| 00:04.05 | ujjain | Hi, are grandsteam and linksys ata adapters same quality? |
| 00:04.15 | ujjain | Or is one brand better? It's for my parents |
| 00:05.06 | ujjain | and what would be the most popular cheap adapter with 1x FXO, 1x FXS |
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| 00:09.35 | WIMPy | Some ppl say that |
| 00:09.39 | WIMPy | ~gs |
| 00:09.39 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
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| 04:51.53 | Torrieri | Hi |
| 04:57.00 | Torrieri | i got problems with my queues, when a member is in a call if a caller joined in the queue the callee received a beeping like a call waiting beep, how can override the beeping? |
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| 05:09.52 | ChannelZ | Torrieri: you have to turn off call waiting on the device |
| 05:11.50 | Torrieri | ChannelZ, im looking in the gateway options |
| 05:11.50 | Torrieri | ty |
| 05:12.20 | ChannelZ | or investigate 'ringinuse' and getting device states working for your SIP devices |
| 05:12.58 | ChannelZ | but basically if the phone is able to accept more than one call which most do, * tries to send a call to the agent not really knowing they're already on the phone, and their phone accepts the call, so... |
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| 05:20.22 | Torrieri | ChannelZ, ima turn off call waiting on the device for a while, ill investigate more. thanks u |
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| 05:59.30 | ChannelZ | have fun |
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| 06:18.39 | mickecarlsson | WIMPY ping I am here now |
| 06:19.12 | mickecarlsson | WIMPy ping |
| 06:20.52 | ChannelZ | he'll never hear you with a wimpy ping like that.. |
| 06:21.00 | mickecarlsson | LOL |
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| 06:28.25 | schmidts | good morning |
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| 06:37.51 | wdoekes2 | good morning |
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| 06:44.47 | kleszcz | morning |
| 06:44.53 | wdoekes2 | at those wondering about my /local libspandsp problem: it was probably a lack of ldconfig after all.. after this nightly restart it uses the right lib |
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| 07:25.15 | sigmounte | hello , i'm using asterisk 1.8 from scrath , and i have a problem with moh and prompt ( moh work in Dial , but not in queue , and somes prompts like thoses in voicemail can be hear but not the ones in queues ) where can i search for problem ? thanks |
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| 07:35.13 | sigmounte | any others asterisk channel available on this irc server ? |
| 07:40.01 | acidjazz | hey all.. so now that android 2.3.4 has wireless calls/etc .. can i setup a sip account and like use/forward my cell # to make/recieve calls? |
| 07:41.32 | Marvelous | ask indorid |
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| 08:25.30 | porche | Hi |
| 08:25.36 | porche | any 1 from Turkey? |
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| 08:35.00 | GreatSUN | no |
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| 09:10.23 | Lipsum | When using CiscoIPPhoneInput on a SPA504G, the phone automatically translates some words from english to the phone's current langauge selection. Anyway to disable this? Since the current input field is for searching? |
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| 09:31.39 | puzzled | hi |
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| 09:36.22 | irooot | greets peeps thought id finally drop in |
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| 09:59.55 | the_5th_wheel | Good day. I have a setup with two asterisk server, and various sip peers which register to these peers. I have a trunk between the two. I need the ANI to propogate over the link. When I route a call over it, it drops the call with a 404, and the sip debug says no matching peer found (and shows the ANI as the peer that wasnt found). Can anyone nudge me in the right direction? The sip peer basically has the ip specified, codecs,context,tos |
| 10:02.27 | kaldemar | the_5th_wheel: you're missing extensions in extensions.conf |
| 10:07.08 | the_5th_wheel | kaldemar: Its failing back to the default context instead of the context specified in the peer |
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| 10:08.49 | kaldemar | the_5th_wheel: that's because you don't have a matching extension in the defined context |
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| 10:15.19 | the_5th_wheel | kaldemar: no, I have the extention in the defined context. I even have an _X. in that context. As I said, it seems to be looking for a different peername |
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| 10:26.23 | irroot | what is in the sip to header ?? |
| 10:27.11 | irroot | Little hack i use Goto(<CONTEXT>,${CUT(SIP_HEADER(to),@,1):5},1) |
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| 10:33.31 | the_5th_wheel | irroot: http://pastebin.com/wW5auNy2 A sanitized Sip debug. Ive Xed out the last three digits of the numbers and the middle two octets of the IPs |
| 10:35.54 | irroot | take it you close here somewhere MTN no ... |
| 10:36.51 | kaldemar | the_5th_wheel: define a peer that matches by ip address of the other end. type=peer and host=x.x.x.x instead of host=dynamic. |
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| 10:40.14 | irroot | use a "guest" user to accept the call |
| 10:41.57 | the_5th_wheel | kaldemar: I have no host=dynamic specified. Both siews have the IPs specified |
| 10:42.27 | irroot | [guest] |
| 10:42.30 | irroot | type=user |
| 10:42.31 | irroot | context=ddi |
| 10:42.33 | the_5th_wheel | ironm: If I use the guest account, anyone will be able to route out via the server? |
| 10:42.39 | irroot | its a dustbin of sorts |
| 10:42.48 | the_5th_wheel | sorry, not irnm, irroot |
| 10:42.49 | irroot | ddi is a open inbound context |
| 10:42.59 | the_5th_wheel | yes, The problem is these are outbound calls |
| 10:43.08 | kaldemar | type=user is what you do NOT want. |
| 10:43.16 | irroot | ok with you |
| 10:43.28 | irroot | thought the call was already out |
| 10:43.46 | kaldemar | take a look at "naming devices" here: http://svn.digium.com/svn/asterisk/tags/1.8.4/configs/sip.conf.sample |
| 10:45.43 | irroot | so you dialing ...,Dial(SIP/peername/number) |
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| 10:46.55 | the_5th_wheel | irroot: yes |
| 10:47.01 | the_5th_wheel | here is is the sip configs |
| 10:47.02 | joobie | mary mary |
| 10:47.03 | the_5th_wheel | http://pastebin.com/DhQh1mC8 |
| 10:47.05 | joobie | ur balls are hairy |
| 10:48.00 | irroot | why not Dial(SIP/<number>@<ip_addr> |
| 10:48.16 | irroot | keep it simple if you want to get it to the other side |
| 10:49.24 | joobie | sip sip s ip |
| 10:49.31 | joobie | KISS |
| 10:49.37 | joobie | Keep it simple stupid |
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| 10:52.07 | kaldemar | the_5th_wheel: remove "port=5060" |
| 10:53.01 | kaldemar | the_5th_wheel: also, insecure=very is invalid unless you're using an old version. it has been replaced by insecure=port,invite. |
| 10:53.21 | joobie | fuk |
| 10:53.22 | kaldemar | however, there's no reason for you to not authenticate calls since both boxes are controlled by you. |
| 10:53.24 | joobie | im still running 1.4 |
| 10:53.38 | joobie | is 1.8 stable now? |
| 10:54.00 | the_5th_wheel | irroot: no difference |
| 10:54.06 | kaldemar | i refrain from giving an opinion on 1.8 stability. |
| 10:54.45 | the_5th_wheel | kaldemar: I just added that in a bit before I came here to test. it made no difference. It still seems to look for a peername which is the ANI |
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| 10:57.15 | ironm | hi the_5th_wheel .. sorry .. I can't follow you |
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| 10:58.13 | atan2 | If one was setting up a small inbound call centre how would they best handle call attended call transfers/blind transfers? Say you have a customer on the phone and need to call your supervisor or something. Would you set something up server-side to take control of the active call by putting it on hold and connecting a new call to the super? |
| 10:58.31 | atan2 | Or would you just let the agent have 3 or 4 lines they could use... hmm. |
| 10:59.10 | the_5th_wheel | ironm: sorry, I autocorrected wrong, was aimed at irroot |
| 10:59.19 | atan2 | Can even you take control of a call with Asterisk once it is connected? |
| 11:01.20 | irroot | dialing SIP/num@ip will not need any peer or other settings should get to the other side and you should be able to debug it there also watch the dialplan not only sip packets set verbose 3 |
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| 11:13.45 | kaldemar | the_5th_wheel: did you remove the port already? |
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| 11:50.41 | Lipsum | When using CiscoIPPhoneInput on a SPA504G, the phone automatically translates some words from english to the phone's current langauge selection. Anyway to disable this? Since the current input field is for searching? |
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| 12:10.26 | irroot | ok with SRTP there is 80bit and 32bit |
| 12:10.59 | irroot | im going to be reworking encryption=yes/no to be yes/no/32bit |
| 12:11.47 | irroot | my idea of making encryption a global option was not a good one ... |
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| 12:22.53 | mikeh | Hi guys, I'm trying to figure out my dial patterns - is there any way I can say where a number _doesn't_ begin with a certain number? |
| 12:22.58 | mikeh | ie. currently the dial pattern is just . |
| 12:23.07 | mikeh | but I want it to exclude numbers beginning with 8 |
| 12:24.17 | kaldemar | mikeh: _[1-79] |
| 12:24.55 | kaldemar | usage of "." is not recommended because it matches the special extensions. |
| 12:25.44 | kaldemar | my example does not include 0, you can add it too if needed. |
| 12:26.15 | *** join/#asterisk dobby156 (~joe@79.135.102.10) |
| 12:26.46 | dobby156 | hi, I am writing a program that converses with asterisk via the AMI and I am having a problem |
| 12:27.24 | dobby156 | when I send the request like "Action: hangup\nChannel: whatever\n\r\n\r" |
| 12:27.44 | dobby156 | i get missing action in request |
| 12:27.51 | dobby156 | everytime |
| 12:27.55 | *** join/#asterisk zkn (~zkn@217-159-228-106-dsl.est.estpak.ee) |
| 12:28.16 | dobby156 | if I log into the same account via a telnet agen the same command works fine |
| 12:28.23 | dobby156 | any ideas, thanks? |
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| 12:35.44 | dobby156 | anyone? |
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| 12:41.44 | emc | hi, is it possible to call any Skype ID with Asterisk or would you have to buy a "connector" for every single Skype ID I want to call? |
| 12:41.57 | heysuits | emc: you can call any Skype ID you want with chan_skype |
| 12:42.09 | heysuits | are you talking FROM Asterisk, or TO asterisk? |
| 12:42.27 | emc | heysuits: from asterisk to a skype ID? |
| 12:42.44 | heysuits | emc: in that case, ya, you just call whatever Skype ID you want from Asterisk |
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| 12:43.00 | heysuits | you can call 1 Skype ID PER license |
| 12:43.10 | emc | heysuits: that's great! thanks! |
| 12:43.10 | heysuits | i.e. one simultaneous call per chan_skype license |
| 12:43.20 | heysuits | just like codec_g729 |
| 12:43.27 | dobby156 | doesn't matter i seem to have fixed my issue by send a blank line after each cmd |
| 12:43.45 | emc | heysuits: I see |
| 12:43.47 | heysuits | so ya, you don't need to 1 license for 700,000,000 skype IDs :) |
| 12:44.02 | heysuits | errrr.... you don't need 1 license per 700M skype IDs :) |
| 12:44.11 | heysuits | whatever you know what I mean |
| 12:44.28 | heysuits | 1 license gets to 1 simultaneous call for any end point on the Skype network |
| 12:44.33 | heysuits | that's what it boils down to |
| 12:44.55 | emc | heysuits: yep :) ... thanks for the info. But I need 700m licenses if I want to call all 700m skype users at the same time, right? :) |
| 12:44.56 | *** join/#asterisk zkn (~zkn@217-159-228-106-dsl.est.estpak.ee) |
| 12:45.14 | emc | heysuits: yeah, great, thanks! |
| 12:45.22 | heysuits | emc: yes :) |
| 12:45.31 | emc | I've heard about chan skype, but their page seems down |
| 12:45.39 | heysuits | 1 license gets you one outbound or inbound call |
| 12:45.47 | emc | so i wasn't sure if they're still going |
| 12:46.13 | heysuits | I think we're talking different things |
| 12:46.17 | heysuits | I'm talking Skype For Asterisk |
| 12:46.17 | heysuits | http://www.digium.com/en/products/software/skypeforasterisk.php |
| 12:46.27 | heysuits | if you're talking about something else, then I've no idea |
| 12:46.51 | emc | http://www.chanskype.com |
| 12:46.59 | heysuits | oh ya, that's a hack |
| 12:47.09 | heysuits | you need a windows box and all that jazz (at least from last I checked) |
| 12:47.25 | heysuits | it's really just a way of connecting your sound card in asterisk to a skype softphone |
| 12:47.37 | heysuits | as far as I understand it anyways |
| 12:47.43 | emc | ah, i see |
| 12:47.47 | heysuits | Skype For Asterisk is native Skype support within Asterisk |
| 12:47.47 | emc | ok, great, thanks for the link heysuits |
| 12:48.04 | heysuits | Juggie: I am online at your request |
| 12:48.11 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-99.ks.ks.cox.net) |
| 12:51.32 | wdoekes2 | heysuits: you can close #19272 (dupe of closed #19192) if you want |
| 12:52.08 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 12:54.15 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 12:59.31 | *** join/#asterisk davlefou (~david@41.225.29.169) |
| 13:00.20 | emc | heysuits: and if we want to make a call Asterisk -> Skypeout -> PSTN ... do you know, would that also require one license? |
| 13:03.15 | heysuits | wdoekes2: awesome will do! |
| 13:03.42 | heysuits | emc: honestly I'm not 100% sure how the Skypeout stuff works in relation to licenses. I would suspect it would be similar and would require a license yes |
| 13:04.17 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-231.ips.direcpath.com) |
| 13:05.02 | emc | heysuits: ok ... thanks. If i'd have to guess, i would have guessed no, you wouldn't need one as skype will make money on the skypeout -> pstn connection already. But I may be wrong. |
| 13:05.57 | heysuits | emc: well the question would be how you get connected to Skypeout -- if it is via SIP, then you're gold |
| 13:06.06 | heysuits | if it's via the Skype network, then ya, you'd need a license |
| 13:06.25 | heysuits | so like SIP for Skype (or whatever Skype calls it) may allow you to use SkypeIn and SkypeOut directly via SIP |
| 13:06.28 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 13:07.46 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 13:07.53 | emc | heysuits: yes, they have someting called "Skype Connect" which is Skype for SIP: http://www.skype.com/intl/en-us/business/skype-connect/ |
| 13:11.03 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 13:11.04 | heysuits | ya that'd make sense |
| 13:11.50 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 13:12.18 | irroot | with M$ getting hold of skype im little hesitant to bother with it at the momennt |
| 13:13.45 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-65-9-252-231.mia.bellsouth.net) |
| 13:18.33 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:e804:d94f:7c5f:3db1) |
| 13:19.28 | luckman212 | gm |
| 13:19.51 | Jcook_5xData | to up grade asterisk from 1.6.* to 1.8 I just compile it and make install like normal? |
| 13:20.02 | *** join/#asterisk davlefou (~david@41.225.250.221) |
| 13:21.04 | luckman212 | my * box (1.8 SVN 319920M) is sending out private (192.168.x) IPs in the SIP INVITEs to my trunk providers. i'm trying to figure out why... I've got nat=yes set in sip.conf and defined my localnet= correctly |
| 13:21.28 | irroot | also need to look at making SRTP optional |
| 13:21.43 | luckman212 | outbound calls work fine, 2-way audio is working but incoming calls get dropped as soon as they get past the IVR and try to bridge to a local sip phone on my lan |
| 13:27.52 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 13:28.01 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 13:28.54 | irroot | luckman212 often this is due to media not been compatible |
| 13:29.07 | luckman212 | just using PCMU |
| 13:29.23 | luckman212 | no I have captured the packet trace using tcpdump, and I see the problem |
| 13:29.25 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:29.30 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:30.05 | irroot | jcook_5xdata no |
| 13:30.10 | luckman212 | the problem is, when * sends the REINVITE to try to route the RTP to my actual phone (which has a LAN IP e.g. 192.168.20.226) it is not replacing the local IP with my extern ip |
| 13:30.12 | irroot | mostly but not completley |
| 13:31.02 | luckman212 | I mean, * is actually sending an INVITE to my sip trunk with a line like "c=IN IP4 192.168.20.226" in the SDP |
| 13:31.09 | irroot | there are some changes between 1.6/1.8 |
| 13:31.20 | luckman212 | this is not valid and I get a "488 Not Acceptable Here" msg back from the trunk |
| 13:31.36 | luckman212 | whic is to be expected, since its not a routable IP |
| 13:31.39 | irroot | turn off reinvite or enable features in dial |
| 13:32.21 | luckman212 | dont want to turn off reinvite, because I don't want my * server proxying the media |
| 13:32.51 | luckman212 | i just need the correct (external) IP sent in my sip headers, not sure why * is sending the local one |
| 13:34.06 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:34.35 | *** join/#asterisk m4xx (~m4xx@75-144-154-165-NewEngland.hfc.comcastbusiness.net) |
| 13:37.01 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 13:37.09 | skrusty | afternoon |
| 13:37.46 | skrusty | does anyone know if it's possbile to pass a channel or sip device id to chanspy? |
| 13:38.23 | skrusty | (the channel would be a sip channel too) |
| 13:44.51 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:44.55 | irroot | luckman sounds like config typo |
| 13:47.01 | *** join/#asterisk Yoe (~wouter@samba.grep.be) |
| 13:47.09 | Jcook_5xData | irroot, no? Can you point me to a upgrade guide |
| 13:47.12 | luckman212 | cant be.. i checked everything & am using freepbx gui anyway so that is writing out my configs |
| 13:47.49 | Yoe | hi -- I upgraded asterisk from 1.4 to 1.6, and now outgoing calls over a Linksys SPA ATA don't work properly anymore |
| 13:48.00 | Yoe | that is to say, the call is made and I can hear people talking, but they can't hear me talking |
| 13:48.03 | Yoe | anyone seen that before? |
| 13:48.41 | irroot | Jcook_5xData in the 1.8 source directory see UPGRADE.txt |
| 13:48.49 | irroot | to see what applies to you |
| 13:49.00 | Jcook_5xData | ok thanks |
| 13:49.17 | billmania | skrusty: According to http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy, the answer to your question seems to be "yes". Have you already tried that? |
| 13:49.37 | *** join/#asterisk davlefou (~david@41.225.23.248) |
| 13:50.22 | *** join/#asterisk byronc (~byron@byron.theclarkfamily.name) |
| 13:50.43 | irroot | Jcook_5xData it was mostly trivial i use realtime and had to make some schema adjustments and some dialplan changes it worked with warnings |
| 13:57.08 | _justdave | I'm having a strange upgrade-related issue as well... just upgraded a box from 1.4 to 1.8 in my case. |
| 13:57.15 | _justdave | done it with 3 other boxes and had no issues |
| 13:57.25 | _justdave | using flat-file configs in /etc/asterisk, not realtime |
| 13:57.44 | _justdave | in my case, inbound calls from a trunk are getting rejected with an error that the extension doesn't exist |
| 13:57.52 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 13:57.52 | _justdave | but 'dialplan show' says it does |
| 13:58.29 | _justdave | (or at least a wildcard match that matches that pattern anyway) |
| 13:59.23 | *** join/#asterisk mclaro (~mclaro@190.183.222.194) |
| 13:59.25 | _justdave | " request '''237@from-inside' does not exist " I see too many quote marks up front though, I suspect that's related. wonder what's causing that... |
| 14:00.25 | luckman212 | _justdave: why are your inbound trunk calls coming into the "from-inside" context? |
| 14:00.35 | _justdave | because it's an inter-office trunk |
| 14:00.40 | _justdave | not an outside one |
| 14:00.41 | luckman212 | ah |
| 14:01.15 | luckman212 | hmm, not sure then |
| 14:03.45 | irroot | Ok its going to be done encryption= in sip has to change !! make it a global option with more than yes/no add 32bit/try/try32bit as options |
| 14:03.52 | irroot | any objections ?? |
| 14:04.07 | pigpen | As I type this question, I ask myself, "Why do I get stuck with this weird stuff..." |
| 14:04.55 | pigpen | Is there a way (i have been researching and come up empty) that if a caller is in a queue, and they decide to hangup, I could run a custom script? |
| 14:05.05 | irroot | encryption=try will include 80bit but not fail if it fails it will remove the srtp struct and carry on as if it was not set |
| 14:05.14 | malcolmd | irroot: sounds like what bbeers was working on. +1 to that |
| 14:05.21 | pigpen | ie: user entered queue, gets impatient, hangs up, script runs (maybe to email some CID info) |
| 14:05.23 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:05.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.33 | irroot | 32bit will put a _32 bit for snom compat into the SDP |
| 14:06.10 | malcolmd | irroot: to that, i think we also need to accept AVP if it includes crypto lines, since that seems to be what people are doing for "optional" encryption these days, in absence of an official RFC'd stance. |
| 14:06.33 | *** join/#asterisk asterisk0001 (~root@216.204.211.179) |
| 14:06.37 | irroot | indeed must go both ways |
| 14:06.41 | malcolmd | irroot: it's not just snoms that like 32, it's also linksys phones |
| 14:06.49 | asterisk0001 | Hi Guys! |
| 14:06.54 | malcolmd | howdy |
| 14:07.10 | malcolmd | irroot: newer snoms (maybe it's just a newer firmware) have support for 80bit, i think? |
| 14:07.12 | irroot | malcomd if its "try" then AVP/SAVP can be used |
| 14:07.15 | asterisk0001 | I have a stupid issue |
| 14:07.18 | pigpen | wow, look, it is the first asterisk!!! |
| 14:07.42 | irroot | no must not look at SAVP at all |
| 14:07.58 | malcolmd | irroot: agree. |
| 14:07.58 | asterisk0001 | I am configuring queue and agent but i am getting issue |
| 14:08.02 | asterisk0001 | Agent/7202 (Invalid) has taken no calls yet |
| 14:08.20 | asterisk0001 | what the hell is (Invalid) ??? |
| 14:08.38 | irroot | polycom and snom and i suspect linksys will continue with AVP if there is no SAVP |
| 14:08.48 | malcolmd | irroot: doesn't help backwards compatibility, but in hindsight, maybe it makes more sense to have an encryption=yes/no/try option and a encryption-key-length=32/80 option? |
| 14:09.09 | irroot | im with you there |
| 14:09.19 | irroot | or like t38pt_udptl |
| 14:09.44 | irroot | encryption=[yes/no/try][,30/80] |
| 14:09.46 | malcolmd | yeah. yes/no/try,80/32 |
| 14:09.55 | irroot | snap |
| 14:10.08 | malcolmd | that'd totally work, and not harm backwards compatibility |
| 14:10.33 | irroot | the default will be 80 of course |
| 14:11.00 | malcolmd | yes. are you in the code, patching away? |
| 14:11.03 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 14:11.25 | irroot | started hacking it earlier |
| 14:11.38 | irroot | got a rather happy customer with locks on there phones |
| 14:12.14 | malcolmd | yeah, locks on my blink phone make me happy |
| 14:12.52 | irroot | apparently im paranoid but you would be too if black helicopters were looking for you ;) |
| 14:13.29 | _justdave | ok, it's one server it doesn't like :| |
| 14:13.50 | _justdave | If I place the same call to it from a different office it works fine |
| 14:13.55 | malcolmd | i think we have another problem, in that if a device tags on the key lifetime, we barf on it |
| 14:14.01 | _justdave | pokes at the context settings |
| 14:14.31 | irroot | malcomd aint seen that yet so a example will be usefull |
| 14:14.51 | malcolmd | http://forums.asterisk.org/viewtopic.php?f=1&t=77789&p=155992&hilit=srtp+lifetime#p155992 |
| 14:15.12 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:15.28 | irroot | ok looks like SIP_PAGE2 is full moving using SIP_PAGE3_SRTP_KEY32 |
| 14:19.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:19.41 | _justdave | aha, I think I found it |
| 14:20.36 | Jcook_5xData | what is NT mode? |
| 14:20.52 | _justdave | 1.8 doesn't let you use quotes to delimit strings, it becomes part of the string. |
| 14:20.53 | Jcook_5xData | I have search google but I find nothing on what it is |
| 14:20.59 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 14:21.08 | irroot | jcook_5xData if you using ISDN ?? |
| 14:21.24 | _justdave | I had an old outbound route entry on the source server that had single quotes around an empty variable |
| 14:21.28 | Jcook_5xData | irroot, T1/PRI |
| 14:21.30 | irroot | NT Mode is network mode |
| 14:21.49 | irroot | ah T1/E1/J1 its pri_net |
| 14:21.56 | Jcook_5xData | aaahhhh |
| 14:21.58 | irroot | BRI NT |
| 14:22.22 | irroot | dont think ISDN BRI is big if available at all in states ?? |
| 14:23.31 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:24.31 | Jcook_5xData | irroot, I am try to find out why I see this: chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1. I am have bouncing on my pri. I am trying to find out if a telco problem or mine |
| 14:24.34 | _justdave | specifically, needs to be double-quotes, not single-quotes |
| 14:25.02 | malcolmd | nah, ISDN BRI is hard to come by in the states; some in california, some in virginia (government), but there's not much else |
| 14:25.33 | _justdave | I have ISDN BRI on one of my servers |
| 14:25.38 | _justdave | but it's in Paris. :) |
| 14:25.47 | irroot | Jcook_5xData hehe if you were here in south africa id tell you beyond a doubt it is telkom our local [most useless] monopoly |
| 14:25.55 | _justdave | and yes, we get that same error constantly |
| 14:26.11 | _justdave | dahdi_scan from the shell says the line is in red alarm |
| 14:26.22 | _justdave | but asterisk thinks it works fine (other than tossing us that error constantly) |
| 14:26.23 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 14:26.33 | _justdave | and we can place and receive calls without any issue |
| 14:26.59 | _justdave | seems to be a driver issue in dahdi is my guess. |
| 14:27.05 | skrusty | billmania: yes i have, i get an odd error, one moment |
| 14:27.11 | malcolmd | irroot: scratch that, it wasn't a lifetime problem; i believe it was because there's no support for the optional MKI parameter. i think that says we're busting if it's present on the crypto line |
| 14:27.13 | _justdave | we had a ticket open with Digium on it for a while, I'm not sure it ever got resolved. |
| 14:27.16 | Jcook_5xData | mine work will, then randomly it will red alarm in then come back up |
| 14:28.11 | _justdave | it actually wasn't working originally, config changes Digium gave us through that ticket got it working, but didn't make the errors go away |
| 14:28.50 | irroot | malcolmd see so i should have a patch on RB latter/tommorow for the SRTP hacks |
| 14:29.20 | Jcook_5xData | That why I ask about upgrading to 1.8 see if fixes the problem |
| 14:29.54 | malcolmd | irroot: i'm happy to test, if i can round up some devices that only do 32bit; |
| 14:30.20 | irroot | i got cpl 100 snoms all over |
| 14:30.49 | skrusty | billmania: pbx.c:9278 device_state_cb: Received invalid event that had no device IE |
| 14:31.08 | Jcook_5xData | just happen again pri reset |
| 14:31.27 | irroot | need to test interop with others |
| 14:31.49 | billmania | skrusty: You've exceeded my experience with ChanSpy(). :-( |
| 14:31.57 | skrusty | doh :) |
| 14:32.08 | skrusty | all i want to do is pas it the channel to spy on :/ |
| 14:32.15 | skrusty | why is it being an arse! :) |
| 14:32.53 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 14:33.10 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:35.53 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 14:36.12 | billmania | skrusty: Which version of asterisk are you using? I'm using "ChanSpy(SIP/1234)" successfully with 1.6.0.22. |
| 14:36.48 | Jcook_5xData | If anyone is willing to take a look at the pri debug when the reset happen http://pastebin.com/3UzMME8R I would greatly be appreciative |
| 14:36.59 | skrusty | 1.6.2.1 - i should upgrade i suppose |
| 14:37.08 | skrusty | either way |
| 14:41.06 | Jcook_5xData | :) If I can not get this fixed I need to go back to old crappy Nortel system |
| 14:41.53 | QuantumSchema | Hey all! |
| 14:42.06 | *** join/#asterisk glam (~glam@113.88.6.41) |
| 14:42.07 | QuantumSchema | Jcook_5xData: I just saw your post. |
| 14:42.10 | QuantumSchema | What's goin' on? |
| 14:42.25 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:42.25 | QuantumSchema | I'm tied to a Nortel Option 11c and went through all the headaches... |
| 14:42.55 | QuantumSchema | The company is currently on the Opt11 and the IT Dept is on Asterisk tied to the Opt11. |
| 14:43.09 | QuantumSchema | Here very soon we'll be migrating all users off of the Opt11 and over to Asterisk... |
| 14:43.21 | Khratos | Good morning |
| 14:43.34 | Jcook_5xData | pri I reset randomly. here the pri debug when the drop happen http://pastebin.com/3UzMME8R |
| 14:43.42 | Katty | so how's the end of the world coming for your aussie folks |
| 14:44.18 | QuantumSchema | Jcook_5xData: What's the Nortel system? |
| 14:45.12 | Jcook_5xData | Nope Asterisk 1.6.2.17, I have had ATT on the horn for over two weeks they have replace every thing in the field but it still happens. |
| 14:46.13 | QuantumSchema | Jcook_5xData: Oooo... so your Asterisk box is tied via T1 to the Telco? |
| 14:46.16 | Jcook_5xData | from the pri debug it looks to me like they ATT is sending a reset command. but that a guess |
| 14:46.37 | *** join/#asterisk ideaman (~ihaveapla@c-98-202-74-138.hsd1.ut.comcast.net) |
| 14:46.39 | Katty | is everyone prepared for tmw? |
| 14:46.45 | Katty | hugs leifmadsen |
| 14:46.46 | leifmadsen | yes! |
| 14:46.50 | Katty | leifmadsen: are you prepared for zombies?! |
| 14:46.53 | leifmadsen | I fully embrace the rapture |
| 14:46.58 | Jcook_5xData | yup, If I can not get it to stop resetting I need to move back to the old nortal system |
| 14:47.03 | Katty | leifmadsen: where are you holding up at? |
| 14:47.10 | leifmadsen | Katty: ON A BOAT! |
| 14:47.11 | Katty | leifmadsen: i need to check with some aussies and see how the Good Fight is going |
| 14:47.19 | QuantumSchema | Jcook_5xData: What protocol are you using? |
| 14:47.21 | Katty | leifmadsen: oh that sounds lovely |
| 14:47.32 | Katty | leifmadsen: i have a wedding to attend first, but then i think we may go hold up at target. |
| 14:47.38 | leifmadsen | nice |
| 14:47.39 | Jcook_5xData | QuantumSchema, in dahdi? |
| 14:47.43 | QuantumSchema | Yup |
| 14:47.53 | luckman212 | guys, should you ever have "nat=yes" defined for a trunk's context? |
| 14:47.57 | QuantumSchema | Jcook_5xData: Switch type? |
| 14:48.17 | *** join/#asterisk glam (~glam@113.88.6.41) |
| 14:49.25 | luckman212 | i would think that e.g. if your * server is at 192.168.1.200 and you have a sip phone also behind the SAME nat, that for that particular sip phone, the correct setting would be "nat=no" since there is no NAT between * and the sip phone (even though the phone itself IS behind a nat) |
| 14:50.02 | _justdave | Jcook_5xData: I'd be surprised if BRI doesn't work better on 1.8 than it does on 1.6, but the system I was having those issues with was 1.8 from the start, and still had them |
| 14:50.05 | luckman212 | or is nat=yes for ANY device behind a nat, regardless of whether it happens to be the same subnet as * |
| 14:50.08 | *** join/#asterisk nellyb (~neil@93-96-178-95.dsl.cnl.uk.net) |
| 14:50.16 | _justdave | the built-in BRI support in DAHDI was very new in 1.6 |
| 14:50.25 | Jcook_5xData | QuantumSchema, signalling=fxo_ls |
| 14:50.29 | _justdave | in 1.4 you needed msisdn or something like that |
| 14:50.47 | irroot | i still use misdn in 1.8 |
| 14:51.05 | irroot | has support for USB not tested in dahdi |
| 14:51.35 | QuantumSchema | Jcook_5xData: What's the Switch Type set to? Did they tell you to use FXO_LS for signaling for some reason? Is this a T1 connection? |
| 14:51.45 | *** join/#asterisk glam (~glam@113.88.6.41) |
| 14:52.27 | irroot | leifmadsen lol yeah 6pm GMT ill have the popcorn maybe swing past the hindi shops pick up some fireworks ;) |
| 14:52.29 | Jcook_5xData | QuantumSchema, yes it is T1/PRI - it most be the default setting I did not change it |
| 14:52.54 | *** join/#asterisk GTXComm (~John@cpe-72-128-62-30.kc.res.rr.com) |
| 14:53.39 | QuantumSchema | Jcook_5xData: Hmm.... what's the switch type set to? National ISDN 2, QSig, DMS100? |
| 14:54.26 | Jcook_5xData | QuantumSchema, dms100 it was set to National , but I change it late night just to see if it help... Did not |
| 14:54.49 | malcolmd | Kcook_5xData: running a digium card for that t1? |
| 14:55.21 | malcolmd | er.. Jcook, that is. silly xchat for acqua and it's lack of tab-completion for nicks |
| 14:55.31 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 14:55.45 | QuantumSchema | Jcook_5xData: Here's what I would try.... |
| 14:55.47 | Jcook_5xData | No, rhino R1T1 I wish I bought Digium now :( |
| 14:56.05 | QuantumSchema | <--- Bought a Digium to support the cause. ;-) |
| 14:56.49 | QuantumSchema | Jcook_5xData: Framing/Coding: ESF/B8ZS; Signaling: PRI - CPE; Switchtype: National; |
| 14:57.09 | QuantumSchema | Jcook_5xData: Then do a dahdi restart... |
| 14:57.42 | Jcook_5xData | Alway Have in the passed :q!there support have been good thou, |
| 14:57.55 | Jcook_5xData | sorry lol |
| 14:57.58 | malcolmd | QuantumSchema: many thanks, helps keep the lights on 'n such. |
| 14:58.36 | Jcook_5xData | QuantumSchema, just to verify set this in chan_dahdi.conf |
| 15:00.02 | QuantumSchema | malcomd: Glad I could help! Hopefully I'll be up to the HQ for training here soon. I might not need it by then but I wanna learn from the best and help donate. :-D |
| 15:00.33 | QuantumSchema | Jcook_5xData: chan_dahdi gets signalling=pri_cpe switchtype=national |
| 15:00.42 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59.dsl.dyn.forthnet.gr) |
| 15:02.10 | QuantumSchema | Jcook_5xData: your dahdi system.conf should have something like span=1,1,0,esf,b8zs |
| 15:02.20 | Jcook_5xData | it does |
| 15:02.26 | malcolmd | QuantumSchema: plenty of opportunities; roughly once/month for classes in HSV :) |
| 15:03.45 | Qwell | malcolmd: To what address should trucks filled with money be delivered? |
| 15:03.46 | QuantumSchema | malcomd: I should be up there in the next month or so! |
| 15:05.18 | malcolmd | Qwell: ATTN: Accounts, 445 Jan Davis Drive NW, Huntsville, AL 35806, United States or, if you're filling out a web form because those truckloads are coming from somewhere in elbonia, then it's John Smith, 100 Rodeo Dr., Beverly Hills, CA 90210, United States |
| 15:05.47 | QuantumSchema | Jcook_5xData: how's it lookin? |
| 15:05.48 | malcolmd | QuantumSchema: drop me a note when you're here ( <-@digium.com) and i'll pop by for one of the lunches |
| 15:06.12 | leifmadsen | speaking of lunch, I'm starving! |
| 15:06.22 | QuantumSchema | malcomd: Definately! ::adding to contacts now:: |
| 15:06.35 | Jcook_5xData | QuantumSchema, I have the setting in waiting for a channels to clear up so I restart it |
| 15:06.56 | leifmadsen | irroot: I'm a fan of fireworks :) |
| 15:07.04 | leifmadsen | irroot: nice to see you on IRC! |
| 15:07.13 | leifmadsen | good work on the bug tracker and patches lately! |
| 15:08.14 | irroot | liefmadsen ta its a pleasure we have just over 100 systems we moved to 1.8 recently and i want them as stable as possible |
| 15:08.58 | leifmadsen | me too |
| 15:09.12 | leifmadsen | I think we've made some good progress recently |
| 15:09.36 | malcolmd | leifmadsen: we're leaving the end of the world bug in, right? the one that'll cause them all to stop working tomorrow? the ones in Japan are already starting to shut down, right? |
| 15:09.45 | irroot | its really helped having access to svn i can now merge and keep branches 1.8/trunk |
| 15:09.56 | leifmadsen | malcolmd: yes that is correct, the Rapture Patch is active |
| 15:10.04 | leifmadsen | irroot: heck ya! |
| 15:10.12 | malcolmd | leifmadsen: thank goodness, i can sleep much better tonight |
| 15:10.28 | leifmadsen | malcolmd: yep, should be nice to get through a night without people calling me |
| 15:10.42 | irroot | its like y2k all over |
| 15:11.07 | irroot | ill do the same be on standby .... for any rocking party |
| 15:12.13 | *** part/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 15:12.38 | leifmadsen | zombie party! |
| 15:12.50 | irroot | good theme indeed |
| 15:13.51 | irroot | no one has yet taken me up on the wager if they dont go i get there salery :P |
| 15:17.20 | irroot | cool sip encryption has new options ... yes|no|try[,32bit] |
| 15:17.29 | malcolmd | w00t |
| 15:19.12 | Qwell | 32bit? |
| 15:19.41 | irroot | there are 2 key lengths for SRTP 80/32 |
| 15:19.53 | Qwell | That should be a different config option, I'd think. |
| 15:19.55 | irroot | asterisk only sends 80bit |
| 15:20.10 | irroot | this breaks many devices who are 32bit only |
| 15:20.11 | Qwell | enckeylen=32, enckeylen=80, etc |
| 15:21.00 | irroot | Qwell want to keep it as backward compat as possible |
| 15:21.23 | QuantumSchema | Jcook_5xData: Still waiting? |
| 15:21.43 | irroot | did a similar thing to t38pt_udptl |
| 15:21.46 | Jcook_5xData | QuantumSchema, yup |
| 15:22.21 | Jcook_5xData | QuantumSchema, getting close to lunch So it will drop off soon |
| 15:23.03 | QuantumSchema | Jcook_5xData: Makes sense. |
| 15:23.08 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 15:23.41 | skrusty | i have the music from Shaun of the Dead in my head now... |
| 15:23.51 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:25.33 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:29.53 | Qwell | irroot: yeah, but it wasn't an option before. there's no compat to keep there. |
| 15:31.27 | irroot | true ill do some testing and can change it latter |
| 15:31.31 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59.dsl.dyn.forthnet.gr) |
| 15:32.20 | *** join/#asterisk Freeaqingme_ (~Freeaqing@2001:980:45b3:1:216:3eff:fe00:5) |
| 15:33.00 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 15:33.54 | Qwell | wwgd: ping? |
| 15:34.09 | QuantumSchema | Jcook_5xData: Gonna go grab some food.... |
| 15:34.34 | Khratos | Qwell: wwgd will just join/part in a while(TRUE) loop... |
| 15:34.59 | Qwell | Khratos: yes, that's the point of the ping |
| 15:36.03 | Khratos | Qwell: have one of you noticed that when you export some parts of the documentation (Wiki) to PDF, the content of some parts do not get exported? E.g., in the AEL documentation: https://wiki.asterisk.org/wiki/display/AST/General+Notes+about+AEL+Syntax |
| 15:36.10 | Jcook_5xData | QuantumSchema, cool |
| 15:36.23 | Khratos | If you export that to PDF, the AEL lines will be empty |
| 15:36.26 | Qwell | Khratos: never tried |
| 15:36.36 | Jcook_5xData | QuantumSchema, thanks for your help much appreciated |
| 15:37.01 | Khratos | That happens too in a file that gets downloaded with the source of Asterisk 1.8 (that file is probably exported from the Wiki) |
| 15:37.14 | Khratos | Its name is AST.pdf |
| 15:37.40 | Qwell | Khratos: malcolmd may know a little more about that stuff |
| 15:38.59 | leifmadsen | Khratos: ya that script just generates the data automatically |
| 15:39.33 | leifmadsen | some things may need tweaking because the stuff in the wiki was imported from LaTeX to the wiki, and is now being exported from the wiki to a PDF |
| 15:40.10 | leifmadsen | it's like www.translationparty.com |
| 15:41.04 | Qwell | http://www.translationparty.com/#9334395 |
| 15:41.49 | *** join/#asterisk davlefou (~david@196.203.146.36) |
| 15:41.51 | leifmadsen | wtf.... |
| 15:41.52 | Khratos | haha |
| 15:42.15 | Qwell | leifmadsen: <3 |
| 15:45.39 | malcolmd | looks like the wiki doesn't like exporting code that's in the "code" macro; but doesn't have a problem with the noformat macro. changed those instances to "noformat" and now they all appear in a pdf export |
| 15:45.50 | mickecarlsson | If I want to keep track of the changes in Asterisk 1.8, which is best to check out, branches/1.8 or trunk? |
| 15:47.00 | Qwell | mickecarlsson: trunk is trunk, branches/1.8 is 1.8 |
| 15:47.24 | mickecarlsson | Thanks Qwell, so co of 1.8 then |
| 15:48.17 | Khratos | malcolmd: maybe that's something related to permissions (to protect copyright) |
| 15:49.05 | Khratos | It should be configurable somewhere as code in {code} is far more legible than in {noformat} |
| 15:49.46 | irroot | mickcarlsson remember branches is always +1 ie /1.8 is going to be 1.8.5-rc1 as 1.8.4 is tagged and the tags are only worked on for bugs in the -rc series |
| 15:50.45 | mickecarlsson | irroot thanks for the clarification |
| 15:50.57 | Khratos | And, if possible, documents on personal spaces could be made accesible to public without registration |
| 15:54.02 | leifmadsen | all changes are in the branches/1.8, then when we do an RC1, that is all changes in the branch since the previous RC1 (i.e. 1.8.5-rc1 has all the changes since 1.8.4-rc1). When we do an RC2, it is a direct copy of the previous RC (i.e. 1.8.4-rc2 is exactly the same as 1.8.4-rc1, and not changes after the 1.8.4-rc1) except for specifically merged changes (cherry picked changes). |
| 15:54.19 | leifmadsen | http://blogs.asterisk.org/2010/09/02/the-monthly-asterisk-release-cycle/ |
| 15:54.51 | mickecarlsson | is reading the blog |
| 15:55.37 | asterisk0001 | How to add SIP/XXX in queue static ? |
| 15:56.38 | mickecarlsson | leifmadsen obvious question, when does a ticket/issue show up in the roadmap? |
| 15:57.22 | Qwell | mickecarlsson: when it gets put there |
| 15:57.38 | mickecarlsson | did see that coming |
| 15:59.08 | leifmadsen | mickecarlsson: http://pastebin.com/BW42mzaX |
| 15:59.36 | leifmadsen | asterisk0001: just add it to the queue in queues.conf.... |
| 15:59.39 | leifmadsen | member => SIP/XXX |
| 16:01.09 | leifmadsen | goes to lunch |
| 16:02.29 | irroot | ok qwell / liefmadsen question not answered yet with the svn team structure i have added a branch with patchset i push to customers with all the work / hacks i am busy with this helps a great deal there is no problem with having it in tree i hope ?? |
| 16:04.27 | pigpen | Hi all, I will re-post as it has been 3 hours and don't like to repost. My guess is the possibility is slim: |
| 16:04.33 | pigpen | Is there a way (i have been researching and come up empty) that if a caller is in a queue, and they decide to hangup, I could run a custom script? |
| 16:04.40 | *** join/#asterisk AnSkywalker (~anskywalk@200.69.203.181) |
| 16:04.41 | pigpen | ie: user entered queue, gets impatient, hangs up, script runs (maybe to email some CID info) |
| 16:04.54 | irroot | sure pigpen that can be done |
| 16:04.56 | mickecarlsson | queue events? |
| 16:05.14 | pigpen | Only on hangups? |
| 16:05.15 | irroot | put a macro to run on the hangup is a nice idea |
| 16:05.46 | pigpen | but the hangup would be the caller initiated...I thought that ended the dialplan? |
| 16:05.52 | irroot | pigpen define a h exten in the quee and run it there |
| 16:06.17 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 16:06.24 | pigpen | so stick the queue call in a special context, include an "h" exten. |
| 16:06.27 | mickecarlsson | pigpen: http://www.voip-info.org/wiki/view/asterisk+manager+events |
| 16:06.36 | mickecarlsson | Look at the end of that page |
| 16:06.41 | mickecarlsson | 'Hangup' Event |
| 16:06.55 | pigpen | won't that grab all hangups, even if the call was accepted by a queue member? |
| 16:07.04 | mickecarlsson | Yes |
| 16:07.36 | pigpen | yeah, I am only wanting to just grab the ones that are "impatient" and hangup before being connected to a queue memeber. |
| 16:07.52 | pigpen | but good info on the other, I didn't know that was possible (never needed it) |
| 16:09.20 | AnSkywalker | Hello... sorry for the interruption... someone knows what CentOs version is used in AsteriskNow 1.7.1? |
| 16:09.26 | mickecarlsson | Anyone here that use Realtime for Voicemail? |
| 16:09.52 | irroot | mickcarlsson use it for as much as possible |
| 16:09.58 | pigpen | mickecarlsson, yeah, realtime voicemail, sip exen's, dialplan..... |
| 16:09.59 | pigpen | you name it. |
| 16:10.11 | mickecarlsson | Well, I am rewriting FreePBX voicemail module to use it |
| 16:10.25 | mickecarlsson | And I have some issues regarding [general] context |
| 16:10.33 | mickecarlsson | It seems not to be used |
| 16:10.48 | mickecarlsson | All settings are per user as I have tetsed it |
| 16:11.04 | mickecarlsson | If I set an antry in [general] it wont be oicked up |
| 16:11.13 | mickecarlsson | needs a new keyboard |
| 16:11.18 | Qwell | AnSkywalker: 5 |
| 16:11.23 | *** join/#asterisk davlefou (~david@41.225.247.108) |
| 16:12.09 | irroot | mickcarlsson if there is a col in the DB it will use that always make sure all options are set per user one option is to drop the col that you want to have global from schema |
| 16:12.49 | AnSkywalker | Qwell: Tankyou! i got to build a NIC driver and i need to download kernel-devel package |
| 16:13.23 | Qwell | what driver? |
| 16:13.44 | Khratos | malcolmd: could this be our wiki issue? https://67-221-237-7.contegix.com/browse/CONF-21823 |
| 16:14.18 | mickecarlsson | irroot so if I have all columns that are available in my db, it is per user and general is not needed? |
| 16:14.20 | AnSkywalker | a driver for the onboard NIC of an DH61BE intel board |
| 16:15.00 | mickecarlsson | I have columns context, set that to default for users and general for [general] |
| 16:15.48 | AnSkywalker | the system did not detect the NIC interface... i have no other net connection... i hope the intel CD driver work |
| 16:15.50 | irroot | mickcarlsson yeah setting defaults on the cols is a way of having "genral" so if the insert is missing fields the default kicks in always good to have sane defaults |
| 16:17.02 | AnSkywalker | Qwell: a driver for the onboard NIC of an DH61BE intel board |
| 16:17.14 | AnSkywalker | Qwell: the system did not detect the NIC interface... i have no other net connection... i hope the intel CD driver work |
| 16:17.26 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 16:19.10 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 16:19.23 | neurosys | anyway to debug multiple seg faults? :/ |
| 16:19.52 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 16:20.23 | *** join/#asterisk W0rdm (~rdm@ip174-70-54-174.om.om.cox.net) |
| 16:20.31 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 16:20.52 | Khratos | maybe a ~backtrace on infobot would be helpful |
| 16:21.28 | Qwell | ~backtrace |
| 16:21.28 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 16:21.34 | Qwell | like so? :P |
| 16:21.47 | Khratos | oops |
| 16:23.41 | Katty | hello my asterisk does not work at all how to fix plz |
| 16:23.52 | Qwell | Katty: Have you tried turning it off and then on again? |
| 16:24.43 | QuantumSchema | Jcook_5xData: I'm back. Any luck? |
| 16:25.44 | Khratos | So, thanks to Murphy's Law, Confluence 3.4.9 has the bug with the {code} tags, and it was fixed in 3.5 |
| 16:26.12 | Khratos | And the wiki results to be 3.4.9 |
| 16:27.00 | *** part/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 16:28.14 | *** join/#asterisk mazpe (~lesterm@li315-205.members.linode.com) |
| 16:28.36 | mazpe | anyone running asterisk 1.8 at linode? |
| 16:29.10 | irroot | weekend is go ... lets bring it on like donkey kong localtime 18:30 |
| 16:29.31 | W0rdm | Not me, but I'm considering it |
| 16:29.57 | mazpe | i'm trying to switch from EC2 |
| 16:30.57 | Khratos | so, the world is already over in china... |
| 16:31.27 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 16:31.59 | coppice | no its not |
| 16:32.05 | Yoe | mazpe: EC2 outage giving you headache? |
| 16:32.29 | Khratos | uuum |
| 16:32.56 | mazpe | Yoe: yeah, that... their support sucks! unless you pay extra for it. Not quite fond of EC2 at the moment. |
| 16:33.04 | *** join/#asterisk cj (~cjac@adsl-207-32-169-17.rockisland.net) |
| 16:33.07 | cj | hey folks |
| 16:33.21 | cj | any of you know where I might find docs on nortel/avaya sip phone provisioning? |
| 16:38.08 | irroot | cj the unistm conf does a good job of explaining it |
| 16:38.34 | irroot | had a few in the office a while back they work but not much more |
| 16:39.23 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 16:40.04 | mazpe | uhm.. debian asterisk-mysql still on 1.6 |
| 16:40.21 | cj | irroot: alrighty. I'll take a look. thanks! |
| 16:40.33 | _justdave | ok, so after my Asterisk 1.4 -> 1.8 upgrade, none of my Cisco phones will register |
| 16:40.34 | Jcook_5xData | QuantumSchema, I restart by dahdi and asterisk. so far so good! now come the waiting game to see if happen again |
| 16:40.39 | _justdave | the Polycom ones are all working fine |
| 16:41.07 | _justdave | running a sip debug on one of them shows asterisk is tossing back a 401 Unauthorized response back to all of the registration attempts from Cisco phones |
| 16:41.22 | Qwell | leifmadsen: ^^ |
| 16:42.13 | cj | pabelanger: have you published a new version of the .deb since 1:1.8.3.3-1digium1~squeeze? |
| 16:42.18 | cj | checks with apt-cache |
| 16:42.51 | Qwell | asterisk-mp3_1.8.4-1digium1~squeeze |
| 16:42.53 | Qwell | err |
| 16:42.56 | Qwell | - -mp3 |
| 16:42.57 | neurosys | wth is this? 0:00 rasterisk r |
| 16:43.10 | Qwell | neurosys: asterisk -r ~= rasterisk |
| 16:43.14 | QuantumSchema | Jcook_5xData: Cool! Hopefully we see something good come from it! ::crosses fingers:: |
| 16:44.14 | Jcook_5xData | QuantumSchema, I hope so. or it will be a long weekend.. :D |
| 16:45.19 | mazpe | oh my! |
| 16:45.21 | mazpe | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 16:45.23 | mazpe | fancy |
| 16:45.36 | mazpe | I wonder if the DAHDI package will work in linode |
| 16:47.28 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 16:48.00 | WIMPy | mickecarlsson: ping |
| 16:48.05 | ChannelZ | ...so is Skype going to start charging in July to use their user manager thing required for using Skype For Asterisk? |
| 16:48.10 | mickecarlsson | WIMPy pong |
| 16:48.13 | _justdave | google suggests you'll get that behavior with cisco phones if you don't have proxy_register: "1" in SIPDefault.cnf |
| 16:48.17 | _justdave | but I have that already |
| 16:48.20 | WIMPy | Great. |
| 16:48.36 | Qwell | _justdave: what version of Asterisk? |
| 16:48.41 | _justdave | 1.8.4 |
| 16:48.45 | mickecarlsson | WIMPy, I must run in five minutes |
| 16:48.46 | Qwell | wait a few hours |
| 16:49.03 | _justdave | Qwell: we've rebooted a few of the phones, they just sit there trying to register forever |
| 16:49.12 | Qwell | _justdave: I mean, for 1.8.4.1 |
| 16:49.17 | _justdave | oh. |
| 16:49.24 | WIMPy | mickecarlsson: re the issue with the wrong hangupcause. When I thought about the delay, it actually might be a feature. Could it be that those changes allow outgoing overlap dialing? |
| 16:49.32 | _justdave | so it's a bug in 1.8.4 that got fixed in 1.8.4.1? |
| 16:49.33 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:49.38 | Qwell | _justdave: yes |
| 16:49.50 | Qwell | https://issues.asterisk.org/view.php?id=18951 |
| 16:50.00 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 16:50.03 | mickecarlsson | WIMPy: well, it might be a feature, but not for trunk dialing |
| 16:50.17 | pabelanger | cj: yes, 1.8.4 should be released |
| 16:50.24 | WIMPy | mickecarlsson: Exactely tor that. |
| 16:50.35 | cj | pabelanger: yeah, looks like I pulled that one down. thanks! |
| 16:50.42 | *** join/#asterisk davlefou (~david@41.225.250.223) |
| 16:51.07 | mickecarlsson | WINPy: As we in Sweden can have phone numbers that are from 8 to more than 10 digits |
| 16:51.11 | pabelanger | looks like I'd do .18.4.1 later today |
| 16:51.52 | cj | cool beans |
| 16:51.53 | WIMPy | mickecarlsson: And that feature migh enable you to dial additional digits if the number was incomplete. |
| 16:51.54 | mickecarlsson | WIMPy: so, if I dial 04059240 it is a to short number, bit 040592400 is correct |
| 16:52.16 | mickecarlsson | WIMPy, no, as the provider already has closed the channel |
| 16:52.39 | WIMPy | mickecarlsson: Sore, but you might set up another one durig that timeout. |
| 16:53.03 | mickecarlsson | WIMPy: It is like this, Grandstream phones has a feature that they dial any number keyed on the phone 4 seconds after the last digit |
| 16:53.05 | WIMPy | Sure* |
| 16:53.51 | mickecarlsson | So, we have sales people keying in ahlf a phone number, then look for the rest of it, then the phone dials, and the providers return SIP 484 |
| 16:54.10 | WIMPy | mickecarlsson: Most sip phones use en block dialing, but not all. |
| 16:54.25 | mickecarlsson | In asterisk 1.4 I have that caught and play a voice telling the dialer that the number as to short |
| 16:54.34 | mickecarlsson | And that fails on 1.8 |
| 16:54.41 | WIMPy | That's exactely my point. Might that change enable them to complete the number afterwards? |
| 16:54.52 | mickecarlsson | You would be suprised how often this happens |
| 16:55.04 | _justdave | did that get broken specifically in 1.8.4? (like if I back down to 1.8.3 for now will it work?) |
| 16:55.10 | WIMPy | I'm not. I know that problem pretty well. |
| 16:55.18 | mickecarlsson | Well, how do I know that the number is complete? |
| 16:55.23 | cj | irroot: hurm. I was looking for tips on AVP format for the dialplan.txt file sent to the phones by the "provisioning server" aka tftp/dhcp/pxeboot server |
| 16:55.27 | WIMPy | It's quite annoying to many users :-( |
| 16:55.37 | mickecarlsson | I can dial 08660777555 and that is a valid number |
| 16:55.46 | WIMPy | mickecarlsson: By sending it to your provider until they accept it. |
| 16:55.50 | _justdave | then I can pull 1.8.4.1 this afternoon when it comes out |
| 16:56.03 | mickecarlsson | Well, they just return 484 and hang up |
| 16:56.05 | Qwell | _justdave: could also apply the patch on that issue |
| 16:56.09 | WIMPy | mickecarlsson: The same way it would happen in the PSTN. |
| 16:56.10 | mickecarlsson | As they should |
| 16:56.24 | mickecarlsson | Yes, so it is a bug in Asterisk :-) |
| 16:56.30 | mickecarlsson | Or a regression |
| 16:56.34 | WIMPy | mickecarlsson: Do you have overlapdialing enabled in sip.conf? |
| 16:56.37 | mickecarlsson | As it work like a charm in 1.4 |
| 16:56.40 | mickecarlsson | No |
| 16:56.48 | WIMPy | I think it might be a useful feature. |
| 16:57.03 | WIMPy | But it obviousely muse be configurable. |
| 16:57.08 | WIMPy | must |
| 16:57.08 | irroot | cj mmm dont remember doing that actually cant rember what we did ;) |
| 16:57.22 | cj | hokay |
| 16:57.32 | mickecarlsson | Could be, when it is fixed in 1.8 I will upgrade, and I am staying on 1.4 for the moment |
| 16:57.46 | WIMPy | mickecarlsson: Does the current version still produce a timeout? |
| 16:57.51 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:57.51 | mickecarlsson | Yes |
| 16:57.54 | irroot | cj if it was more than 43200 seconds ago its erased |
| 16:57.57 | mickecarlsson | Ah, No actually |
| 16:58.06 | mickecarlsson | it returns 111 instead of 28 |
| 16:58.09 | cj | irroot: :) |
| 16:58.17 | WIMPy | mickecarlsson: But immediately? |
| 16:58.20 | mickecarlsson | Yes |
| 16:58.34 | WIMPy | That's clearly not a feature. |
| 16:58.44 | mickecarlsson | I got to go, soccer begins now, and I wont miss that |
| 16:58.44 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 16:58.49 | mickecarlsson | See you later |
| 16:58.58 | WIMPy | ok. HF |
| 16:59.05 | Eitan | gentlemen, what would u guys recomend as a PoE switch... 24 port |
| 16:59.38 | citywok | Catalyst 3750G |
| 17:00.16 | mazpe | fancy |
| 17:00.32 | citywok | probably a bit overkill, but he didn't give any other requirements :P |
| 17:04.11 | citywok | Eitan: in all seriousness i bought a 24 port network smartswitch that was 10/100 POE for like $350 for a small business and it works great |
| 17:04.20 | citywok | s/network/netgear/ |
| 17:05.30 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 17:05.33 | cusco | hi folks |
| 17:05.40 | citywok | marco |
| 17:05.47 | pigpen | polo |
| 17:05.48 | ChannelZ | Are all 24 ports PoE? I have a couple of unmanaged Netgear 16-port switches but only the first 8 are PoE |
| 17:06.05 | citywok | ChannelZ: yea i have a few of those things too, but the 24 port is 24 ports of POE |
| 17:06.22 | cusco | ChannelZ: I have a compule of swiches that only the first 2 ports are PoE |
| 17:06.27 | citywok | ChannelZ: although most POE switches don't have enough power to run all 24 ports @ 15.4 watts, the netgear can only run all 24 @ 7 |
| 17:06.27 | ChannelZ | hmm nice |
| 17:06.29 | cusco | :p |
| 17:06.40 | ChannelZ | nice-ish |
| 17:06.51 | cusco | ok |
| 17:06.54 | citywok | even the more expensive ones frequently don't have that much power |
| 17:07.12 | Eitan | ok, i was gonna go netgear |
| 17:07.16 | Eitan | for same price like $350 |
| 17:07.33 | cusco | I'm trying to compile chan_datacard for asterisk 1.8 and if fails. I have read here: http://code.google.com/p/datacard/issues/detail?id=72 -- that asterisk.h has 1.6 headers ?? |
| 17:08.23 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053225.ics.uci.edu) |
| 17:08.37 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:12.00 | cj | anyone here have experience with the Diameter CDR? |
| 17:13.50 | cj | what type of wattage do FXO ports put out at the -48v? |
| 17:17.35 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 17:17.50 | *** part/#asterisk emc (~cburgdorf@host81-149-6-231.in-addr.btopenworld.com) |
| 17:21.12 | irroot | cj not as much wattage as when you get a techie to tounge test it with a 89v ring voltage ... for everything else there is mastercard |
| 17:22.24 | irroot | oops cj that was a FXS FXO not much at all |
| 17:22.49 | cj | irroot: orite. got 'em mixed up again. |
| 17:23.05 | cj | FXS is the one that connects to a dumb terminal. FXO is the one that accepts powers. |
| 17:23.36 | Eitan | so i have an HP smart array 1000 |
| 17:23.42 | *** join/#asterisk deadpigeon (~deadpigeo@office.xpressamerica.net) |
| 17:23.47 | Eitan | can i do something cool with it for my asterisk server |
| 17:24.02 | cj | step one: put -48v pair in mouth. step two: call mouth. |
| 17:24.04 | Yoe | Eitan: correct me if I'm wrong, but a smart array is a disk cabinet |
| 17:24.05 | cj | nom nom nom |
| 17:24.17 | Eitan | yes sir |
| 17:24.27 | Eitan | a very heave large Raid array of some sort |
| 17:24.36 | *** join/#asterisk [GABRI] (d5fbab4a@gateway/web/freenode/ip.213.251.171.74) |
| 17:24.45 | Yoe | yes. And how would you do something "cool" asterisk-related with it? |
| 17:25.06 | Eitan | not sure, thats why im in the #asterisk room asking that |
| 17:25.07 | Eitan | lol |
| 17:25.09 | Qwell | I have a hard drive. Can I do anything cool with it, with Asterisk? |
| 17:25.18 | Eitan | yeah, u can store ur shit |
| 17:25.23 | Qwell | There you go then. |
| 17:25.37 | citywok | lol, you can store every call you've ever made... |
| 17:25.42 | Eitan | qwell: u cant do anything cooler than storing ur shit on your hjd |
| 17:25.45 | citywok | i'm up to about 100gb in the last year in my call center... |
| 17:25.53 | Eitan | there we go |
| 17:25.58 | Qwell | citywok: Only newbs count in gb. |
| 17:26.03 | Eitan | hahahaha |
| 17:26.13 | Qwell | pb or bust. |
| 17:26.14 | cj | Eitan: you could store call records and put up a pretty web interface to your PBX :) |
| 17:26.17 | citywok | heh, i'm not cool enough for a t or a p |
| 17:26.19 | [GABRI] | Hi people i am newbie with asterisk and agi . i am encocing my fisrt agi script but i have an error in asterisk console: http://pastebin.com/v1YrKdBa |
| 17:26.37 | cj | yay! I'm no longer a n00b! |
| 17:26.42 | Eitan | hahahaha |
| 17:26.49 | Eitan | i guess thats what i will ahve to do |
| 17:26.51 | cj | has a 4.5t array :) |
| 17:27.01 | Eitan | oh look at you |
| 17:27.06 | citywok | for asterisk? jesus! lol |
| 17:27.14 | cj | citywok: no. just for general purposes :) |
| 17:27.17 | irroot | you haz spam posted the afternoons srtp hacking on RB |
| 17:27.24 | citywok | oh, if we were counting general purposes i've probably got 40tb of storage |
| 17:27.27 | cj | citywok: though I expect I'll need something that size when the CLEC takes off :) |
| 17:27.39 | Eitan | i just came up on a bunch of server equip, and this array was just sitting there |
| 17:27.45 | Eitan | i guess i could store call recordings and such |
| 17:27.55 | citywok | Eitan: is it an MSA 1000 w/ fibre, or is it just a disk shelf? |
| 17:28.04 | cj | Eitan: is it a bunch of old 38G SCSI devices? :) |
| 17:28.09 | Qwell | [GABRI]: Can you execute that directly from the commandline? |
| 17:28.30 | [GABRI] | Qwell: with php -q mandarsilla.agi? |
| 17:28.30 | Eitan | citywok: its got fiber |
| 17:28.39 | Qwell | [GABRI]: /path/to/ma...blah.agi |
| 17:28.42 | Eitan | cj: 78 gig scsi :) |
| 17:28.50 | cj | awesome! |
| 17:28.52 | *** join/#asterisk cHarNe2 (thorn@gateway/shell/bshellz.net/x-kwqbcynxubnhvxay) |
| 17:29.02 | Eitan | it must weigh about 100 lbs |
| 17:29.03 | Yoe | Eitan: it's more like a san thing then. Nice. |
| 17:29.08 | cj | Eitan: SX fibers? |
| 17:29.10 | Yoe | that was expensive when it was bought |
| 17:29.11 | citywok | Eitan: ah, so it's the MSA1000. be careful, we had one of those with mixed controllers and when one controller failed the other locked up and erased the config for the entire array of 3 shelves. |
| 17:29.21 | Eitan | oh shit |
| 17:29.38 | Yoe | citywok: good thing you had backups (you did, right?) |
| 17:29.39 | citywok | i re-created the array with the same config and it worked for a few hours... then tanked permanently. |
| 17:29.42 | Eitan | let me get the model number iof this bad boy |
| 17:29.42 | cHarNe2 | so witch of the 97 conf-files do i need to edit to get a working asterisk? o.O |
| 17:29.52 | [GABRI] | mm Qwell my php is executed but the output is black |
| 17:29.54 | citywok | cHarNe2: most of them |
| 17:29.55 | citywok | Yoe: yes :) |
| 17:30.10 | citywok | it was a good test of our backup plan, we only lose 4 hours worth of data, and a day's worth of business. *only* |
| 17:30.12 | cHarNe2 | citywok: most as in 50 of them? |
| 17:30.14 | cj | cHarNe2: have you bought the book yet? :) |
| 17:30.18 | Eitan | hp storage works MSD San switch |
| 17:30.28 | cHarNe2 | cj: no |
| 17:30.35 | citywok | ~book |
| 17:30.35 | infobot | For more information about the Asterisk book, see ~thebook |
| 17:30.37 | Yoe | citywok: at this place where I used to work, the dba actually did rm -rf / on the data warehouse |
| 17:30.53 | cj | cHarNe2: whelp, you might want to grab a copy of the old one and reduce the configs to bare minimum |
| 17:30.55 | irroot | charne2 define working first :P |
| 17:31.05 | cj | then start adding "features" as you find need for them |
| 17:31.11 | Yoe | this was back in 2001, when a TiB was a lot of data, and then they found out that two out of their fifteen tapes were bad |
| 17:31.13 | cj | or you could just use the defaults and tweak |
| 17:31.27 | cHarNe2 | irroot: i only wont simple sip to sip calls |
| 17:31.38 | Eitan | hp storage works MSD San switch litterally weighs 100lbs... thats gotta mean its expensive |
| 17:31.48 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:32.35 | Yoe | Eitan: electronics aren't sold by the kilo (though sometimes I wish that were true) |
| 17:32.58 | Eitan | hahah :) |
| 17:33.23 | Eitan | they should be, cause i would buy a large amount of CPU's |
| 17:33.42 | [GABRI] | Qwell: this is correct? $destino=$agi->get_variable("DESTINO"); |
| 17:33.44 | Yoe | and a small amount of PSUs? |
| 17:33.58 | cusco | I can configure trunk asterisk with --with-ssl just fine |
| 17:34.03 | cusco | but on 1.8.4 I get: configure: *** The OPENSSL installation appears to be missing or broken. |
| 17:34.06 | cusco | why ? |
| 17:34.13 | cusco | what does it check for |
| 17:34.29 | Eitan | lol, maybe a few psus |
| 17:34.41 | cusco | ssl.h |
| 17:35.02 | _justdave | wonders how they ever manage to build the asterisk RPMs if the SRPMs they post don't build without modification |
| 17:35.29 | cj | okay... one more time: Is anyone here familiar with the Diameter CDR? |
| 17:36.36 | Qwell | _justdave: How do they not build? |
| 17:36.44 | _justdave | error: Installed (but unpackaged) file(s) found: /usr/lib/asterisk/modules/chan_gtalk.so /usr/lib/asterisk/modules/chan_jingle.so /usr/lib/asterisk/modules/res_config_ldap.so /usr/lib/asterisk/modules/res_jabber.so |
| 17:37.11 | Qwell | _justdave: you have extra deps installed |
| 17:37.32 | _justdave | because the BuildRequires for the SRPM made me install them |
| 17:37.37 | _justdave | before I could build |
| 17:37.50 | Qwell | The buildrequires did not make you install libiksemel from source. :) |
| 17:38.18 | _justdave | I don't have anything installed from source, this is all RPMs on this box |
| 17:38.30 | Qwell | libiksemel is not part of CentOS. |
| 17:38.37 | _justdave | I'm on RHEL |
| 17:38.45 | Qwell | libiksemel is not part of RHEL. |
| 17:38.50 | Yoe | _justdave: that's the same thing, mostly |
| 17:39.06 | _justdave | looks like it came out of rpmforge |
| 17:39.14 | Qwell | as for ldap, I actually have no idea what deps are needed for that |
| 17:39.15 | Yoe | _justdave: the only difference between CentOS and RedHat is the price and the logo |
| 17:39.41 | Qwell | _justdave: patches welcome, when you figure it out |
| 17:39.58 | _justdave | Qwell: I did already, they've been in digium's bugtracker for 5 or 6 months |
| 17:40.04 | _justdave | nobody's looked at them yet |
| 17:41.01 | m4xx | can someone please explain the difference between argN and valN when calling an ODBC_function ? |
| 17:41.10 | cj | hey paul? |
| 17:41.11 | cj | $ svn co --non-interactive --trust-server-cert https://origsvn.digium.com/svn/dpkg/asterisk-1.8/branches/squeeze asterisk-squeeze |
| 17:41.14 | cj | svn: OPTIONS of 'https://origsvn.digium.com/svn/dpkg/asterisk-1.8/branches/squeeze': SSL handshake failed, client certificate was requested: SSL error: GnuTLS internal error. (https://origsvn.digium.com) |
| 17:41.43 | Qwell | m4xx: Set(${ODBC_FOO(arg1,arg2,arg3)}=val1,val2,val3) |
| 17:41.56 | m4xx | yes, i see that on voip-info.org |
| 17:42.03 | Qwell | m4xx: the val stuff may be wrong, but you get the idea |
| 17:42.04 | cj | pabelanger: ^^ |
| 17:42.47 | m4xx | i still dont understand |
| 17:43.02 | Qwell | cj: Do you have a cert? |
| 17:43.07 | [GABRI] | People i need help with this: http://pastebin.com/xyujTUv4 |
| 17:43.07 | pabelanger | cj: origsvn is private server, it still have to be exposed on the public side. Should be done for next week |
| 17:43.15 | [GABRI] | I can't see any error |
| 17:43.29 | [GABRI] | But i haven't got any mail from agi script |
| 17:43.33 | Eitan | does anyone else find the sound of multiple servers running soothing |
| 17:43.35 | Eitan | or is that just me |
| 17:43.39 | cj | alrighty. apt-get source it is |
| 17:44.14 | cusco | cj: apt won't fetch you 1.8 |
| 17:44.26 | cusco | (on debian repos that is) |
| 17:44.46 | Yoe | Eitan: I used to sleep in a room with multiple systems running 24/7. That wasn't soothing, at all. |
| 17:44.48 | cj | Qwell: no. feel like issuing me one? :) |
| 17:44.51 | citywok | why don't you just use the real source? :P |
| 17:44.59 | cj | cusco: that's alright. I'm configured to use paul's |
| 17:45.02 | Qwell | cj: $999.94 |
| 17:45.17 | cj | Qwell: under $1K! |
| 17:45.38 | m4xx | Qwell, can you please elaborate a little? |
| 17:46.14 | Qwell | m4xx: I don't know how I can make that clearer. |
| 17:46.33 | m4xx | what is valN ? |
| 17:46.44 | m4xx | the a return from the function? |
| 17:47.44 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:48.25 | pabelanger | cusco: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 17:48.35 | Khratos | exit |
| 17:48.40 | Khratos | oops |
| 17:48.44 | Qwell | m4xx: no |
| 17:48.45 | citywok | don't forget that / |
| 17:49.19 | cusco | pabelanger: I did correct mysefl by stating debian repos |
| 17:49.33 | pabelanger | nods |
| 17:49.38 | cusco | but now I know there are asterisk builds packaged for debian :) |
| 17:49.57 | pabelanger | cusco: however tzafrir_laptop just pushed 1.8.3 into debian testing |
| 17:50.00 | cusco | tho I always compile from source, Im having troubles now using --with-ssl |
| 17:50.08 | cusco | did he? |
| 17:50.08 | [GABRI] | Qwell: can you see any problem in my code? |
| 17:50.14 | cusco | Im using testing and just searched |
| 17:50.39 | cusco | Provides: asterisk-1.6.2 |
| 17:51.51 | pabelanger | cusco: my mistake, 1.8.4 and into unstable |
| 17:53.41 | cusco | ah! |
| 17:53.57 | cusco | kk |
| 17:54.36 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:55.39 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:55.43 | *** join/#asterisk g_r_eek (~g_r_eek@p50894F6B.dip0.t-ipconnect.de) |
| 17:56.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 17:59.09 | leifmadsen | does anyone here, who happens to be using Asterisk 1.8, also utilize queues which have "ringinuse=yes" enabled? |
| 17:59.42 | leifmadsen | the reason I ask, is I'm trying to track down this "ringback" sound in the receptionists ear, and I've setup her phone to do exactly the same thing as every other phone here, and hers is the only one that gets that ringback sound. |
| 17:59.54 | leifmadsen | So I'm curious if it could potentially be a bug or something else in the Queue() application. |
| 18:00.16 | *** join/#asterisk aliverius (~quassel@athedsl-393831.home.otenet.gr) |
| 18:01.26 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 18:04.33 | m4xx | are valN and argN interchangeable? |
| 18:05.35 | m4xx | looking at example1 on http://www.voip-info.org/wiki/view/Asterisk+func+func_odbc is totally confusing |
| 18:05.54 | leifmadsen | m4xx: no they are not |
| 18:06.09 | leifmadsen | Set(ODBC_WRITE_FUNCTION(arg1,arg2)=val1,val2) |
| 18:06.25 | leifmadsen | m4xx: you shouldn't use that wiki -- much newer documentation exists elsewhere |
| 18:06.48 | m4xx | chapter 16 in the book goes from an exteremly simple example to an extremely complex example |
| 18:08.16 | leifmadsen | m4xx: in which spot? |
| 18:09.38 | m4xx | you have A Gentle Introduction to func_odbc, then the Hot Desking example |
| 18:10.24 | leifmadsen | m4xx: basically ARGx is only used when reading values: i.e. Set(Result=${GET_SOMETHING(arg1,arg2)}) |
| 18:10.33 | leifmadsen | and VALx is only when writing |
| 18:10.34 | leifmadsen | ie. |
| 18:10.45 | leifmadsen | Set(PUT_SOMETHING()=val1) |
| 18:10.56 | m4xx | thank you =] |
| 18:11.06 | Qwell | has no idea how that's different from what he said |
| 18:11.20 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-051.mobile.uci.edu) |
| 18:11.50 | leifmadsen | Qwell: well no one believes what you say |
| 18:12.06 | leifmadsen | m4xx: although you make a valid point about the gentle intro could maybe be expanded a bit |
| 18:12.12 | leifmadsen | I'll note it in my wiki |
| 18:14.56 | leifmadsen | one |
| 18:14.57 | leifmadsen | done* |
| 18:16.12 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 18:20.23 | _justdave | here's all those SRPM patches, fwiw: https://issues.asterisk.org/search.php?project_id=1&reporter_id=6185&sticky_issues=on&sortby=last_updated&dir=DESC&hide_status_id=90 |
| 18:21.17 | *** join/#asterisk oliver1 (~oliver@manz-5f74a29a.pool.mediaWays.net) |
| 18:28.16 | dan__t | jfc, seriously? |
| 18:28.33 | dan__t | i've been banging on this bullshit freetds for three days now and the first thing i read on that patch page was about freetds |
| 18:28.36 | dan__t | i threw up in my mouth a little bit. |
| 18:30.56 | dan__t | what's worse than freetds? |
| 18:30.59 | dan__t | anything by EasySoft |
| 18:31.58 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:34.02 | *** join/#asterisk unitz2 (~uni@adsl-69-234-108-85.dsl.irvnca.pacbell.net) |
| 18:35.43 | irroot | its that time of night get funky with naked woman or code .... |
| 18:36.23 | leifmadsen | irroot: it's only 2:30pm here, but ok! |
| 18:37.18 | irroot | lol leifmadsen and been winter its dark sooner and getting cold ... add to the confusion :P |
| 18:38.04 | leifmadsen | irroot: indeed, it's getting hotter here :) |
| 18:38.15 | leifmadsen | I've just come out of hibernation recently |
| 18:40.17 | *** part/#asterisk zorp75ck (~zorp75ck@146.186.115.103) |
| 18:44.55 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-036.mobile.uci.edu) |
| 19:01.54 | dan__t | been kinda cold here too |
| 19:01.56 | dan__t | like 75F |
| 19:06.10 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 19:06.18 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053225.ics.uci.edu) |
| 19:09.29 | cj | whiner |
| 19:09.46 | *** join/#asterisk oDesk (~oDesk@188.52.83.59) |
| 19:09.51 | oDesk | hello all |
| 19:10.25 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 19:14.17 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 19:15.48 | *** join/#asterisk oDesk (~oDesk@188.50.75.193) |
| 19:16.03 | oDesk | hello all |
| 19:23.38 | _justdave | so the patch on that bug doesn't apply cleanly to 1.8.4. 1.8.3.3 still has the bug apparently, downgrading to that didn't help |
| 19:23.43 | _justdave | (this is the cisco SIP thing) |
| 19:24.11 | _justdave | tries to apply the patch by hand |
| 19:26.24 | ectospasm | sounds painful |
| 19:31.37 | _justdave | this patch must be for a different version of asterisk. the code doesn't even look close to what it's trying to patch |
| 19:31.47 | *** join/#asterisk Janos (~cramos@186.32.57.8) |
| 19:32.31 | _justdave | so how do I find out what the datestamp was on r304241 on the 1.4 branch? |
| 19:32.35 | Janos | hello, question, is there such a thing as standard set of dial codes for features like transfer etc ? |
| 19:32.39 | _justdave | apparently that's the revision that introduced this bug |
| 19:32.53 | _justdave | (which was cross-committed to 1.6 and 1.8, too, apparently) |
| 19:33.15 | _justdave | Janos: yeah, defaults are listed in features.conf |
| 19:33.29 | _justdave | (which is also where you'd change them) |
| 19:33.48 | m4xx | is it possible to test an odbc function from the cli? |
| 19:36.28 | _justdave | 2011-01-26 20:43 +0000 [r304245] Matthew Nicholson <mnicholson@digium.com> |
| 19:36.35 | _justdave | that looks like the offending commit on the 1.8 branch |
| 19:36.50 | _justdave | tries to figure out which version included that |
| 19:37.56 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 19:39.15 | _justdave | hmm, somewhere between 1.8.3 and 1.8.4 |
| 19:40.08 | _justdave | 1.8.3.3 obvious has it already |
| 19:40.14 | _justdave | so something earlier 1.8.3 probably doesn't |
| 19:40.26 | Janos | _justdave, i mean like international standard and not asterisk specific |
| 19:40.46 | _justdave | Janos: ah, no clue then, sorry |
| 19:41.07 | _justdave | for that type of thing I suspect it varies by carrier and there's no real standard |
| 19:41.57 | Janos | i see, ok |
| 19:43.17 | _justdave | ok, as far as I can tell from the changelogs, 1.8.3.3 should *not* include this commit. so why's it still busted? :) |
| 19:43.38 | pabelanger | _justdave: try 1.8.4.1 tag from subversion |
| 19:43.45 | pabelanger | it should be fixed there |
| 19:44.12 | pabelanger | http://svn.asterisk.org/svn/asterisk/tags/1.8.4.1/ |
| 19:44.45 | pabelanger | m4xx: yes |
| 19:46.10 | _justdave | thanks |
| 19:46.12 | _justdave | checks out |
| 19:46.38 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:48.06 | billmania | Which Asterisk event(s) causes the variable ${UNIQUEID} to be assigned a new value? |
| 19:49.36 | *** join/#asterisk f2knight (~ben@c-71-59-210-143.hsd1.or.comcast.net) |
| 19:49.59 | f2knight | I need some help peering 2 asterisk boxes over sip. |
| 19:50.27 | f2knight | I have one box running asterisk realtime,. and a client PBX running straight asterisk. |
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| 19:50.35 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:50.48 | f2knight | I am sure its just a config option i am missing. because I can register a phone with the same account information with out a problem. |
| 19:52.04 | *** join/#asterisk oliver1 (~oliver@manz-5f74a29a.pool.mediaWays.net) |
| 19:55.07 | anonymouz666 | anyone in here using sangoma with e1? |
| 19:55.40 | WIMPy | ~ask |
| 19:55.40 | 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. |
| 19:59.39 | anonymouz666 | FAS errors and line code violation increasing... |
| 19:59.54 | anonymouz666 | LOSS OFF and LOS ON |
| 20:00.14 | anonymouz666 | line signal ON, already tried to change the framing... but nothing seems to work. |
| 20:00.42 | anonymouz666 | I also see that RX level is too low -40db to -44db |
| 20:01.23 | Chainsaw | anonymouz666: If the signal level is too low, fix that first. |
| 20:01.49 | anonymouz666 | until what value is acceptable? |
| 20:01.53 | anonymouz666 | I read about -2.5 |
| 20:02.12 | Chainsaw | anonymouz666: If I can't hear you because you're too far away, I don't spend money on buying you a megaphone or buying me a hearing aid. I move closer. |
| 20:02.18 | _justdave | 1.8.4.1 indeed fixes it. yay |
| 20:02.23 | Chainsaw | anonymouz666: This is quite similar. |
| 20:02.46 | Chainsaw | pabelanger: Neat, is that out yet? |
| 20:02.47 | anonymouz666 | already changed the cables |
| 20:02.55 | anonymouz666 | nothing changed |
| 20:03.06 | anonymouz666 | I am remote |
| 20:03.27 | malcolmd | _justdave: cisco phone? what model and what version of firmware on it? |
| 20:04.00 | pabelanger | Chainsaw: We're trying to verify the Cisco fix, but looks like monday now for the tarball release |
| 20:04.09 | Chainsaw | pabelanger: *nod* Thanks. |
| 20:05.21 | f2knight | I need some help peering 2 asterisk boxes. The documents I have found on line don't do seem to work. |
| 20:13.01 | _justdave | malcolmd: variety of them, mostly 5941 and 7941 |
| 20:13.24 | _justdave | SIP firmware 8.5.2 on the one I just snagged a UserAgent from |
| 20:14.24 | _justdave | er, 7945 was the first one, I guessed wrong on which digit was different. |
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| 20:25.13 | GreatSUN | re |
| 20:33.39 | Khratos | malcolmd: The Wiki open source license allows it to upgraded to recent versions? |
| 20:34.02 | malcolmd | Khratos: i *think* so. |
| 20:35.42 | Khratos | Ok, because the thing about the wiki not exporting code inside '{code}' tags, is a bug that was fixed after v3.5 |
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| 21:45.14 | citywok | okay so my 1.6.2.11 has shit a brick. |
| 21:45.35 | citywok | queueadd is no longer an AMI command, but it works from the CLI. queue reload doesn't work, but if i do queue (tab tab) it tells me reoad is an option. |
| 21:46.18 | malcolmd | hooray? :( |
| 21:48.42 | citywok | yea... had to restart the whole thing, waited until there were no active calls (well, there were 3 but none were connected) |
| 21:51.58 | malcolmd | Khratos: cool. thanks for the heads up on that. i've put in a request for information on upgrading the wiki to the maintainers of the hardware it's running on. |
| 21:54.04 | *** part/#asterisk W0rdm (~rdm@ip174-70-54-174.om.om.cox.net) |
| 21:54.36 | Khratos | Great. As new source releases comes with documentation extracted from the Wiki, it is important that it might be fixed |
| 22:03.30 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:05.28 | malcolmd | should work the other way around. documentation from the wiki gets pulled directly from the sources of the files by the wikibot. |
| 22:05.34 | malcolmd | afk; home. |
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| 22:47.35 | [GABRI] | Hi |
| 23:09.59 | carrar | hi |
| 23:19.31 | ChannelZ | HEY! |
| 23:19.46 | WIMPy | ho |
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| 23:47.29 | f2knight | I cant believe it .. all day and not a single person knows how to bridge 2 asterisk servers over sip? |
| 23:48.14 | WIMPy | I think you're missing a "not". |
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| 23:59.24 | carrar | f2knight, what are you talking about this channel is dedicated to ASCII Character 42 (2A) |