisioning via tftp doens't catch changes to the xml files (fixed). The latest hasn't show any issues. To be honest the Cisco SPAs are 6 months old or so |
| 23:33.39 | Naikrovek | the SPAs? |
| 23:34.09 | WIMPy | Nice hardware design (from the outside) but usability went past them. |
| 23:34.12 | nny | I dunno, I have 40 or 50 SPAs in the field at various sites, and only a half dozen minor requests or features that they want |
| 23:34.22 | nny | no real complaints per se |
| 23:34.35 | nny | and even the bugs were only known and handled by me, they didn |
| 23:34.40 | nny | 't even realize sidetone was missing |
| 23:35.07 | nny | I am a bigger nazi about the phones than anyone else heh |
| 23:35.38 | nny | then again this is the Cisco SPAs predominately, the linksys ones lacked some useful features that the ciscos do normally |
| 23:36.10 | *** join/#asterisk generalhan (~asd@about/windows/staff/generalhan) |
| 23:36.13 | WIMPy | Just ask someone used to a modern telephone system and they will usually be quite unhappy with any voip phone. |
| 23:36.35 | nny | like customizable line keys (change to BLF, speed dials) and programmable soft keys (which is still not perfect) they end up doing the button combination on a new line |
| 23:36.44 | nny | my number one "complaint" is shared line appearance related |
| 23:36.55 | nny | blfs that show parked call states pretty much rides that out |
| 23:37.04 | WIMPy | Yes, complete bullocks. |
| 23:37.13 | nny | most people are used to saying pick up line one etc |
| 23:37.21 | WIMPy | (the SLA stuff) |
| 23:38.01 | WIMPy | People stuck in the 70s that is. |
| 23:38.24 | nny | yeah |
| 23:38.44 | nny | tbh I tell people it's a hold over from key systems, there is no reason to use it on a modern VoIP system |
| 23:38.53 | nny | and then show them park ;) |
| 23:39.03 | nny | i even numbered the parking lots 1-20 heh |
| 23:39.11 | nny | so they can still say "pick up one" |
| 23:39.16 | WIMPy | Unfortunaletly people in non-american countries usually aren't. ;) |
| 23:40.12 | ChannelZ | Hmmm. Does the ksoftirqd process just sit there consuming ~15-20% CPU for anyone else? |
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| 23:44.02 | nny | hmm on the whole change led states thing. With the Line Key LED Pattern changes, plus a sip header modification, you could in theory do something with park + led patterns. probably have to do some "manual" parking stuff though |
| 23:45.00 | nny | not sure how much control i have over the park application, meh, back to my first issue |
| 23:45.12 | nny | [TK]D-Fender: WIMPy thanks for the heads up |
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| 23:47.13 | devmod | can reliably I use ChanIsAvail on SIP endpoints registered to asterisks to know if they are online or not? |
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