| 01:09.05 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 01:09.05 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1 (2011/04/01), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 01:23.07 | *** join/#asterisk ccesario (~ccesario@201-42-156-58.dsl.telesp.net.br) |
| 01:33.07 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 01:37.45 | *** join/#asterisk dr00d (~rtp.aster@b2A50.static.pacific.net.au) |
| 01:42.22 | atan | Does Barbados still have stupid high VOIP rates? |
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| 02:06.36 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
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| 02:20.11 | *** join/#asterisk kaushal (~kaushal@182.72.14.170) |
| 02:35.06 | kaushal | what does w1g1 mean ? |
| 02:38.22 | carrar | call sangoma |
| 02:38.36 | kaushal | ok |
| 02:38.45 | carrar | sounds like a firmware issue |
| 02:38.54 | carrar | need a upgrade or something or rather |
| 02:38.55 | kaushal | carrar: i need help for FreePBX |
| 02:39.45 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 02:40.15 | carrar | try #freebpx |
| 02:41.05 | kaushal | Is there a step by step guide to Configure IVR(Inbound and Outbound) in AsteriskNow using FreePBX ? |
| 02:41.53 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
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| 04:07.28 | suxx | hello everybody ! |
| 04:15.35 | Sertys | :) |
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| 04:47.33 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 04:49.00 | ghostmediapro | please help with dialing prefixes, i have two account gvoice which uses prefix 48 and gvoice2 which uses 58, problem is when dialing from either the same cid is show on my cell, which is 7086839000 |
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| 04:56.45 | *** join/#asterisk luckman212_ (~irc@pool-74-108-1-53.nycmny.fios.verizon.net) |
| 04:57.20 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 04:59.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:03.17 | kb3ien | how are you setting the cid ? |
| 05:03.35 | ghostmediapro | no cid set |
| 05:03.53 | ghostmediapro | hoping the cid will be picked by the extention i use |
| 05:05.16 | kb3ien | you can set the callerid in the dialplan, or you can set the callerid in the TECH.conf file that corresponds to the device that initiates the call. do you mean to do the latter? |
| 05:08.52 | ghostmediapro | kb3ien: when i dial 58 which is gvoice2 i want that number to display |
| 05:23.03 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-173.telkomadsl.co.za) |
| 05:26.50 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:28.46 | WiretapSeven | kb3ien, he's using freepbx and chastising people who try to help him in that channel :) |
| 05:33.24 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.27.93) |
| 05:39.06 | kb3ien | if you want to set the callerid based on what is dialed NOT who dialled it, use the dialplan. Set(CALLERD(all)=name<number>). |
| 05:39.42 | ChannelZ | Who says being a lying bastard can't be easy!? |
| 05:39.57 | kb3ien | as to "want that number" not sure which number "that" is. |
| 05:40.04 | kb3ien | Not me. |
| 05:40.16 | kb3ien | oh, wait i do. QED |
| 05:41.26 | kb3ien | i suppose 'exten => _58X.,n,Set(CALL' .... is where you headed. |
| 05:41.42 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
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| 05:48.16 | *** join/#asterisk [T]ank1 (~Tank@c-76-27-9-24.hsd1.ut.comcast.net) |
| 05:51.13 | [T]ank1 | hey all, still trying to get a transfer from one asterisk server to another to work. When i use the dial command, the call works just fine. if i change the command from dial to transfer i get a status unknown error. here are a few logs and configs: http://pastebin.ca/2048544 |
| 05:51.19 | [T]ank1 | any help would be appreciated |
| 05:53.14 | wdoekes2 | [T]ank1: the 302 moved contact is bad |
| 05:53.30 | wdoekes2 | Contact: Transfer <sip:11839@ICR> |
| 05:53.42 | wdoekes2 | you want that to be something that can be looked up by the avaya |
| 05:54.07 | [T]ank1 | what do you mean? |
| 05:54.08 | wdoekes2 | i.e. s/ICR/ip.of.asterisk2/ |
| 05:54.27 | WiretapMac | quick question, does the debian install of asterisk include g.722 by default? |
| 05:54.55 | [T]ank1 | where would i set that up at? |
| 05:55.10 | wdoekes2 | "the debian install"? most people compile from source because the packages lag too much |
| 05:55.22 | wdoekes2 | in the dialplan.. where you do Transfer(...) |
| 05:55.57 | WiretapMac | wdoekes2: compiling on a VIA C3 is no fun |
| 05:56.50 | wdoekes2 | I can dig that. open the .deb and check if it has codec_g722.so |
| 05:57.09 | WiretapMac | I'll know now where to find it, thanks |
| 05:57.13 | WiretapMac | hiding in asterisk's modules |
| 05:57.21 | WiretapMac | (its already installed) |
| 05:57.30 | [T]ank1 | wdoekes2: so instead of a Transfer(SIP/11839@ICR) would it be Transfer(SIP/11839@10.40.0.8)? |
| 05:57.36 | WiretapMac | yep, installed, sweet |
| 05:57.53 | wdoekes2 | indeed [T]ank1.. if that avaya can reach that and intereprets 302 correctly, it will work |
| 05:58.23 | [T]ank1 | k, ill try that. |
| 05:58.31 | wdoekes2 | *gone*, good luck |
| 06:01.16 | [T]ank1 | still gives me a -- Executing [11839@default:1] Transfer("SIP/<ipremoved>-1851e820", "SIP/11839@10.40.0.8") in new stack |
| 06:01.16 | [T]ank1 | <PROTECTED> |
| 06:05.29 | *** join/#asterisk FastEddie (~fast@c-76-29-218-21.hsd1.il.comcast.net) |
| 06:05.42 | FastEddie | Evening/morning all |
| 06:06.17 | FastEddie | Just installed 1.8, trying to get inbound calling working. |
| 06:06.18 | FastEddie | <PROTECTED> |
| 06:06.49 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 06:06.54 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 06:08.10 | WiretapMac | FastEddie: it is trying to authenticate the endpoint |
| 06:08.32 | WiretapMac | and finding that the username sent by the remote endpoint is not what it expects |
| 06:09.13 | FastEddie | WiretapMac, thanks for the response. I was running 1.4 before, so I'm trying to modify my existing sip.conf to work :) |
| 06:09.16 | WiretapMac | your username is amusing to me as Fast Eddie was the name of a little kid who featured in an ad for the country's largest telco :P |
| 06:09.52 | FastEddie | In the UK? |
| 06:10.10 | WiretapMac | NZ |
| 06:11.26 | FastEddie | Ah, I'll have to ask a friend that lives there about it. |
| 06:11.59 | WiretapMac | it may not be known to the public |
| 06:12.15 | WiretapMac | I was associated with the company at the time and it was as far as I know, just an internal codename |
| 06:12.59 | FastEddie | Gotcha, I won't ask him then :) |
| 06:13.25 | WiretapMac | essentially a 5 year old who played the part of a 'fast busmissim', one sec, I'll see if its on yt |
| 06:13.45 | FastEddie | I am using "Vitelity" as my provider. The error is chan_sip.c:13450 check_auth: username mismatch, have <vitel-inbound>, digest has <vitelusernamehere>. So vitelusernamehere is my username with vitelity that I register with. vitel-inbound is a sip friend I have defined. |
| 06:13.55 | WiretapMac | http://www.youtube.com/watch?v=1HpJMV8ia54 look at that, title and all :P |
| 06:13.55 | FastEddie | Thanks, I'd actually like to see it |
| 06:14.23 | WiretapMac | FastEddie: some providers require that the trunk context match the DID they have assigned you |
| 06:14.35 | WiretapMac | otherwise it doesn't work |
| 06:14.36 | FastEddie | That is hilarious, thanks for sharing. |
| 06:15.17 | FastEddie | WiretapMac: ok, I will try that now. What's sad is I am using the "wizard" generated code from them for the sip info and register statement :) |
| 06:15.31 | WiretapMac | lol XD |
| 06:15.36 | WiretapMac | I primarily use freepbx |
| 06:16.17 | FastEddie | Vitelity being the provider, has a feature to "generate asterisk SIP.conf" to use. It's obviously not working for 1.8, but did for 1.4 |
| 06:16.47 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 06:18.05 | FastEddie | Man, renaming the [vitel-inbound] friend in sip.conf to my vitelity username it was referring to in the mismatch error worked. You rock WiretapMac, thanks~! |
| 06:18.20 | FastEddie | Seems obvious in hindsight, so something changed within 1.8 on the handling of that |
| 06:18.53 | FastEddie | Thank you for responding to my question, btw! |
| 06:19.49 | WiretapMac | FastEddie: changed in 1.6 |
| 06:19.53 | WiretapMac | as far as I know |
| 06:20.02 | WiretapMac | as Trixbox 2.8 required it too |
| 06:45.58 | [T]ank1 | anyone out there well versed in Avaya and Asterisk? Enough to get a call to come from the Avaya session manager to Asterisk then transfer off of asterisk to another asterisk server without bridging the calls? I am willing to pay for services. |
| 06:51.14 | *** join/#asterisk marlowe_ (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
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| 07:12.47 | wolfe | I might as well hand around here :) |
| 07:14.46 | *** join/#asterisk Infin1ty|work (~erez@pdpc/supporter/active/infin1ty) |
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| 08:50.13 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 08:50.13 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1 (2011/04/01), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 08:54.47 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 08:59.02 | *** join/#asterisk jizzzum6 (~jizzzum6@titan.darksideresearch.com) |
| 09:06.19 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 09:06.46 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 09:14.37 | *** join/#asterisk davlefou (~david@196.203.21.69) |
| 09:23.28 | *** join/#asterisk Smirker (Smirker@14-202-69-225.static.tpgi.com.au) |
| 09:27.17 | Smirker | Howdy. I have many DIDs coming into my Asterisk box. Handling incoming calls, I use lots of Gotos to achieve various goals. This works wonderfully, however in my calls logs the 'src' field is correct (the person who is calling one of my DIDs), however the 'dst' field always shows up as 's' (where it should be the DID the caller called i would have thought). can anyone shed some light on |
| 09:27.17 | Smirker | this or know how i would fix it |
| 09:29.15 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 09:33.29 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 09:36.36 | wdoekes2 | Smirker: the extension that does the Dial() should look like your dst did: exten => +1234567,1,Dial(SIP/someuser) ; now +1234567 will show up as dst |
| 09:41.19 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
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| 10:07.17 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 10:17.46 | *** join/#asterisk lep (~Adium@ppp-24-57.21-151.libero.it) |
| 10:22.22 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 10:23.16 | as001 | Hi I have great problem with asterisk manager interface. It worked fine until yesterday and I did not change it but today it connects and asterisk emidietely disconnect it |
| 10:23.28 | *** join/#asterisk mzb (~mzb@150.101.108.88) |
| 10:24.25 | as001 | CLIENT 3 got disconnect |
| 10:29.23 | as001 | Client 3 got read error 0 (Normal disconnection) |
| 10:30.45 | *** part/#asterisk lep (~Adium@ppp-24-57.21-151.libero.it) |
| 10:41.02 | jizzzum6 | as001 what did you change? |
| 10:41.12 | as001 | nothing |
| 10:41.24 | jizzzum6 | before you changed that, what did you change? |
| 10:41.49 | as001 | I didn't change single line in script which connects to ami |
| 10:42.06 | jizzzum6 | hmmmm |
| 10:42.15 | jizzzum6 | i guess you need to change something now |
| 10:43.21 | ixyd_ | have you turned on full logging and watched the log? |
| 10:43.33 | kaldemar | as001: did you upgrade asterisk? |
| 10:46.11 | as001 | no |
| 10:47.17 | as001 | i can see just client 3 got read error 0 what does that mean |
| 10:47.26 | as001 | client can not read something ? |
| 10:48.21 | as001 | i disabled auto_logoff part which starts during connection time and has bunch of mysql queries and it can connect.. |
| 10:48.51 | as001 | it seems it is mysql problem from server 2.. Thanks anyway. |
| 10:49.02 | *** join/#asterisk Iiiak (~Iiiak@AMontpellier-551-1-96-21.w92-145.abo.wanadoo.fr) |
| 10:56.07 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 11:11.33 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
| 11:16.20 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 11:16.22 | hetii | Hello :) |
| 11:17.12 | hetii | i use: DAHDI Version: 2.4.1 and have 4x SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) The driver used by card is: wcb4xxp |
| 11:17.40 | hetii | Im able to recive call but for the outgoing i got thi: |
| 11:17.40 | hetii | s |
| 11:18.36 | hetii | <PROTECTED> |
| 11:19.29 | hetii | on misdn drivers i be able to set msn numbers, maybe here this is also required. How can i do that ? |
| 11:19.54 | hetii | the CID i have set but i suppose MSN!=CID |
| 11:20.03 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:24.23 | *** join/#asterisk sourcode (~code@58.8.73.154) |
| 11:28.56 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 11:29.55 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 11:33.20 | eject_ck | What does it mean "Channel 'Local/XXXXXXX@outbound-allroutes-0f7f;2' has no RTP, not doing anything" ? |
| 11:35.12 | *** join/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 11:37.34 | kaldemar | eject_ck: that Local channel driver has no RTP interface and it won't be bridged. |
| 11:38.38 | hetii | btw this is my confing of dahdi file for * http://pastebin.ca/2048614 |
| 11:39.09 | hetii | why there are on one time group=0,11 and group = 63 |
| 11:43.43 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:45.04 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 11:45.06 | nix8n82 | if I download the new asterisk book from safaribooksonline.com do the people that write it get any thing from the free tokens I used to get it? |
| 11:49.58 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:09.37 | *** join/#asterisk carl0s- (~carl@host217-37-179-49.in-addr.btopenworld.com) |
| 12:10.11 | *** join/#asterisk tasca (~tasca@189.34.27.64) |
| 12:11.14 | carl0s- | I have a feature code (*5023 at the moment) that calls a bash script to open a door via a USB controlled relay. How can I make it so that I can do it during a call? I guess I could use a http url but I was thinking of something more universally compatible with different phones ? I have a mixture of Snom 370s, Polycom 501s, and Yealink T28Ps |
| 12:14.25 | ixyd_ | @carl0s have a look at http://www.freepbx.org/forum/freepbx/tips-and-tricks/features-conf-custom-command |
| 12:14.36 | carl0s- | (other than totally re-writing my dialplan to listen for the digits within normal call flow). I sort of thought there might be a standard way of programming a hard/soft key on most SIP phones, like a hold/transfer key but to call my custom script instead. |
| 12:15.05 | carl0s- | ixyd_, thanks.. looking at that now |
| 12:15.30 | carl0s- | ixyd_, damn.. that's almost too easy! |
| 12:15.36 | ixyd_ | ;) |
| 12:15.39 | carl0s- | ixyd_, right on the subject too |
| 12:15.40 | carl0s- | cheers :) |
| 12:15.46 | ixyd_ | have fun :) |
| 12:17.18 | carl0s- | thanks |
| 12:18.59 | ixyd_ | np :) |
| 12:19.17 | *** join/#asterisk devil_evoxxx (~d3v1l@157.27.183.122) |
| 12:19.26 | devil_evoxxx | hi all :) |
| 12:20.34 | *** join/#asterisk mawhii (~mawhii@170.220.119.70.cfl.res.rr.com) |
| 12:22.24 | carl0s- | just one thing.. that will happen during any call won't it? Any simple way of restricting it to only calls coming in on a certain zap channel ? |
| 12:22.38 | carl0s- | or sip trunk.. since I'm going to use an FXS ATA onto the door phone thingy |
| 12:27.53 | carl0s- | I guess I can use some sort of if/goto thing.. I'll just read up on basic dialplan syntax (I'm a cheating FreePBX user) |
| 12:31.09 | *** join/#asterisk m_tadeu (~quassel@89.180.63.7) |
| 12:32.12 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
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| 12:36.24 | devil_evoxxx | i'm not sure, try to put a NoOp(${CHANNEL}) |
| 12:37.31 | devil_evoxxx | and look in cli what this command print |
| 12:44.31 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 12:48.40 | m_tadeu | I have callers waiting in a queue, listening to a nice moh. how can I warn them that the call is about to be answered by the agent? |
| 12:55.10 | *** join/#asterisk glam (~glam@183.13.29.252) |
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| 13:09.59 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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| 13:17.27 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 13:18.58 | *** join/#asterisk maxxya (c30522cb@gateway/web/freenode/ip.195.5.34.203) |
| 13:19.08 | maxxya | hi |
| 13:21.45 | maxxya | i need help, who can do it? |
| 13:22.24 | *** join/#asterisk dandate2 (~dan@222.127.64.240) |
| 13:22.33 | dandate2 | op_server.pl using a lot of cpu today, anyone have any idea why? |
| 13:22.47 | anonymouz666 | op_server.pl is FOP a perl written script. |
| 13:22.56 | anonymouz666 | it kills the machine performance under heavy load |
| 13:23.28 | dandate2 | yeah what should i do? |
| 13:23.38 | dandate2 | it wasnt an issue before, just since i turned on periodic announcements has it been acting up |
| 13:24.07 | anonymouz666 | report the author. |
| 13:24.39 | anonymouz666 | If the problem is easy to reproduce then probably will get fixed, otherwise... |
| 13:26.05 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 13:26.08 | maxxya | how connect 3com 3102/3 phone to asterisk????? |
| 13:26.52 | dandate2 | will amportal stop_fop break my pbx in anyway? |
| 13:28.28 | anonymouz666 | stop_fop will stop the op_server.pl |
| 13:28.30 | anonymouz666 | not asterisk |
| 13:28.39 | anonymouz666 | I don't know what is amportal |
| 13:30.51 | *** join/#asterisk zorp75ck (~zorp75ck@2610:8:7800:f:c62c:3ff:fe19:aabb) |
| 13:31.45 | dandate2 | well stop_fop fixed it |
| 13:32.20 | anonymouz666 | heh... you restarted the fop. whatever makes the server to eat a lot of CPU, it will start again sooner or later |
| 13:32.49 | dandate2 | damn |
| 13:32.52 | dandate2 | how to disable fop forever? |
| 13:33.06 | anonymouz666 | don't run fop |
| 13:33.38 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:33.38 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:34.53 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 13:34.56 | maxxya | hey, somebody know how connect 3com 3103 to asterisk??? |
| 13:35.25 | anonymouz666 | maxxya: is this a SIP phone? |
| 13:35.52 | maxxya | yes |
| 13:35.56 | luckman212_ | [2011-04-20 09:33:29] WARNING[9247] loader.c: Error loading module 'res_fax_spandsp.so': /usr/lib/asterisk/modules/res_fax_spandsp.so: undefined symbol: t30_set_tx_page_header_info |
| 13:36.26 | luckman212_ | trying to compile ast 1.8.3.2 with res_fax_spandsp |
| 13:37.05 | luckman212_ | it builds but I see that in /var/log/asterisk/full during startup and module show like fax |
| 13:37.41 | anonymouz666 | maxxya: so what's the problem? I should connect your phone just like all other SIP phones |
| 13:37.46 | anonymouz666 | *you |
| 13:38.15 | anonymouz666 | luckman212_: spandsp mismatch version? |
| 13:38.57 | luckman212_ | i did a 'make clean' before compiling... so versions should be ok |
| 13:40.42 | maxxya | anonymouz666: do u know how do it? |
| 13:41.13 | maxxya | anonymouz666: what is NCP? |
| 13:41.21 | anonymouz666 | luckman212_: tell that to res_fax_spandsp.so :P |
| 13:41.39 | anonymouz666 | maxxya: I don't have idea |
| 13:43.21 | maxxya | anonymouz666: it dont want connect to asterisk, want connect to ncp |
| 13:45.17 | anonymouz666 | sorry I can't help you |
| 13:45.48 | luckman212_ | anonymouz666: how do I tell if its really a version mismatch? i don't see how that would be possible |
| 13:47.44 | aberrios | maxxya: and what makes you say it wants to connect to ncp? |
| 13:50.41 | maxxya | aberrios: it write on display: Connecting to NCP... |
| 13:51.25 | aberrios | maxxya: have you done a factory reset on this phone? |
| 13:52.53 | maxxya | aberrios: yes. in config NCP IP i write ip of asterisk, but dont work |
| 13:54.02 | aberrios | maxxya: just found this on the mailing list: http://lists.digium.com/pipermail/asterisk-users/2006-March/145510.html |
| 13:54.19 | aberrios | maxxya: has this changed since 2006? |
| 13:58.43 | *** join/#asterisk bratner (~bratner@95.211.21.37) |
| 13:59.18 | maxxya | aberrios: do u mean my phone year model? |
| 13:59.41 | bratner | Hi! I'm trying to debug my sip proxy using asterisk 1.4. What do i need to do to make astersik send re-INVITEs? |
| 13:59.49 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:59.59 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:00.16 | aberrios | maxxya: The post says that the 3com IP phones only support 3com PBXs via SIP |
| 14:00.27 | aberrios | maxxya: that post was made in 2006 |
| 14:00.45 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:01.37 | *** join/#asterisk Freeaqingme| (~dolf@dsl-083-247-011-232.solcon.nl) |
| 14:01.54 | *** join/#asterisk Freeaqingme| (~dolf@83.247.11.232) |
| 14:04.06 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:5caf:54dc:72d7:c033) |
| 14:04.33 | *** join/#asterisk tahiralmas (~tahir@182.177.209.40) |
| 14:06.18 | maxxya | aberrios: thx for help |
| 14:08.14 | *** join/#asterisk af_ (~getsmart@78.134.22.21) |
| 14:09.36 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 14:09.54 | *** join/#asterisk agroman (~agroman@fuckyeah.org) |
| 14:11.05 | agroman | is there a way to increase the logging level for res_calendar_ews? debug mode? |
| 14:13.58 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:13.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:20.16 | pigpen | agroman, I think just setting "core set debug X" will do the job. But I haven't had the need to debug res_calendar_ews. |
| 14:20.28 | *** join/#asterisk wonderworld (~ww@port-92-201-218-117.dynamic.qsc.de) |
| 14:22.26 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 14:24.50 | agroman | pigpen: thanks, I'm having trouble with it and I'm not certain if I've entered the correct url in the conf file. |
| 14:26.41 | pigpen | well, I think the url you are referring to is of the exchange server. Do a netstat or a pack sniff to see if it is in fact communicating with it. |
| 14:26.56 | pigpen | tcpdump, wireshark, etc... |
| 14:27.33 | agroman | pigpen: that is difficult see, because it uses SSL. i can't view the HTTP headers/requests |
| 14:27.54 | pigpen | you can at least see if it is sending/receiving packets. |
| 14:28.23 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 14:28.24 | agroman | oh, it is. i ran tcpdump yesterday and it is definitely initiating connect to the correct IP. |
| 14:28.35 | pigpen | ah...so it is trying. |
| 14:28.37 | pigpen | that is good. |
| 14:28.52 | agroman | yes, but no log entries on success/failure. |
| 14:29.05 | agroman | and it shows that my calendar is empty/free. which is not correct. |
| 14:29.27 | pigpen | ah. yeah, was going to mention logs next. |
| 14:30.06 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:30.49 | agroman | yes, in the logs I can get it to print messages when I insert the incorrect username/password... http://pastebin.com/zzh1Tq6e |
| 14:31.42 | agroman | but with correct username/pass it appears to authenticate (ie, the messages in the pastebin url stop being logged) ... but I dont know why it's not getting my calendar. |
| 14:32.40 | MrTelephone | I switched over to asterisk 1.8 today. Did the g1 for group dialing change to i1? |
| 14:34.15 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:34.40 | pigpen | do you mean dahdi groups? |
| 14:34.43 | *** part/#asterisk tahiralmas (~tahir@182.177.209.40) |
| 14:35.19 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 14:35.43 | pigpen | g1, g2, g3, are typically used in dial strings such as DAHDI/g1 |
| 14:36.07 | pigpen | but you say group dialing. so I am confused. |
| 14:36.39 | Cadey | Hi guys, is there anyway to see if a channel that has been formed got created by the Queue App as from the AMI all I can see if the channel being created and set to up, then rinings. Nothing in the dialplan out put or the call output etc informs that this channel is from the queue app |
| 14:39.06 | Cadey | its ok |
| 14:39.32 | Cadey | we can work it out with the ~HASH~SIP_CAUSE~SIP variable |
| 14:41.24 | MrTelephone | calls from the PSTN say DAHDI/i1/<NUMBER> |
| 14:41.41 | _Corey_ | MrTelephone: That's for incoming |
| 14:44.44 | MrTelephone | ok |
| 14:45.06 | agroman | http://www.asterisk.org/doxygen/trunk/res__calendar__ews_8c.html#8ff35cbf25b1087e5b177d5d6b75d933 --- from my error, I'm assuming that res_calendar_ews only uses HTTP Basic authentication. is that correct? |
| 15:09.37 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:09.58 | angryuser | Hello, i have 2 clock sources in my 1.6 system, how to know which one is used ? |
| 15:11.35 | angryuser | cd /sys/devices/system/timer/timer0/ show me nothing |
| 15:12.51 | *** join/#asterisk Tim_Toady (~moi@79.103.50.246) |
| 15:18.10 | *** join/#asterisk clintc (~clintc@128.227.125.7) |
| 15:18.37 | luckman212_ | when compiling * under the Utilities section, there are a bunch of extras (aelparse, astcanary, astman, check_expr, check_expr2, conf2ael, hashtest, hashtest2, muted, refcounter, smsq, stereorize, streamplayer) -- I can't find docs on any of these; what they do etc. Anyone know?? |
| 15:23.06 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 15:23.55 | angryuser | Its cat /proc/dahdi/* |
| 15:24.06 | angryuser | answerring my question |
| 15:28.44 | *** join/#asterisk jkprg (~jarda@ip-62-245-93-150.net.upcbroadband.cz) |
| 15:29.57 | *** join/#asterisk Ecco (~User@anj75-2-88-162-180-91.fbx.proxad.net) |
| 15:30.03 | Ecco | Hi everyone |
| 15:30.22 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 15:31.03 | Ecco | Quick question : is asterisk the right choice if all I'm ever going to use is SIP ? |
| 15:32.21 | angryuser | Ecco, well depends of usage |
| 15:32.31 | angryuser | Ecco, describe your needs |
| 15:32.34 | Ecco | Sure |
| 15:32.38 | Ecco | We're a 10 ppl team |
| 15:32.56 | Ecco | we're currently paying a SIP provider, but it only allows a max of 3 simultaneous connexions |
| 15:33.14 | Ecco | also, we would like to be able to easily transfer calls from one person to another |
| 15:33.21 | Ecco | and... that's pretty much it :-) |
| 15:33.28 | Ecco | Everyone's always going to use a SIP softphone |
| 15:33.40 | _Corey_ | Ecco: A lot of people using SIP providers use Asterisk... |
| 15:33.44 | Ecco | ok :-) |
| 15:34.08 | angryuser | Ecco, you will still have only 3 simultaniout calls to outside, for all the rest asterisk is good for you |
| 15:34.57 | Ecco | Actually AFAIK it's limited to a single simultaneous call |
| 15:35.02 | Ecco | (already) |
| 15:35.07 | Ecco | but 3 people can connect |
| 15:35.12 | Ecco | only just one of them can call |
| 15:35.31 | Ecco | Ok, so it's good to know Asterisk is the right tool |
| 15:35.48 | Ecco | Now (and that's where things get dirty) |
| 15:35.50 | Ecco | I'm running asterisk on the same box that does the NAT-ing of our network |
| 15:36.05 | Ecco | (because obviously, we only have one public IPv4 address) |
| 15:37.12 | angryuser | Ecco, you need to protect asterisk with firewall |
| 15:37.26 | angryuser | Ecco, i advice to change default bindport from 5060 to something else |
| 15:37.37 | Ecco | Actually there's iptables running already |
| 15:37.48 | angryuser | Ecco, Also reistrict sip traffic to your users & provider |
| 15:37.51 | Ecco | With pretty strict rules on the "outbound" interface |
| 15:38.02 | Ecco | yep, this I've already done |
| 15:38.13 | angryuser | Ecco, in this case all you need is a book |
| 15:38.17 | angryuser | ~book |
| 15:38.17 | infobot | For more information about the Asterisk book, see ~thebook |
| 15:38.24 | angryuser | ~thebook |
| 15:38.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 15:38.34 | therawr | yawns |
| 15:38.41 | wolfe | yawns back at The_Boy_Wonder |
| 15:38.42 | Ecco | I've downlaoded the book |
| 15:38.43 | wolfe | oop |
| 15:38.45 | therawr | lol |
| 15:38.48 | wolfe | yawns back at therawr |
| 15:38.50 | Ecco | thing is, it doesn't cover SIP so well |
| 15:38.55 | therawr | tab completion +1 |
| 15:38.59 | Ecco | nor the issues you may have when dealing with a nat-ed environment |
| 15:39.07 | wolfe | damn you tab completion, damn you! |
| 15:39.12 | therawr | wolfe: ;) |
| 15:39.30 | angryuser | Ecco, well, try and come back when you have problems |
| 15:39.41 | angryuser | nat=yes for phones accounts |
| 15:39.47 | wolfe | anyone here a VTC engineer? |
| 15:39.57 | Ecco | Yep |
| 15:40.05 | Ecco | So, here's my issue : outgoing calls work like a charm |
| 15:40.16 | Freeaqingme| | pull the plug, and your issue is solved |
| 15:40.28 | wolfe | Ecco: Affirmation to what for the "Yep"? |
| 15:41.05 | *** join/#asterisk Tim_Toady (~moi@79.103.50.246.dsl.dyn.forthnet.gr) |
| 15:41.12 | Ecco | (to 'I should come back when I have issues') |
| 15:41.18 | wolfe | ah |
| 15:41.25 | angryuser | Ecco, continue |
| 15:41.30 | Ecco | So dialing out works just fine. But for some reason when someone calls on our "shared" SIP line |
| 15:41.33 | Ecco | at all |
| 15:41.33 | angryuser | Ecco, dont be shy |
| 15:41.36 | Ecco | it doesn't ring |
| 15:41.41 | Ecco | :-) |
| 15:42.05 | angryuser | Ecco, you have added register lines right ? |
| 15:42.22 | angryuser | Ecco, 'sip show registry' show what ? |
| 15:42.30 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 15:42.34 | Ecco | 1 SIP registrations. |
| 15:42.41 | angryuser | State |
| 15:42.43 | Ecco | State -> Registered |
| 15:42.54 | angryuser | sip set debug on |
| 15:43.02 | Ecco | ok |
| 15:43.06 | angryuser | Call in and pastebing everything |
| 15:43.07 | Ecco | let me make a call to myself then |
| 15:43.17 | angryuser | pastebin* |
| 15:43.40 | *** join/#asterisk shaggy2 (~shaggy982@unaffiliated/shaggy2) |
| 15:43.48 | luckman212_ | is it true that app_fax and res_fax/res_fax_spandsp are mutually exclusive? |
| 15:43.51 | shaggy2 | anyone got any info on nortel phones, I can get it to connect to the PBX system using unistim.conf abd I can call the nortel phone, I can call other extensions, but I can not call an outside number, now I also have a cisco spa501g and I can do everything on that without a problem, any advice on the nortel??? |
| 15:44.06 | luckman212_ | i'm trying to compile * 1.8.3.2 with spandsp |
| 15:44.29 | angryuser | luckman212_, i think yes |
| 15:45.22 | luckman212_ | angryuser: when I compile with res_fax_spandsp, the module doesn't load right it errors out with "res_fax_spandsp.so: undefined symbol: t30_set_tx_page_header_info" |
| 15:45.49 | Ecco | angryuser: Here you go : http://pastie.org/1815829 |
| 15:45.53 | angryuser | luckman212_, you googled the error ? |
| 15:46.26 | angryuser | Ecco, what is your account number for that provider ? |
| 15:46.32 | luckman212_ | yep.. nothing |
| 15:46.34 | angryuser | Ecco, i mean it's a DID ? |
| 15:47.09 | Ecco | what's DID ? |
| 15:47.20 | angryuser | Ecco, Phone number |
| 15:47.27 | angryuser | Ecco, anyways 2 things |
| 15:47.39 | Ecco | Oh, yep, it's a "public" phone number :-) |
| 15:48.05 | angryuser | 1: register line lack this number at the end : register =>username:password@domain/NUMBER |
| 15:48.44 | Ecco | Oh, ok. Thing is, my SIP provider has a sample Asterisk configuration, and they ommited it. I did try adding the number, but it wouldn't fix the issue. I can add it back though |
| 15:48.50 | angryuser | 2: Your sip peer need to authorise insecure invites, option 'insecure=port,invite' |
| 15:49.05 | angryuser | unsecure* |
| 15:49.12 | angryuser | For the provider peers |
| 15:49.37 | Ecco | Should it be in "international" form, or the "local" form (i.e. 012345 or 3312345 (french phone numebr)) |
| 15:49.45 | Ecco | As for 2, I think I did it already actually |
| 15:49.46 | Ecco | " insecure=port,invite |
| 15:49.48 | angryuser | Ecco, and third |
| 15:50.16 | angryuser | add the number and reload, also show me the context where this peer is |
| 15:50.30 | angryuser | Ecco, in extensions.conf |
| 15:50.35 | Ecco | sure |
| 15:50.52 | jkprg | Hi. I'd like to configure conference calls using MeetMe in asterisk 1.6 for FreeBSD. ztdummy is not available. Would dahdi dummy driver be enough? Although I don't know if it's available for fbsd 8. Thx |
| 15:51.19 | angryuser | Ecco, if you have this unsecure option, you should not reply with "SIP/2.0 401 Unauthorized" to your provider |
| 15:51.59 | Ecco | http://pastie.org/1815851 |
| 15:53.32 | Ecco | angryuser : in the register line, the NUMBER should be in what format ? |
| 15:53.54 | angryuser | Ecco, username |
| 15:54.04 | Ecco | well, my username is not a phone number |
| 15:54.18 | Ecco | it's a textual login |
| 15:54.44 | *** join/#asterisk coppice (~chatzilla@220.232.166.62) |
| 15:55.02 | angryuser | Ecco, ok its not a problem, basicly it forms a uri location at your provider |
| 15:55.09 | Ecco | ok |
| 15:55.22 | angryuser | Ecco, you can type whatever you want, many provider allow that |
| 15:55.29 | angryuser | Show me context from_ippi |
| 15:55.42 | Ecco | oh, ok :-) |
| 15:56.55 | Ecco | http://pastie.org/1815868 |
| 15:57.14 | angryuser | Ecco, in short here is the deal asteris identify your provider and using 'ippi_outgoing' for that |
| 15:57.16 | angryuser | Found peer 'ippi_outgoing' for '33681904485' from 213.215.45.230:5060 |
| 15:57.24 | Ecco | Indeed |
| 15:58.13 | angryuser | In short |
| 15:58.32 | angryuser | move [ippi_incoming] upper in sip.conf than outgoing |
| 15:59.43 | Ecco | Huh... ok :-) |
| 15:59.50 | Ecco | It seems weird that order matters |
| 15:59.51 | Ecco | but ok :-) |
| 16:00.29 | angryuser | Ecco, well you have 2 peers with same provider IP |
| 16:00.40 | Ecco | holy shit |
| 16:00.41 | Ecco | it worked |
| 16:00.47 | Ecco | angryuser, you're a genius |
| 16:00.53 | Ecco | Now, why the fuck? :-) |
| 16:01.13 | angryuser | read the difference type=peer user of friend |
| 16:01.29 | angryuser | and about "asterisk peer mathing" |
| 16:01.34 | Ecco | Allright, thanks a lot |
| 16:01.47 | angryuser | matching |
| 16:01.55 | angryuser | De rien |
| 16:01.57 | Ecco | :-) |
| 16:02.12 | Ecco | May I ask another question, while I'm at it ? |
| 16:02.40 | angryuser | ~ask |
| 16:02.40 | 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. |
| 16:03.00 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 16:03.51 | Ecco | How should I do to let people "transfer" calls to each other ? |
| 16:04.05 | Ecco | I guess I should create more "friend" accounts |
| 16:04.22 | Chainsaw | And have them use "feature keys" to "transfer" their "calls". |
| 16:04.23 | angryuser | Ecco, option T and t in dial |
| 16:04.30 | Ecco | Oh, ok |
| 16:05.02 | Ecco | Thing is, when someone calls in on our "shared" number, I would like all "inside" accounts to ring |
| 16:05.03 | angryuser | Ecco, some phones use REFER, depends |
| 16:05.29 | Ecco | but once one of them has answered, it'd be cool if calls could be transfered from one phone to another |
| 16:05.46 | angryuser | Ecco, Dial(SIP/Account1&SIP/Account2) |
| 16:05.47 | Ecco | (as for "how" to do it on the client side, it's pretty easy, the softphone we use has a simple "transfer" function) |
| 16:05.53 | Ecco | Oh, awesome |
| 16:05.58 | angryuser | Ecco, read the dial options |
| 16:06.04 | Ecco | I will |
| 16:06.05 | Ecco | Thanks |
| 16:06.06 | angryuser | core show application Dial() |
| 16:06.45 | Ecco | As for the transfer per se, what 'number' should the calls be transfered to ? |
| 16:06.50 | angryuser | Ecco, and if you use some old phones connected to the FXS adaptors you need to setup features.conf |
| 16:07.14 | Ecco | Oh, ok. But it's only going to be SIP for now |
| 16:07.32 | angryuser | Ecco, so take some sofphone with sip transfer |
| 16:07.43 | angryuser | Ecco, there is nothing to setup |
| 16:07.49 | Ecco | Indeed |
| 16:07.58 | Ecco | Unfortunately I don't have choice for a softphone |
| 16:08.04 | Ecco | for the softphone |
| 16:08.43 | MrTelephone | What is the proper syntax for asterisk 1.8. exten => receive,n,Set(GLOBAL(FAXCOUNT)=[ ${GLOBAL(FAXCOUNT)} + 1 ] |
| 16:11.02 | seraphie | MrTelephone: Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ] |
| 16:11.12 | seraphie | [] and {} both preceeded by $ |
| 16:11.22 | MrTelephone | ok |
| 16:12.48 | atan | Ahh crap voipms server is giving me troubles with inbound DIDs : |
| 16:13.08 | atan | Anyone aware of a cheap inbound DID provider I could use maybe to replace them, and just use them for outbound calls? :D |
| 16:14.00 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 16:14.02 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:17.49 | *** part/#asterisk atan (~atan@unaffiliated/atan) |
| 16:26.38 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:28.08 | *** join/#asterisk [loy] (~nobody@95.72.237.59) |
| 16:28.08 | paulc | atan: link2voip have worked well for me (use paulc as a referral if you like) |
| 16:28.17 | paulc | DOH! laste.. |
| 16:28.20 | paulc | late even |
| 16:28.27 | paulc | goes off to wake up properly and learn how to read + type |
| 16:28.47 | Chainsaw | recommends coffee |
| 16:33.05 | Ecco | well angryuser , thank you very much for your help |
| 16:42.23 | *** join/#asterisk el3slave (~email@68.4.133.145) |
| 16:50.58 | *** part/#asterisk tasca (~tasca@189.34.27.64) |
| 17:00.15 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:00.30 | *** part/#asterisk clintc (~clintc@128.227.125.7) |
| 17:04.46 | MrTelephone | What is a fax for asterisk training error ? |
| 17:05.29 | Tozz_ | sounds to me like the Fax failed the handshake with the other end |
| 17:08.05 | coppice | MrTelephone: it means the image modem failed to train |
| 17:10.21 | *** join/#asterisk KarstenS (~KarstenS@p5B0C890B.dip0.t-ipconnect.de) |
| 17:12.23 | KarstenS | hi everybody :) has anybody time to help me out with a problem concerning gemeinschaft 3.1 and chan_capi? |
| 17:13.37 | *** join/#asterisk w4llnu55 (~w4llnu55@89.204.153.113) |
| 17:16.39 | KarstenS | nobody? :| |
| 17:16.51 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 17:31.28 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:36.25 | jaytee | how many RTP streams does a SIP call using dtmfmode=rfc2833? |
| 17:38.17 | KarstenS | hi everybody :) has anybody time to help me out with a problem concerning gemeinschaft 3.1 and chan_capi? |
| 17:39.41 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 17:43.25 | *** join/#asterisk w4llnu55 (~w4llnu55@82.113.99.3) |
| 17:43.48 | *** join/#asterisk tyrrexrrg (~roger@200.71.44.38) |
| 17:43.54 | pabelanger | Jasnejac: 2? |
| 17:44.01 | pabelanger | jaytee: ^ |
| 17:45.07 | Jasnejac | pabelanger: huh? lost connection earler. Tiz jasnejac |
| 17:45.23 | pabelanger | Jasnejac: wrong person |
| 17:48.26 | *** join/#asterisk JerJer (~jj@asterisk/original-h323-guy/JerJer) |
| 17:48.37 | JerJer | yawns |
| 17:48.43 | jaytee | pabelanger, I was thinking it was 4 with rfc2833 since it's using additional RTP streams for out-of-band dtmf signalling. |
| 17:49.43 | Ecco | Hi again |
| 17:49.51 | Ecco | Is it possible to define "alphanumeric" extensions in Asterisk ? |
| 17:49.58 | JerJer | Ecco: sure |
| 17:50.00 | Ecco | or does that just make no sense ? |
| 17:50.07 | JerJer | exten => bob,1,Answer |
| 17:50.11 | Ecco | Oh, ok |
| 17:50.43 | anonymouz666 | exten => jerjer,1,Hangup(H323) |
| 17:50.52 | Ecco | :-) |
| 17:50.55 | pabelanger | jaytee: Oh, in total, ya. I thought you wanted to know just for DTMF |
| 17:51.11 | Ecco | Any possibility to match this with a regexp then ? |
| 17:52.31 | Ecco | exten => [a-z]. doesn't seem to do the trick |
| 17:56.21 | Ecco | Arfg |
| 17:56.26 | Ecco | Forgot the underscore |
| 18:06.27 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v009-053.mobile.uci.edu) |
| 18:08.23 | *** join/#asterisk arielb27 (~chatzilla@unaffiliated/abatista) |
| 18:08.39 | therawr | interviewee is now rescheduled for 4-30-7pm |
| 18:08.42 | therawr | dies |
| 18:08.52 | therawr | whoops |
| 18:08.57 | therawr | i fail today |
| 18:08.59 | therawr | restarts |
| 18:15.52 | Ecco | Is it possible to get asterisk to route RTP traffic "peer-to-peer" ? |
| 18:15.59 | Ecco | People in my NAT cannot call each other :( |
| 18:17.17 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 18:19.03 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:19.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:21.43 | paulc | .c |
| 18:25.14 | *** join/#asterisk joshaidan (~brianj@host-216-26-197-55.tbaytel.net) |
| 18:27.34 | tzanger | file: foo. |
| 18:28.20 | leifmadsen | anyone run into an issue where polycom phones have a ringing sound in the background? |
| 18:28.32 | leifmadsen | it's like the call waiting tone is a ringing tone.... |
| 18:29.21 | _Corey_ | leifmadsen: It's a provisioning option on those phones to have ringing as a call waiting tone |
| 18:29.32 | tzanger | leifmadsen: wow, that's kind of fucked up |
| 18:29.37 | _Corey_ | normally you direct it to the handset |
| 18:29.55 | _Corey_ | so, instead of call waiting beep, the phone rings... we have a couple customers who prefer it |
| 18:30.02 | leifmadsen | _Corey_: ya... I can only guess this is what it is. I thought it might have had somethign to do with the call forwarding on the Bell line, but that is ported now, so it's directly SIP on the lines now |
| 18:30.08 | _Corey_ | if it's ringing in the handset I assume something's wrong |
| 18:30.14 | leifmadsen | _Corey_: ya I have a customer whose phone is doing it and hates it |
| 18:30.24 | leifmadsen | _Corey_: ya it's ringing in the handset |
| 18:30.41 | leifmadsen | I have no idea how it would do that as it's a stock polycom configuration deployment other than the registration data |
| 18:30.42 | _Corey_ | I wouldn't think that would be a popular "feature" |
| 18:31.03 | leifmadsen | the sip-interop.cfg file has: <call.callWaiting call.callWaiting.ring="beep"> |
| 18:31.14 | _Corey_ | yeah, that's the one you want |
| 18:31.36 | _Corey_ | you'd replace that with "ring" I think to make the base unit ring |
| 18:32.21 | _Corey_ | May be worth ringing Polycom on it |
| 18:32.41 | leifmadsen | I'm changing all the custom ringers from type="ring" to type="beep" |
| 18:32.45 | leifmadsen | then I'll reboot and see what happens |
| 18:32.50 | leifmadsen | otherwise I might ahve to check with polycom... |
| 18:33.50 | Ecco | Is there a "catch-all" extension in Asterisk ? |
| 18:34.28 | _Corey_ | yeah, in that CALLPROGRESS seciton is where it can also be messed with |
| 18:34.48 | Ecco | Hmm, my question wasn't clear |
| 18:34.57 | leifmadsen | Ecco: not really, no |
| 18:34.57 | Ecco | I meant, when exten => name,priority,application() |
| 18:34.59 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:35.06 | Ecco | is it possible to catch all in "name" |
| 18:35.13 | leifmadsen | literally "name" ? |
| 18:35.24 | Ecco | well |
| 18:35.30 | Ecco | I would like to catch any incoming name |
| 18:35.33 | Ecco | like 0123 |
| 18:35.35 | Ecco | or "foobar" |
| 18:35.36 | Ecco | anything :-) |
| 18:35.50 | leifmadsen | exten => _[A-Za-z0-9].,1,Set(IncomingExtension=${EXTEN}) |
| 18:35.55 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 18:36.12 | Ecco | meh |
| 18:36.15 | Ecco | indeed |
| 18:36.16 | leifmadsen | matches any 2 digit or more extension starting with alphanumeric |
| 18:36.20 | leifmadsen | you can do _. |
| 18:36.23 | Ecco | Oh, ok |
| 18:36.24 | leifmadsen | but it's not recommended |
| 18:36.28 | Ecco | Why ? |
| 18:36.44 | KarstenS | Hi everybody. How can i check if chan_capi recognizes incomming calls.. in the CLI (asterisk -rvvv) is no output when a call comes in. :( |
| 18:37.04 | leifmadsen | because it will match any character at all -- see the README-SERIOUSLY.bestpractices.txt file in your asterisk source |
| 18:37.11 | leifmadsen | Ecco: ^^^ |
| 18:37.21 | Ecco | Thanks leifmadsen |
| 18:37.34 | leifmadsen | KarstenS: all you can do is check debugging to see if it gives you anything -- if not, then it's probably not working |
| 18:37.53 | KarstenS | capi debug on? |
| 18:38.25 | leifmadsen | no idea |
| 18:38.30 | leifmadsen | whatever the debug command is for capi |
| 18:38.33 | leifmadsen | I've never used it |
| 18:39.23 | KarstenS | its "capi debug".. > CAPI Message Debugging Enabled |
| 18:39.52 | KarstenS | but no debug info when the call comes in :( |
| 18:41.18 | KarstenS | how can i check if the chan_capi module is loaded? if it works with outgoing calls.. does that mean the chan_capi must work? |
| 18:41.19 | leifmadsen | then it sounds like a configuration issue |
| 18:41.25 | leifmadsen | module show like capi |
| 18:41.49 | *** join/#asterisk [T]ank1 (~Tank@c-76-27-9-24.hsd1.ut.comcast.net) |
| 18:42.32 | [T]ank1 | anyone here that is familiar with both Avaya session manager and asterisk that would be looking to make a few bucks doing some quick support? |
| 18:43.02 | KarstenS | ah, ok :) 1 module loaded.. |
| 18:43.15 | KarstenS | Use Count: 0 |
| 18:43.21 | KarstenS | strange.. |
| 18:44.21 | KarstenS | does that mean that i dont use chan_capi for outgoing calls?! mh.. |
| 18:46.33 | leifmadsen | Use Count just means it is not actively being used by anything |
| 18:46.41 | leifmadsen | I usually just ignore that part |
| 18:46.45 | KarstenS | ah, ok |
| 18:46.53 | *** join/#asterisk luckman212 (~irc@pool-74-108-1-53.nycmny.fios.verizon.net) |
| 18:48.44 | luckman212 | is http://downloads.digium.com down? |
| 18:48.59 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-qviqsyktbxyghxkn) |
| 18:49.57 | Chainsaw | luckman212: It seems to be. Neither downloads.digium.com nor downloads.asterisk.org is willing to load here. |
| 18:50.04 | *** join/#asterisk hugogee (~hugogee@desert-cpe-31-117.flashbyte.us) |
| 18:50.08 | luckman212 | arg.. need to download something too |
| 18:50.13 | hugogee | greetz all. |
| 18:50.17 | Chainsaw | luckman212: We might have the file you are after on the Gentoo mirrors. Can I ask which one it is? |
| 18:51.29 | pabelanger | Chainsaw: Ack, somebody is working on it |
| 18:51.30 | luckman212 | http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.4.1.2+2.4.1.tar.gz |
| 18:51.46 | hugogee | Asterisk + PF. inbound sound only. Does anyone have |
| 18:51.55 | hugogee | any pf rules they can share. |
| 18:51.56 | pabelanger | Chainsaw: try now |
| 18:52.21 | Chainsaw | luckman212: I have http://distfiles.gentoo.org/distfiles/dahdi-linux-2.4.1.tar.gz if that helps. |
| 18:52.32 | Chainsaw | pabelanger: Much better, thank you. |
| 18:52.37 | Chainsaw | luckman212: But cancel that, pabelanger's saved the day. |
| 18:53.36 | luckman212 | :-) |
| 18:59.48 | *** join/#asterisk Micc (~Micc@98.232.41.66) |
| 19:01.04 | Micc | I've been getting my asterisk hanging the last two days. Running 1.6.2.17.2. Seems to happen at peak times about 30-40 active calls, lots of blf activity. And when asterisk restarts I see an error about no more UDPTL sockets. |
| 19:03.06 | Micc | the exact error is WARNING[28745] udptl.c: No UDPTL ports remaining |
| 19:03.50 | leifmadsen | Micc: check the udptl.conf file, and you might need to disable t38 (unless you're using it of course). Or perhaps it could be the ulimit size is too small. |
| 19:05.30 | leifmadsen | otherwise you can check the bug tracker for open issues regarding that, or use google |
| 19:14.45 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 19:15.59 | psilikon | hugogee, Pfsense or you just have an openbsd firewall? |
| 19:16.43 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:17.12 | Micc | yeah we are using t38 |
| 19:17.21 | cusco_ | hi... |
| 19:18.03 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 19:18.04 | beek | my "Asterisk: Definitive Guide" has been shipped from Amazon. Kudos leifmadsen et al |
| 19:18.49 | cusco_ | peer 150 calls some dialplan that has Queue(bla) - peer 200 answers blabla and hangs up. in the 'h' extension, from peer 150 how can I get to know that peer 200 answered the call? |
| 19:18.56 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 19:19.50 | leifmadsen | beek: YAY!!!! I just saw someone post that theirs shipped on facebook |
| 19:20.06 | leifmadsen | cusco_: you can't |
| 19:20.26 | leifmadsen | cusco_: unless there is a channel variable that gets set -- check DumpChan() to see what gets set on the channel |
| 19:20.31 | leifmadsen | exten => h,1,DumpChan() |
| 19:21.01 | cusco_ | cdr has dstchannel |
| 19:21.17 | cusco_ | but... |
| 19:21.59 | *** join/#asterisk phyburn (~phyburn@wsip-70-165-35-234.oc.oc.cox.net) |
| 19:22.21 | cusco_ | what if peer 200 in queue(bla) has interface set to Local/200@context and in dialplan _XXX I set a variable... this is a different channel from the one that originally dialed... |
| 19:22.49 | cusco_ | can I get that set variable back in the first and lastting channel= |
| 19:22.51 | cusco_ | ? |
| 19:23.21 | *** join/#asterisk Suregard (~Suregard@41-134-26-122.dsl.mweb.co.za) |
| 19:23.36 | Suregard | Evening |
| 19:26.16 | Micc | leifmadsen, is there something I can turn on in build like dont optimize to lessen the frequency of hangs? Or what that increase the frequency? |
| 19:26.22 | leifmadsen | shrugs |
| 19:26.26 | Micc | leifmadsen, seems to me like some kind of lock race condition. |
| 19:26.37 | leifmadsen | could be, in which case you needs to do 'core show locks' |
| 19:26.48 | Micc | oh |
| 19:26.59 | Micc | so when I see it happen that may give helpful information? |
| 19:27.10 | cusco_ | no such command |
| 19:27.10 | leifmadsen | for a developer sure |
| 19:27.14 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 19:27.26 | leifmadsen | cusco_: you need to enable that command |
| 19:27.34 | leifmadsen | DEBUG_THREADS in menuselect |
| 19:27.41 | Micc | core show locks says invalid command |
| 19:27.46 | Micc | oh |
| 19:27.50 | Micc | got it. |
| 19:27.50 | leifmadsen | it's not enabled by default |
| 19:28.06 | cusco_ | ok |
| 19:28.17 | cusco_ | he got fad up with our boring questions :p |
| 19:28.58 | Aut0ExeC | yeah he doesnt have time for our lame questions |
| 19:29.00 | Aut0ExeC | :| |
| 19:30.09 | cusco_ | Micc: what udp port rante have you defined in sip.conf ? |
| 19:30.26 | Micc | 10k-20k I think |
| 19:30.42 | cusco_ | I have in my sip.conf: t38pt_udptl=t38UDPFEC ; fax |
| 19:30.48 | cusco_ | don't remember what it is tho |
| 19:31.14 | cusco_ | also udptl.conf has some stuff |
| 19:31.27 | cusco_ | like: |
| 19:31.28 | cusco_ | /etc/asterisk/udptl.conf:udptlstart=4000 |
| 19:31.28 | cusco_ | /etc/asterisk/udptl.conf:udptlend=4999 |
| 19:31.59 | Micc | 6000-9999 |
| 19:33.02 | cusco_ | also have udptlfecentries = 3 and udptlfecspan = 3 |
| 19:33.05 | cusco_ | dunno what for tho |
| 19:33.22 | Micc | I wonder if I need a larger range of ports |
| 19:33.41 | cusco_ | doubt it |
| 19:34.25 | Aut0ExeC | is there such thing as a aix2 phone that can better handle remote connections to asterisk? |
| 19:34.39 | Aut0ExeC | handles nat better |
| 19:34.43 | Aut0ExeC | no need to open ports |
| 19:35.26 | cusco_ | Micc: can check file descriptors usage |
| 19:35.30 | keith4 | there are IAX softphones |
| 19:35.32 | cusco_ | lsof -n -p $PID |
| 19:35.39 | cusco_ | or ls -lia /proc/$PID/fd/ |
| 19:35.50 | cusco_ | where $PID is the pid of asterisk |
| 19:35.57 | keith4 | don't think i've even seen a hard phone that supported IAX, though |
| 19:36.24 | cusco_ | keith4: Im sure there are some |
| 19:36.31 | Aut0ExeC | keith4: ok thanks |
| 19:36.43 | keith4 | there might be. i'd love to try one |
| 19:36.47 | keith4 | if anyone finds one, let me know |
| 19:36.50 | cusco_ | http://www.voip-info.org/wiki/view/Asterisk+IAX+clients |
| 19:36.51 | Aut0ExeC | lol |
| 19:37.06 | cusco_ | I probably have one lying arround but we use sip anyway |
| 19:37.30 | Aut0ExeC | cusco_: how does that handle nat? |
| 19:37.37 | Aut0ExeC | without opening ports? |
| 19:37.43 | keith4 | i had to use some IAX soft phone last time I was out of the country, because I was double-NAT'd, or some crazy shit, where we were staying. couldn't get SIP/RTSP to cooperate |
| 19:37.58 | keith4 | Aut0ExeC: it uses a single port, instead of ~10k |
| 19:38.01 | cusco_ | dunno its internal |
| 19:38.05 | cusco_ | really dunno |
| 19:38.08 | Aut0ExeC | ohh ok |
| 19:38.09 | cusco_ | yes |
| 19:38.11 | cusco_ | that makes sense |
| 19:38.20 | Aut0ExeC | nice |
| 19:38.26 | Aut0ExeC | still somewhat open tho |
| 19:38.29 | keith4 | so it plays better with stateful firewalls, for example |
| 19:38.38 | Aut0ExeC | k |
| 19:39.09 | keith4 | I think I used Zoiper. it kinda sucked, iirc, but it got the job done |
| 19:39.10 | Micc | cusco_, looks like 39 udp ports. funny thing is I have no t38 going on right now but see a lot of ports within my udptl rang |
| 19:39.25 | cusco_ | Micc: sip also uses udp |
| 19:39.29 | Micc | and whenever a normal call starts it says something about UDPTL cos mark |
| 19:39.49 | Micc | cusco_, yes but should be using different ports shouldn't it? |
| 19:40.07 | cusco_ | Micc: should be using those 10k to 20k as you said |
| 19:40.21 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 19:41.16 | cusco_ | Micc: any of them 6000-9999 ? |
| 19:41.50 | ssureshot | afternoon,, anyone around that can give me a hand with cisco 7940 ? |
| 19:42.35 | ssureshot | my issue is this... when the phone boots up it flashes File not found SEP(IP address) then reboots |
| 19:42.39 | cusco_ | ssureshot: really depends on the problem. Don't ask to ask, just ask and maybe you'll get an answer :p |
| 19:43.04 | ssureshot | the configuration works on another 7940 so I'm not sure whats happening |
| 19:43.19 | ssureshot | cusco_: yeah I know that too lol.. my bad |
| 19:43.39 | cusco_ | is that regarding tftp autoconf? |
| 19:44.33 | ssureshot | cusco_ yes it is tftp autoconf and the error is File not found : SEP(Mac) |
| 19:44.49 | ssureshot | hmm wait a second,, Ikm |
| 19:44.51 | cusco_ | I never used that... anything in log files? |
| 19:44.57 | cusco_ | (asterisk) ? |
| 19:45.10 | cusco_ | perhaps tftp connection unsucessfull or wrong path |
| 19:45.22 | cusco_ | tftp server |
| 19:45.45 | ssureshot | I'm using a SIP file but isn't there aSEP file also in xml.. .. log files are virtually empty asside from the dhcp logs |
| 19:46.34 | cusco_ | im really not into that... |
| 19:47.00 | ssureshot | the tftp server works I have 75 other phones on it ,, this is a new phone I'm trying to configure and it's being difficult |
| 19:47.02 | cusco_ | and next I would recon some tool like tcpdump to see whats hapenning :p |
| 19:47.11 | keith4 | you need tftp logs |
| 19:48.18 | Micc | cusco_, yes almost half of them are 6000-9999 |
| 19:48.42 | ssureshot | hmm,, fedora core 2 tftp logs,, < scratches head googles |
| 19:48.52 | ssureshot | and yes it's asterisk |
| 19:48.53 | keith4 | which tftp daemon? |
| 19:48.56 | Micc | maybe more like a third or quarter of them are in the udptl range. |
| 19:49.12 | Micc | but it still seems like a lot to me when it shouldn't be doing an t38 |
| 19:49.29 | *** part/#asterisk Suregard (~Suregard@41-134-26-122.dsl.mweb.co.za) |
| 19:49.44 | Micc | every single call shows Using UDPTL TOS bits 160, Using UDPTL CoS mark 6 |
| 19:49.56 | Micc | which could be just informational, not saying its using a UDTPL port. |
| 19:50.14 | keith4 | ssureshot: if it's tftp-hpa, it's probably running under [x]inetd... so you might just have to add a logging flag |
| 19:50.27 | Micc | I think I reported this once before and they said it was normal. |
| 19:50.37 | Micc | but it still doesn't seem right to me. |
| 19:50.50 | ssureshot | keith4: roger that.. Ill take a look thank you |
| 19:52.46 | cusco_ | Micc: are you hanging up faxes upon transmission/receival |
| 19:52.47 | cusco_ | ? |
| 19:53.36 | cusco_ | Micc: TOS and CoS are for QoS. I think that is only informational |
| 19:53.58 | cusco_ | that is so you can set QoS on your router for tos 160 to have higher priority |
| 19:56.22 | Micc | cusco_, most faxes work fine and hangup when they are done. |
| 19:57.08 | cusco_ | and is 'core show channels concise' showing any faxes? |
| 19:57.28 | Micc | cusco_, another funny thing I just noticed, netstat -n showed a bunch of connections in TIME_WAIT that were connecting to port 80 on a bunch of different servers. I'm not sure what process would be doing that. Does centos have something that would do that for updates? |
| 19:57.41 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 19:57.55 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 19:58.01 | beek | Micc: what does netstat -nlp show? |
| 19:58.48 | cusco_ | I dunno centos but try netstat -tapn |
| 20:00.49 | Micc | they aren't there anymore. |
| 20:00.59 | Micc | what is pluto? |
| 20:01.29 | Micc | looks like ipsec related |
| 20:02.07 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 20:02.30 | phyburn | is there anything out there that can help you figure out average number of concurrent calls? |
| 20:02.39 | *** join/#asterisk jkenney_ (~jkenney@mail2.helpdeskandsupport.com) |
| 20:04.08 | keith4 | phyburn: install the SNMP agent and trend/graph the channel data using mrtg/cacti/whatever |
| 20:04.44 | Micc | there were about 10 different IP addresses with 2 port 80 connections each |
| 20:05.16 | drfreeze | I have a client using zoiper over OpenVPN. They are use a Logitech headset or a Asus travelite hs-1000W headset |
| 20:05.42 | drfreeze | They don't hear the line crackling, but the other end does |
| 20:06.23 | drfreeze | I'm reading that maybe the problem is the sound codec can't deal with the line problems |
| 20:06.34 | drfreeze | Does anyone have some suggestions on how to resolve this issue? |
| 20:07.11 | drfreeze | The crackling is sporatic - happens about every 5-10s, duration 200-2000ms and varying noise levels |
| 20:12.30 | pabelanger | drfreeze: try a different codec, see what happens |
| 20:14.53 | keith4 | phyburn: or... use a CDR analysis tool |
| 20:16.15 | jkprg | Hi. Is it possible to change timeout for extension during blind xfer? thx |
| 20:22.15 | Micc | they were all mirros. |
| 20:22.24 | Micc | for yum packages I think. |
| 20:25.11 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 20:33.13 | drfreeze | pabelanger: tried gsm, not much improvement |
| 20:33.38 | drfreeze | was thinking of trying g729. I saw a free version of this codec. Is that legit? |
| 20:33.48 | JonathanRose | no |
| 20:34.08 | drfreeze | anyone know how got get a legit version of g729? |
| 20:34.33 | Freeaqingme| | buy a license drfreeze ? |
| 20:34.46 | JonathanRose | Sipro |
| 20:34.52 | JonathanRose | They are the ones who sell it I think. |
| 20:35.10 | Freeaqingme| | doesn't digium sell it themselves too? |
| 20:35.15 | Freeaqingme| | or at least sublicense |
| 20:35.17 | drfreeze | http://store.digium.com/productview.php?product_code=G729CODEC |
| 20:35.27 | JonathanRose | Yeah |
| 20:35.30 | JonathanRose | I should have known that... |
| 20:35.58 | JonathanRose | It can get expensive though. |
| 20:36.11 | JonathanRose | I mean, you have to buy a license for every channel you use that does G729 |
| 20:36.19 | drfreeze | I only need it for on line/channel |
| 20:36.22 | JonathanRose | Well that's good. |
| 20:36.24 | drfreeze | looks like $10 per channel |
| 20:36.30 | drfreeze | not sure if there is recurring fees or not |
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| 21:05.42 | drfreeze | FYI - the $10 license is valid until 2026 |
| 21:05.51 | drfreeze | so, that is really cheap, IMO |
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| 21:17.46 | _Corey_ | be careful not to make too many system changes where you guys activate those g729 licenses |
| 21:18.06 | _Corey_ | we added some NICs recently at a customer and realized that we had to re-register the licenses |
| 21:18.28 | _Corey_ | there ARE a finite number of activations before the license blows up |
| 21:18.42 | p3nguin | Then you have to pay a fee to Digium to fix it. Big deal. |
| 21:18.53 | _Corey_ | theoretically :) |
| 21:19.27 | p3nguin | The NICs messed up your licenses because the registration key is built around the MAC addresses of the host. |
| 21:19.53 | p3nguin | Change everything else if you want, but don't change the NICs. |
| 21:19.59 | _Corey_ | Yeah, it wasn't a mystery... I knew immediately what had happened. The guy who unlocked it at Digium for me warned me they could only do it one more time |
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| 21:22.26 | cusco_ | can't you just set the mac address to clone the old one? |
| 21:22.38 | cusco_ | that should go round that problem |
| 21:22.50 | drfreeze | my bad, it expires 2031 |
| 21:22.53 | cusco_ | (dunno I don't have g729 licences) |
| 21:23.32 | cusco_ | drfreeze: free implementation of g729 is legit in europe |
| 21:24.13 | cusco_ | that may depend on the law of where you live |
| 21:24.34 | drfreeze | at $10 per line, that is pretty inexpensive |
| 21:25.20 | p3nguin | Not if you need several hundred or thousands of them. |
| 21:25.34 | p3nguin | Then it's very expensive. |
| 21:26.03 | cusco_ | hehe |
| 21:26.30 | _Corey_ | yeah, the customer I was working with has 1,000 channels so if I tell them they have to re-purchase because someone monkeys with the NIC it probably won't go over so well |
| 21:26.44 | _Corey_ | :) |
| 21:28.45 | _Corey_ | p3nguin: Did you ever get anywhere with a working skinny channel on 1.8? |
| 21:31.02 | p3nguin | Nope. It's junk and there's apparently no support for it. |
| 21:31.15 | p3nguin | My only choice was to keep using 1.4 and chan_sccp-b. |
| 21:31.35 | _Corey_ | too bad |
| 21:31.39 | p3nguin | very |
| 21:31.44 | p3nguin | I'm interested in 1.8. |
| 21:32.10 | _Corey_ | (i had someone who's already on 1.8 ask me about it today) |
| 21:32.19 | p3nguin | But I have to have SCCP, and there's a disconnect between those mechanisms. |
| 21:33.25 | _Corey_ | yeah, it seems like it's not going anywhere unless another developer adopts it |
| 21:34.22 | p3nguin | I don't see why it can't receive the same attention as any other channel driver. |
| 21:34.48 | p3nguin | It's certainly not any less important of a component. |
| 21:34.59 | _Corey_ | Just a market thing... not enough people who are interested in using Cisco phones w/Asterisk |
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| 21:43.51 | beardy | Why do you have to? |
| 21:44.26 | beardy | Which model? No sip image? |
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| 21:44.58 | _Corey_ | beardy: Cisco SIP software isn't as feature-rich |
| 21:45.33 | beardy | _Corey_: Tell me which feature you're using that's missing. |
| 21:45.57 | p3nguin | Besides that, just assume SCCP is the only protocol my phones operate on. Not everyone can jump at the suggestion of "just use SIP and be happy about it." |
| 21:47.07 | p3nguin | If I was going to use SIP, I wouldn't care about having an SCCP channel driver working right. |
| 21:48.20 | _Corey_ | I use a Polycom phone so I can only share a couple things I know for sure... We have people complain that the call-transfer options are more limited on the newer SIP firmware, call forwarding seems to be limited to 4-digits, etc |
| 21:48.55 | _Corey_ | I would just characterize the SIP load as "basic" |
| 21:49.06 | _Corey_ | It was better on the older hardware (7940/7960) |
| 21:49.28 | _Corey_ | You could throw a logo on there and mesmerize people enough to overlook the other stuff typically |
| 21:49.35 | _Corey_ | not so much on the newer sets though |
| 21:49.42 | beardy | I see. |
| 21:50.22 | beardy | I don't have newer sets, I didn't know they were crapped down. That angers me. |
| 21:50.40 | _Corey_ | eh, Cisco doesn't care about writing a good SIP firmware for those phones |
| 21:51.04 | _Corey_ | They bought Sipura and pretty much figure if you're going to use their stuff with non-Call Manager you should buy those SPA-xxxx phones |
| 21:53.15 | beardy | I don't see any "Sipura" in the lastlog. |
| 21:56.10 | _Corey_ | huh? |
| 21:59.52 | jkenney_ | Good Evening is anyone using the soundpoint 550 with the new 3.1.1 3.3.1.0933 |
| 22:01.05 | jkenney_ | sip revision |
| 22:11.04 | jkprg | Hi. Can Asterisk 1.6 provide Consultation Hold? Thx |
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| 22:36.14 | Micc | jkprg, what is consultation hold? |
| 22:36.38 | Micc | jkenney_, I've used 3.3.1f |
| 22:36.45 | *** part/#asterisk KarstenS (~KarstenS@p5B0C890B.dip0.t-ipconnect.de) |
| 22:37.10 | jkprg | Micc: http://www.voip-info.org/wiki/view/PBX+Consultation+Hold |
| 22:37.42 | jkprg | Micc: for SIP users in my case |
| 22:38.21 | _Corey_ | jkprg: So, the idea is while a call is ringing that you can put it on hold w/o answering it? |
| 22:39.32 | jkprg | _Corey_: while conversation is establish I can put peer on hold make another call and get back to the original peer |
| 22:40.41 | _Corey_ | Not totally clear, you seem to be describing a normal hold/pickup scenario |
| 22:41.12 | _Corey_ | So, you want to put a caller on hold... get another line, make a call... and then pick the original caller up? |
| 22:41.46 | jkprg | _Corey_: yes. |
| 22:41.55 | _Corey_ | um, ok... any phone can do that |
| 22:42.44 | jkprg | _Corey_: are you sure? |
| 22:42.54 | _Corey_ | yeah |
| 22:43.04 | _Corey_ | try for yourself |
| 22:43.21 | jkprg | _Corey_: I'd like but I don't know how :-) |
| 22:44.09 | _Corey_ | Consult your phone's manual |
| 22:44.53 | jkprg | _Corey_: phone is saying "Only one simultaneous call is allowed" |
| 22:45.37 | jkprg | _Corey_: That's why I was thinking to simulate it using asterisk |
| 22:46.18 | jkprg | _Corey_: So far it looks asterisk doesn't support it :-( |
| 22:46.57 | _Corey_ | Not sure I can help you |
| 22:48.21 | *** join/#asterisk fofware (~Fabian@host5.186-124-226.telecom.net.ar) |
| 22:48.47 | jkprg | anyway thx |
| 22:49.11 | [T]ank1 | how do i set anything to do with max-forward headers? |
| 22:49.23 | [T]ank1 | im getting an invalid max-forwards header error 400 |
| 22:50.27 | fofware | hello, I have a problem with TDM410p settings, when I pickup analog phone connected to fxs port when i press any key hanging start, any idea where is my mistake? |
| 22:58.51 | fofware | so any idea :( |
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