| 00:00.02 | ManxPower | paulc, that is less true these days. more and more places in the usa require 11 digit dialing |
| 00:00.42 | Barometer | weird, I mostly do 10 digit dialing |
| 00:01.02 | Barometer | area code + number is only 10 if you don't have to dial the country code |
| 00:01.13 | ManxPower | 1 for toll call |
| 00:01.18 | p3nguin | Only cell phones (around here) even allow 10-digit rather than 11. |
| 00:02.05 | Barometer | land lines and cells here work fine off 10 |
| 00:02.20 | Barometer | only need to dial 10 if you're calling another area code |
| 00:03.11 | Barometer | so, depends on your local carrier |
| 00:05.20 | KNERD | A contab for asterisk...should it be run under root, or as the asterisk user? |
| 00:05.58 | ChannelZ | What does the cron do? |
| 00:06.55 | KNERD | ChannelZ: billing |
| 00:07.50 | ChannelZ | well I mean is it calling binaries, or reading files, etc. If you can run it as asterisk run it as asterisk, provided the asterisk user has read access to whatever files it needs, etc. But we have no idea.. |
| 00:09.26 | KNERD | ChannelZ: it's calling PHP files to generate billing, notifications, invoices, etc |
| 00:10.13 | KNERD | as to opposed to running it as root |
| 00:11.13 | ChannelZ | Ok.. well again this PHP script, is it reading files to get this billing info? Connecting to MySQL? only you can really tell what user it should run as in order to have proper permission to do all the things it needs to do. |
| 00:12.01 | p3nguin | If you can do ALLLLLLL the stuff as a user other than root, you don't need to run the cron job as root. It's as simple as that. |
| 00:12.03 | ChannelZ | I mean you can run it as root and it will be able to do everything. But I'd say it's always preferable NOT to run as root if you can avoid it. |
| 00:12.34 | p3nguin | least privilege |
| 00:13.19 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 00:13.23 | KNERD | Well it is a set of scripts that interact with Asterisk, so I am guess it should be run as asterisk |
| 00:14.48 | *** join/#asterisk nighty^ (~nighty@210.188.173.245) |
| 00:16.22 | *** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk) |
| 00:17.38 | ChannelZ | HOW do they 'interact' with asterisk? Manager? |
| 00:19.07 | ChannelZ | actually it doesn't matter. The end of the story is you need to answer your own question. |
| 00:20.42 | KNERD | I guess directly they don't |
| 00:20.53 | *** part/#asterisk wwalker (~wwalker@mailbox.eclipsing.com) |
| 00:21.07 | KNERD | however they do influence the operation |
| 00:26.38 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:27.26 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 00:30.59 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 00:37.12 | KNERD | well I just tried doing "su asterisk" but was given the reponse of "This account is currently not available." What is the indicative of? |
| 00:37.14 | ruben23 | hi guys i got registartion error on my sip softphones,whn i log to a remote asterisk, but i cnat see any errors on the CLI |
| 00:37.39 | p3nguin | Sounds like asterisk has no shell. |
| 00:37.56 | p3nguin | "getent passwd asterisk" |
| 00:38.51 | p3nguin | It's not unusual for asterisk to not have a shell. |
| 00:43.02 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:44.19 | *** join/#asterisk brettnem (~brett@63.246.161.15) |
| 00:46.03 | *** part/#asterisk eskaypey (~eskaypey@unaffiliated/eskaypey) |
| 00:55.39 | ldiamond | When I call my DID from any phone, I get a weird ring tone (on the caller's phone) (beep----beep----beep) instead of the usual ring. Anyone know at what level the cause may be? I actually took the pbx out of the chain and only got the SIP provider and my SIP client. |
| 00:55.49 | *** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk) |
| 00:56.27 | [TK]D-Fender | ldiamond: "sound" is not a valid means to debug your problem. |
| 00:57.36 | ldiamond | ? |
| 00:58.15 | ldiamond | what do you mean? |
| 00:58.35 | p3nguin | With the phone and the provider talking directly, the same sound is still present? |
| 00:59.31 | [TK]D-Fender | ldiamond: "I heard a funny sound, whats the matter? |
| 00:59.44 | [TK]D-Fender | ldiamond: Not a valid means of find out what the problem is |
| 00:59.51 | ldiamond | Fender, I tried to describe the sound, but what more can I do? |
| 01:00.01 | [TK]D-Fender | ldiamond: LOOK AT THE CALL |
| 01:00.04 | ldiamond | everything works, its just the ring tone. |
| 01:00.32 | [TK]D-Fender | ldiamond: Depends on Every piece in between. Of which you have named NONE of them |
| 01:01.12 | ldiamond | [TK]D-Fender: actually, I did mention. "I actually took the pbx out of the chain and only got the SIP provider and my SIP client." |
| 01:01.15 | TJNII | "I actually took the pbx out of the chain" So ... you want us to debug a problem unrelated to Asterisk? |
| 01:01.29 | p3nguin | lol |
| 01:01.50 | ldiamond | TJNII: I know people here know about VOIP and might know the answer. |
| 01:02.22 | TJNII | True, but the "no details" part will make it a bitch. ("Beep beep beep" doesn't count as a detail.) |
| 01:02.26 | ldiamond | I am not looking for a detailed solution here, just someone to tell me what part may be responsible |
| 01:02.49 | TJNII | I'm gonna go with the phone. |
| 01:02.58 | ldiamond | Well, who is responsible for providing a ringtone? |
| 01:03.02 | ldiamond | the the caller. |
| 01:03.32 | *** join/#asterisk coppice (~chatzilla@m121-203-222-114.smartone-vodafone.com) |
| 01:03.45 | KNERD | p3nguin: the response is "asterisk:x:100:101::/var/lib/asterisk:/sbin/nologin" |
| 01:06.30 | p3nguin | and that's probably why it was unavailable. /sbin/nologin kinda has a tendency to do that. |
| 01:06.52 | TJNII | heheh |
| 01:07.09 | devmod | Is there a way to stop execution of a extension right after AGI() returns? |
| 01:07.21 | TJNII | hangup()? |
| 01:07.37 | devmod | on the same priority with some sort of option? |
| 01:08.07 | devmod | the following priority calls another AGI |
| 01:08.52 | KNERD | p3nguin: then what can be done ? I guess that would be a linux issue |
| 01:09.22 | TJNII | Why do you want to run as the asterisk user, anyways? |
| 01:10.12 | KNERD | To run some Cronjobs |
| 01:10.44 | carrar | cronjobs are cool |
| 01:11.38 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 01:12.11 | KNERD | so I want to run some cron jobs as asterisk |
| 01:13.24 | [TK]D-Fender | [21:01]<ldiamond>[TK]D-Fender: actually, I did mention. "I actually took the pbx out of the chain and only got the SIP provider and my SIP client." <- Yes.. you REMOVED one name and left none |
| 01:13.46 | [TK]D-Fender | ldiamond: For all we know, its your CLIENT that makes it sound that way |
| 01:13.57 | ldiamond | ok |
| 01:14.07 | ldiamond | Then I will try and fix it on the client, thanks |
| 01:16.13 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 01:21.02 | p3nguin | You don't need the user (asterisk) to have a shell for cron jobs to run. |
| 01:22.31 | p3nguin | knerd: ^^ |
| 01:22.34 | KNERD | p3nguin: then what is suggested to run those? |
| 01:22.50 | p3nguin | Are you new to crontab? |
| 01:22.56 | p3nguin | Are you new to Linux? |
| 01:23.07 | KNERD | crontab, yes |
| 01:23.09 | p3nguin | root can install any user's crontab. |
| 01:23.14 | p3nguin | crontab -u asterisk -e |
| 01:23.23 | KNERD | okay, thanks |
| 01:23.57 | p3nguin | and asterisk doesn't need a shell, because asterisk is not logging in. crond uses sh to process the cron jobs. |
| 01:26.38 | *** join/#asterisk QubeZ (~nkasu@110.67.204.68.cfl.res.rr.com) |
| 01:26.43 | QubeZ | hello all |
| 01:27.28 | mdodd | any word on the colors in CLI, guys?? |
| 01:27.33 | mdodd | they're so pretty! |
| 01:27.50 | QubeZ | Will asterisk handle queue priorities across multiple queues? For example, I know that within one queue, I can set several exten's with different priorities to "hop" in front of others. However, how does Asterisk handle multiple queues? |
| 01:42.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:45.55 | ChannelZ | mdodd: I think I answered this last night for someone else - if you run asterisk forked, you won't get colors |
| 01:46.31 | mdodd | ChannelZ: you mean -f? |
| 01:46.45 | mdodd | what if I don't want to run it forked? |
| 01:47.02 | p3nguin | then don't. |
| 01:47.03 | ChannelZ | no, no -f (-f means 'don't fork') |
| 01:47.44 | ChannelZ | normally when you run asterisk it forks off and runs in the background. Since there is no controlling terminal for asterisk in that case, it doesn't output color. |
| 01:48.19 | ChannelZ | One solution is to run the 'safe_asterisk' script instead, which runs a non-forking asterisk process (among other things) so * does see a terminal of sorts and will output color. |
| 01:48.45 | mdodd | how would I go about installing that? |
| 01:48.48 | mdodd | is it a pain? |
| 01:48.57 | p3nguin | It's probably already installed. |
| 01:49.09 | p3nguin | Or just don't fork asterisk when you run it. |
| 01:49.12 | ChannelZ | which safe_asterisk |
| 01:49.16 | p3nguin | Stop trying to complicate things. |
| 01:49.42 | ChannelZ | well that is "complicated" if you are not running it manually everytime you boot your machine |
| 01:50.03 | mdodd | yeah, seriously |
| 01:50.26 | ChannelZ | most of the init scripts that come with asterisk run safe_asterisk, except notably debian |
| 01:57.20 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 02:00.11 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 02:04.13 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 02:09.43 | ChannelZ | argh wtf. My softphone calling through my * to Vitelity works, yet my stupid ass SPA3102 configured pretty much exactly the same way going through the same dialplan is failing |
| 02:19.35 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 02:26.06 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 02:44.34 | ChannelZ | man SIP is sort of a mess of a protocol |
| 03:05.27 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 03:06.16 | ldiamond | anyone know sip providers that support SMS? |
| 03:10.59 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:14.31 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 03:17.20 | *** part/#asterisk ManxPower (~manxpower@195.sub-75-216-192.myvzw.com) |
| 03:19.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:25.14 | sun28 | moin |
| 03:26.35 | p3nguin | loin |
| 03:31.42 | ChannelZ | coin |
| 03:40.23 | *** join/#asterisk frenchy12345 (~Eric@pool-74-105-116-164.nwrknj.fios.verizon.net) |
| 03:41.07 | frenchy12345 | Hi I am new here and hopefully somebody can help me |
| 03:41.24 | p3nguin | ~ask |
| 03:41.41 | 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. |
| 03:41.41 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 03:41.43 | p3nguin | ~answers |
| 03:41.43 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 03:42.11 | frenchy12345 | well I setup Asterisk freePBX with 2 fxo cards |
| 03:43.02 | frenchy12345 | Everything works BUT how do I force outbound call to a specific Dahdi trunk |
| 03:43.16 | p3nguin | ~freepbx |
| 03:43.17 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:43.38 | ChannelZ | plays The Price Is Right fail sound |
| 03:43.53 | frenchy12345 | I see my trunks in g0 |
| 03:44.28 | p3nguin | Continuing to spew information isn't likely to yield different results. |
| 03:44.57 | frenchy12345 | ok thanx |
| 03:45.45 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 03:49.50 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:49.53 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 03:52.24 | *** part/#asterisk frenchy12345 (~Eric@pool-74-105-116-164.nwrknj.fios.verizon.net) |
| 03:54.50 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:56.44 | *** join/#asterisk OrNix (~OrNix@l49-0-149.cn.ru) |
| 03:59.26 | OrNix | comrades, does anybody now, how can i get the channel name with asterisk -rx "core show channels" if name of the channel doesn't fit entirely in line? (f.e. Datacard/datacard0-a49f, but it is displayed as Datacard/datacard0-a)) |
| 04:02.41 | p3nguin | Try "core show channels concise" and "core show channels verbose". |
| 04:03.43 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 04:07.40 | *** join/#asterisk QaDeS_ (~mklaus@p4FC7131D.dip0.t-ipconnect.de) |
| 04:08.28 | ChannelZ | you mean concise |
| 04:08.46 | p3nguin | What'd I say? |
| 04:09.22 | ChannelZ | You said both, but verbose I don't think shows the whole thing which is fun. |
| 04:09.45 | p3nguin | Oh, I was just recommending two things to check... I don't know if either one actually shows what he's asking for. |
| 04:10.39 | OrNix | p3nguin, thank you |
| 04:10.41 | ChannelZ | It might have changed at some point |
| 04:11.10 | p3nguin | I don't seem to have any channel names long enough to figure it out on my own. |
| 04:12.55 | ChannelZ | I think I mostly run into it with psuedo channels |
| 04:15.49 | ChannelZ | or sip channels actually, most of mine have names in the 8 or more char range. That with the call ID always runs it over |
| 04:18.26 | ChannelZ | submits his number to be ported, can't wait to dump the local telco |
| 04:19.03 | carrar | YEAH! |
| 04:19.20 | *** part/#asterisk Barometer (~jfkldasjf@ip72-208-208-187.ph.ph.cox.net) |
| 04:19.40 | ChannelZ | yah sick of paying ~$38/mo for a line. $10/mo for CallerID, fucking scam |
| 04:20.44 | p3nguin | wow |
| 04:20.52 | ChannelZ | especially since I'm barely use the phone |
| 04:22.14 | carrar | and so now how much will you be paying, and do they guarntee caller ID name? |
| 04:22.16 | p3nguin | $1/mo. for a DID and a penny a minute for calls sounds like a much better deal. |
| 04:23.47 | ChannelZ | I'm going with Vitelity, which is local to me, $1.49/mo, $.012/min inbound and $.014/min out |
| 04:24.30 | ChannelZ | CNAME lookups are .009 but I'm not sure I'll even use it yet |
| 04:24.32 | carrar | free local calls? |
| 04:24.59 | [TK]D-Fender | I'm almost never on the phone, only keep a cell @ $40/mo w/ unlimited internet, use my office for LD |
| 04:25.04 | ChannelZ | no it's those rates |
| 04:25.50 | ChannelZ | They have a ~$8/mo unlimited inbound but for as little as I"m on the phone at home I think it'll be cheaper to do the $1.50+mins |
| 04:26.06 | p3nguin | The way I see it, VoIP.ms is a better deal, since they use Vitelity's network and have lower prices. I wish I knew if the use ONLY Vitelity, though. |
| 04:26.44 | ldiamond | Any providers support sending and receiving SMS? |
| 04:37.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:44.15 | [TK]D-Fender | ldiamond: We probably wouldn't know... considering * doesn't support this. |
| 04:45.47 | p3nguin | We should rename the channel to something that better describes what we can help with. |
| 04:46.07 | [TK]D-Fender | me votes for #kittens |
| 04:46.18 | [TK]D-Fender | dangit |
| 04:52.12 | ChannelZ | #peepswhocantusegoogle |
| 04:55.00 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 05:04.12 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-aitrnwarwzifmarw) |
| 05:04.55 | ldiamond | It's not like I don't google before asking here lol. Try and search for SIP provider SMS, nothing helpful. |
| 05:05.39 | ldiamond | and its not like this channel is high traffic. Why would it matter if I ask questions not related to asterisk but related to SIP |
| 05:05.45 | ldiamond | its not like theres a #sip channel. |
| 05:05.48 | roberob | ChannelZ: my vitelity inbound is free.. |
| 05:05.59 | ldiamond | worst case is I don't get an answer here... |
| 05:06.15 | roberob | ldiamond: i googled it earlier when you asked and saw something that may be useful |
| 05:06.36 | roberob | https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-June/009576.html |
| 05:06.41 | roberob | just glanced though.. looks old ;x |
| 05:07.24 | ldiamond | yea I saw that, it's not really what I'm looking for tho |
| 05:07.58 | ldiamond | thx tho |
| 05:08.33 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 05:09.15 | p3nguin | roberob: How did you manage to pull off free calls? |
| 05:10.07 | roberob | free inbound.. |
| 05:10.24 | roberob | and then .0144/min |
| 05:10.26 | roberob | for outbound |
| 05:10.34 | roberob | no idea, thought that was the rate ? |
| 05:10.59 | roberob | wait nevermind |
| 05:11.08 | roberob | i pay 7.95/mo for unlimited |
| 05:11.14 | roberob | forgot about that ;x |
| 05:15.13 | ChannelZ | :) |
| 05:15.14 | roberob | based on the amount of calls i receive i need to switch my plan... |
| 05:15.18 | roberob | doing the math ehre ;x |
| 05:15.19 | roberob | here* |
| 05:16.02 | ChannelZ | that's what I did. It's close to 600 minutes I'd need to use/mo to hit the $7.95 |
| 05:17.25 | ChannelZ | The only thing I noticed with Vitelity, they say everywhere they support IAX but don't give any instructions, just SIP. It doesn't matter to me really, SIP is fine, but did they dump IAX? |
| 05:19.02 | roberob | not a clue, tbqh |
| 05:19.13 | roberob | i've only been using them for a couple of months, basic SIP setup |
| 05:21.31 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 05:22.00 | p3nguin | VoIP.ms provides IAX2. |
| 05:22.03 | p3nguin | I use it. |
| 05:23.17 | ChannelZ | I should just try registering with my credentials and see what happens |
| 05:24.22 | p3nguin | You're just doing termination only right now? |
| 05:25.12 | *** join/#asterisk defslap (~andy@defsdoor.gotadsl.co.uk) |
| 05:27.45 | ChannelZ | yeah. I just signed up last night at like 2am. Was just playing/testing with outbound today and then got my SPA3102 setup on the FXS side to hook to the phones in my house. Seemed to go well so I put in a port order for my POTS number |
| 05:28.34 | ChannelZ | hmm it let me register via iax :) |
| 05:28.53 | carrar | we can't let you do that |
| 05:29.12 | p3nguin | I can send calls outbound via IAX2 or SIP depending on my mood, but the DID has to be set to be sent to me via one or the other and that's what requires the registration. |
| 05:30.28 | *** join/#asterisk autojack (~owen@nerdnetworks.org) |
| 05:31.18 | ChannelZ | yeah thats the part I'm not sure of since I have no DID at the moment. Have to see what it asks once I do. |
| 05:31.27 | autojack | I recently switched to Flowroute for outbound VOIP termination. it's been working well so far, but right now my calls are failing with this error: Got SIP response 486 "Busy here" back from 70.167.153.130 does this actually mean the number I am dialing is engaged? I'm not getting a busy signal back. |
| 05:31.45 | autojack | calls to one specific number, that is. |
| 05:33.07 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 05:36.07 | *** join/#asterisk Kyosh (whoa@pool-74-108-19-33.nycmny.fios.verizon.net) |
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| 05:39.36 | tessier | So I'm debating NAT with a friend who insists that NAT is fine and that SIP sucks. |
| 05:40.06 | p3nguin | What I meant was that you don't need to register to send calls via IAX2 (or SIP for that matter). |
| 05:40.23 | p3nguin | So if you were registering on SIP and felt like making a call via IAX2, it would still work. |
| 05:41.32 | tessier | The claim is that if we would simply not dynamically assign ports and statically assign a destination port for each device and all of our NAT problems would go away. |
| 05:41.37 | tessier | What is wrong with this picture? |
| 05:49.15 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 05:49.34 | ChannelZ | Well it would go away for a single computer on the other side of the nat I guess.. |
| 05:50.22 | ChannelZ | It's one of the arguments for IPv6 though, get rid of NAT.. let everyone have 10 or 100 or however many IPs they need so we don't need NAT |
| 05:51.29 | *** part/#asterisk autojack (~owen@nerdnetworks.org) |
| 05:51.47 | p3nguin | I use IPv6 when I can, but not everything has support for it currently. |
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| 06:27.21 | shamelessn00b | hey, anyone here wanna try out a dial plan building tool I wrote for asterisk |
| 06:28.08 | WIMPy | What are the requirements? And what are the benefits? |
| 06:28.39 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 06:28.41 | shamelessn00b | you visualize your call flow in form of a graph |
| 06:28.54 | shamelessn00b | then populate a database, with nodes and edges |
| 06:29.17 | shamelessn00b | there are 4 types of edges |
| 06:29.43 | shamelessn00b | no-op, where simply your call flow goes to another node without any operation |
| 06:31.24 | shamelessn00b | no-op with value, where a set of commands written on a specific node are repeated specified number of times, before transfering control to the next node |
| 06:31.54 | shamelessn00b | dtmf, where control is transferred to the specified node when the specified dtmf is pressed |
| 06:32.00 | shamelessn00b | and dynamic |
| 06:32.38 | shamelessn00b | dynamic round trip rather |
| 06:33.06 | shamelessn00b | I'll provide you some example dialplans |
| 06:33.11 | shamelessn00b | to get you started |
| 06:33.26 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:33.44 | shamelessn00b | it saves you a lot of effort as you dont have to write execif's, bacground() and loops |
| 06:34.27 | *** join/#asterisk Yonn (~Yon@212.247.19.244) |
| 06:35.25 | WIMPy | is never so sure if such utilities really make it easier or the opposite. But you didn't specify the requirements, yet. |
| 06:37.17 | *** part/#asterisk gelo (~gelo@mx01.quobis.com) |
| 06:37.23 | shamelessn00b | The only requirement is that you design a graph |
| 06:37.32 | shamelessn00b | for your call flow, and populate some tables in a database |
| 06:37.44 | shamelessn00b | instead of writing extensions |
| 06:38.28 | WIMPy | What do you need to runb it? Except for a database. |
| 06:38.32 | WIMPy | -b |
| 06:38.37 | shamelessn00b | the tool |
| 06:38.45 | shamelessn00b | which generates extensions file |
| 06:38.49 | shamelessn00b | from the database |
| 06:38.52 | *** join/#asterisk BANSAL (~bansal@117.199.114.84) |
| 06:38.53 | shamelessn00b | and adds the missing lines |
| 06:38.55 | shamelessn00b | etc |
| 06:40.25 | WIMPy | Nevermind. I'm out anyway. |
| 06:40.42 | shamelessn00b | ok |
| 06:40.54 | shamelessn00b | its about 320 lines of code, groovy ftw |
| 06:41.10 | shamelessn00b | really saves a LOT of effort programming in groovy |
| 06:41.25 | WIMPy | Sounds extremely short. |
| 06:41.48 | shamelessn00b | yeah but if you convert it in java, it would atleast be 2000+ lines |
| 06:41.49 | WIMPy | Is that groovy a language? |
| 06:42.16 | shamelessn00b | an extension to java |
| 06:42.32 | shamelessn00b | takes 2 hrs to get cracking |
| 06:42.59 | WIMPy | Ah. Java. So it could actually be run if you have a jre. |
| 06:43.19 | shamelessn00b | yes |
| 06:43.30 | shamelessn00b | that and the groovy plugin |
| 06:44.08 | shamelessn00b | and the database, cuz its basically manipulating the database to generate its output |
| 06:45.13 | WIMPy | Sounds like far too much effort for me. |
| 06:46.13 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:48.51 | shamelessn00b | idk, if you a spreadsheet to populate whatever data thats needed for the program to run, and then export to csv and load in the database |
| 06:49.03 | shamelessn00b | really reduces a lot of effort |
| 06:50.38 | shamelessn00b | compared to writing extensions.conf and less chances for errors too |
| 06:53.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:57.25 | *** join/#asterisk zoap (~zoap@53501AB4.static.ziggozakelijk.nl) |
| 07:02.11 | zoap | hi there. trying to send my callerid with outgoing calls here. when i set Set(CALLERID(num)=XXXXXXXXX) on outgoing lines, i see it is being set in the cli, but the correct one is not being sent |
| 07:02.43 | zoap | my telco tells me i can send any number within my block, so i guess i'm doing something wrong here... |
| 07:03.09 | zoap | i use a T1 |
| 07:03.25 | shamelessn00b | zoap: what number is being sent |
| 07:03.40 | shamelessn00b | your sip iD? |
| 07:03.44 | *** join/#asterisk slima (slima@unaffiliated/slima) |
| 07:04.07 | zoap | the default number |
| 07:04.36 | zoap | which is the main number we got from our telco in a range of 100 numbers. so only the first one |
| 07:04.59 | WIMPy | zoap: Have you tried different formats? Some telcos only accept a certain format. possibly in the correct combination with the number type. |
| 07:05.01 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 07:05.11 | Micc | wtf, dahdi won't load again. |
| 07:05.25 | zoap | tried different formats yes |
| 07:05.56 | WIMPy | Have you also tried setting the Type Of Number? |
| 07:06.03 | zoap | i guess i need custom callerid feature somewhere, but do not know where to set it |
| 07:06.09 | WIMPy | Or dialpan as dahdi calls it? |
| 07:06.22 | zoap | Type Of Number is set where? |
| 07:06.47 | WIMPy | In the dahdi config. |
| 07:07.01 | WIMPy | IIRC CALLERID(TON) is read-only. |
| 07:07.52 | WIMPy | Maybe someone can give you a better hint if you tell us what Telco you're connected to. |
| 07:08.11 | zoap | Tele2 |
| 07:11.56 | WIMPy | zoap: You could always ask then what format they expect. |
| 07:12.02 | WIMPy | them |
| 07:12.04 | Micc | when I run service dahdi restart, what does this mean? /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting |
| 07:12.39 | zoap | WIMPy: guess i will go and give those guys a call yeah... :) thanks |
| 07:14.38 | Micc | last night when I had dahdi working it showed 50 used for dahdi module when I ran lsmod |
| 07:14.46 | Micc | now its only 3 and its not working. |
| 07:16.44 | Micc | something must not be loading in the right order. |
| 07:19.26 | zoap | ok, it works. apparently i made a typo in the outgoing number.... :/ it expects 9 numbers within our range. fixed. thank you guys |
| 07:23.27 | *** join/#asterisk Da-Geek (~Da-Geek@gw0.tieturi.com) |
| 07:23.42 | Micc | what module has this application in it? ast_register_application2: Already have an application 'DAHDISendKeypadFacility' |
| 07:26.07 | kaldemar | Micc: chan_dahdi.so |
| 07:26.34 | Micc | yeah, I'm trying to load chan_dahdi, but its saying its already registered. |
| 07:27.36 | Micc | should I put chan_dahdi in preload in modules.conf? |
| 07:27.47 | Micc | I had this working last night. I forget what it was that did it though. |
| 07:28.16 | kaldemar | maybe chan_dahdi is already loaded... |
| 07:29.23 | Micc | if it is, its not registering any dahdi cli commands. no dahdi show version, no dahdi restart, dahdi ? does nothing. |
| 07:30.13 | Micc | This wsa working until I rebuilt asterisk again and installed. |
| 07:30.23 | Micc | maybe I removed one of the modules. |
| 07:30.27 | kaldemar | module unload chan_dahdi.so, then module load chan_dahdi.so and pastebin the output. |
| 07:31.03 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 07:31.32 | Micc | ah, it was partially loaded I guess. |
| 07:31.35 | Micc | pri_create_trunkgroup: Failed go get span information on channel 24 (span 1): Inappropriate ioctl for device |
| 07:32.39 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:33.29 | Micc | I seem to remember a problem with the kernel modules loading. |
| 07:36.43 | Micc | ah, maybe its this blacklist file. |
| 07:39.38 | WIMPy | wooot? musiconhold requires dahdi??? |
| 07:39.47 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 07:40.01 | WIMPy | Seems to do fine without in practice, however. |
| 07:40.18 | Micc | Ok, I think it has something to do with this error. /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting |
| 07:41.11 | *** join/#asterisk TimeRider (steve@5ac318c4.bb.sky.com) |
| 07:41.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:43.38 | Micc | I have no /sys/bus/astribanks directory |
| 07:43.38 | Micc | what module creates that directory? |
| 07:43.39 | Micc | or those devices? |
| 07:46.04 | Micc | dahdi_hardware says this: pci:0000:01:01.0 wcte11xp+ e159:0001 Digium Wildcard TE110P T1/E1 Board |
| 07:47.39 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:49.02 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 07:49.25 | WIMPy | Oh, I suddenly get a fancy menuselect. What did I install to get that? |
| 07:53.08 | kaldemar | asterisk has menuselect by default. |
| 07:53.51 | WIMPy | Yes, but so far it used to be B/W, now it's suddenly in colour. |
| 07:55.49 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:02.20 | WIMPy | Ugh. |
| 08:02.42 | WIMPy | Are there any known issues with recent gcc or binutils versions? |
| 08:03.34 | WIMPy | I already had issues on my laptop, but now I also got a defunct Asterisk on the big one. |
| 08:03.56 | WIMPy | It seems to stop loading moduels at startup at some point. |
| 08:05.25 | WIMPy | If I try to load modules manually, others cease to work. |
| 08:06.34 | WIMPy | core show channeltypes only shows me OOH323 and IAX2. Wehn I module load chan_sip, core show channeltypes becomes empty. |
| 08:08.19 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
| 08:09.06 | WIMPy | And Asterisk can only be killed -9. |
| 08:19.16 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 08:20.03 | WIMPy | Ok, time to dig out the backup. |
| 08:23.06 | Micc | dahdi_tool shows nothing, but dahdi_cfg -v shows my card and span. |
| 08:23.47 | Micc | how can I load all dahdi drives with modprobe at once? |
| 08:26.30 | *** join/#asterisk renshen (~renshen@93-63-217-144.ip29.fastwebnet.it) |
| 08:28.00 | Micc | could this mean anything? IRQ 19/wcte11xp: IRQF_DISABLED is not guaranteed on shared IRQs |
| 08:29.31 | *** join/#asterisk jezier (~jezier@nat.zgierz.net.pl) |
| 08:31.13 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:31.41 | jezier | hi.. is it possible to configure queue to call some agents after timeout(in seconds) ... for example... first try calling helpdesk agents for 30s... and after 30s still call helpdesk + receptionists |
| 08:35.20 | Micc | when I run lsmod it should say something is using my wcte11xp, right? but it says 0 use |
| 08:40.10 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 08:50.31 | Micc | I'm about at my wits end with this dahdi problem. |
| 08:51.40 | *** join/#asterisk Weazel (~deazel@net.kolcore.com) |
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| 08:53.06 | Micc | Its not loading the right kernel modules from what I can tell. |
| 08:55.24 | Micc | It was working fine before I rebuilt asterisk. So the modules must have loaded right and it must be chan_dahdi thats different. |
| 08:57.16 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
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| 09:06.03 | hajkym | hello |
| 09:06.20 | hajkym | can me sommebody explain some things? |
| 09:07.49 | hajkym | when i want create call through asterisk i have to use telephone number which is in extensions.conf |
| 09:08.57 | hajkym | but when call incoming no telephone number is receive in INVVITE method...but sip uri |
| 09:09.57 | hajkym | A --> B, INVITE sip:2222@192.168.10.123 SIP/2.0 |
| 09:10.27 | hajkym | and B receive: INVITE sip:pc2@192.168.32.17:5066 SIP/2.0 |
| 09:10.40 | hajkym | is possible use only telephone number? |
| 09:11.07 | hajkym | or only sip uri? |
| 09:11.19 | kaldemar | asterisk does what you tell it to. |
| 09:12.05 | kaldemar | show a cli output of a call. |
| 09:12.36 | hajkym | cli output? |
| 09:12.46 | hajkym | cli is condole |
| 09:12.57 | hajkym | but what command |
| 09:13.02 | kaldemar | asterisk's console, yes. |
| 09:13.07 | kaldemar | core set verbose 10 |
| 09:13.14 | kaldemar | then make a call and pastebin the output. |
| 09:13.17 | kaldemar | ~pb |
| 09:13.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 09:13.37 | tuxx- | woah |
| 09:13.41 | tuxx- | 11 seconds responsetime |
| 09:13.41 | tuxx- | xD |
| 09:13.52 | tuxx- | infobot was still sleeping i guess |
| 09:14.22 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
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| 09:15.11 | hajkym | http://pastebin.org/282644 |
| 09:16.07 | hajkym | call number and sip uri are suported in sip ? |
| 09:16.20 | hajkym | i thing if call number is not only asterisk feature |
| 09:16.27 | hajkym | think |
| 09:16.56 | kaldemar | hajkym: your dial line is Dial(SIP/pc2), use Dial(SIP/pc2/${EXTEN}) |
| 09:17.28 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 09:20.52 | hajkym | kaldemar: and what it should do |
| 09:21.12 | hajkym | i get still uri number when incommin call receive |
| 09:21.26 | hajkym | sip uri no uri number "0 |
| 09:21.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:22.00 | hajkym | is possible use only numbers or use only sip uri? |
| 09:22.09 | kaldemar | hajkym: ${EXTEN} has what ever the extension is. if the extension is 2222@skuska, Dial(SIP/pc2/${EXTEN}) will translate to Dial(SIP/pc2/2222), i.e. asterisk will send number 2222 to the peer. |
| 09:23.00 | kaldemar | with SIP, it is always a URI. |
| 09:23.41 | hajkym | so using call number is feature PBX? |
| 09:23.55 | kaldemar | you're losing me. |
| 09:25.25 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 09:28.51 | hajkym | extension works |
| 09:30.01 | hajkym | and is possible send in header: From: "2222" <sip:2222@192.168.10.123>;tag=as516170f0 instead: From: "pc2" <sip:pc2@192.168.10.123>;tag=as516170f0 |
| 09:31.05 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 09:32.20 | WIMPy | Interesting. Now I hvae three up-to-date systems. One can build a working asterisk, the other two can't. |
| 09:33.29 | WIMPy | The one that can has a very minimal installation, the other two are full installs, so I suspect it's something optional. |
| 09:34.27 | WIMPy | Oh, wait. It's an old Asterisk as well. |
| 09:36.46 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
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| 09:47.17 | WIMPy | Ok, it's the old version. 1.4.21.2 is ok, 1.6.2.0 and 1.6.2.8 are not. |
| 09:53.33 | *** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr) |
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| 10:03.39 | Micc | so I have some kind of kernel driver ioctl structure mismatch. |
| 10:04.07 | Micc | which causes this error, "Inappropriate ioctl for device" when I try to load chan_dahdi |
| 10:04.16 | Micc | But the document doesn't say how to fix it. |
| 10:04.30 | Micc | I've rebuilt the kernel drivers as far as I can tell. |
| 10:06.00 | tzafrir_laptop | Micc, what version of dahdi? |
| 10:06.08 | Micc | 2.3.0 |
| 10:06.12 | tzafrir_laptop | any idea vs. what version it was built? |
| 10:06.39 | Micc | previous version was 2.2.1 I think |
| 10:06.53 | Micc | or even 2.2.0 |
| 10:07.01 | Micc | 2.2.0.2 actually |
| 10:07.42 | Micc | now when I try to build the new version I get this error, "WARNING: could not find /usr/src/asterisk/dahdi-linux-2.3.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/asterisk/dahdi-linux-2.3.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o" |
| 10:08.00 | Micc | which looking it up on google I found its suppose to be a problem with the path. |
| 10:08.06 | Micc | But that is not the problem at all here. |
| 10:08.12 | Micc | then they said it could be ignored. |
| 10:08.29 | Micc | well it looks like my build is failing because of it possibly. |
| 10:10.30 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:11.25 | *** join/#asterisk ecolitan (~aaron@li57-124.members.linode.com) |
| 10:11.32 | WIMPy | Surprise: Trunk won't work, either. |
| 10:13.02 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 10:21.48 | Micc | make install seems to finish fine. |
| 10:24.01 | Micc | should I do a make uninstall in previous dahdi version? |
| 10:25.06 | Micc | Its 3:30am here, and I can't go to sleep till I get this working. |
| 10:28.45 | Micc | I had this same problem the other night and I fixed it, but I don't remember how. |
| 10:29.47 | Micc | I think I remember, but it doesn't seem to be working this time, which makes me think I didn't quite understand the problem in the first place. |
| 10:32.19 | *** join/#asterisk Z_God (~julius@wlan234013.mobiel.utwente.nl) |
| 10:33.57 | tzafrir_laptop | Micc, no need to explicitly uninstall |
| 10:34.54 | Micc | I guess I'm going to try to go back to 2.2.0 |
| 10:35.16 | tzafrir_laptop | I believe that this file should have been downloaded by the firmware downloader target |
| 10:35.25 | Micc | I remember when modules loaded when it worked it seemed like there were much more of them in use. |
| 10:35.30 | tzafrir_laptop | (at build time) |
| 10:35.39 | tzafrir_laptop | was there any error downloading anything? |
| 10:35.47 | tzafrir_laptop | also, maybe try: make dist-clean |
| 10:35.53 | tzafrir_laptop | and re-run 'make' |
| 10:35.58 | Micc | not that I can see. |
| 10:38.51 | Micc | here is what load chan_dahdi says |
| 10:38.52 | Micc | http://pastebin.com/j3H9sVKb |
| 10:39.30 | Micc | is this normal? /bin/sh: /usr/bin/sw_vers: not found |
| 10:40.17 | Micc | maybe I need to rerun configure after installing a new dahdi version so asterisk can find it. |
| 10:40.42 | Micc | does ./configure look for where dahdi headers are? |
| 10:42.55 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 10:42.58 | Micc | dahdi_cfg -vv looks normal. |
| 10:43.43 | Micc | I've got a customer to see in the morning. I'm gonna be so dead tired. |
| 10:44.36 | Micc | at least I know I can call digium for support in the morning since its their card. |
| 10:45.25 | Micc | did I use the right their there they're? I'm so tired I don't know what I'm saying. Bad time to be working on production servers. |
| 10:51.57 | Micc | ok, it looks like it redownloaded the vmad thing |
| 10:52.04 | Micc | after make dist-clean |
| 10:52.11 | Micc | then make |
| 10:52.24 | Micc | now it might be building right. |
| 10:52.26 | Micc | we'll see. |
| 10:52.49 | Micc | I tried going back to previous versions that should have worked. |
| 10:53.47 | Micc | still giving me same error about vmad not finding. |
| 10:55.01 | Micc | when I run lsmod shouldn't my wcte11xp have at least 1 use? |
| 10:55.18 | Micc | dahdi says 1 wcte11xp |
| 10:55.47 | Micc | so wcte11xp is being used by dahdi, but nothing is using wcte11xp |
| 10:56.32 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 10:56.51 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:00.34 | Micc | oh yes! |
| 11:00.45 | Micc | not sure what I did again, but its working. |
| 11:00.53 | Micc | going to try to rebuild asterisk again. |
| 11:02.17 | Micc | now lsmod shows used 50 |
| 11:02.41 | Micc | for dahdi and 24 for wcte11xp |
| 11:02.50 | Micc | probably because asterisk is using it. |
| 11:02.55 | Micc | that makes sense. |
| 11:03.25 | Micc | now its broken again |
| 11:03.48 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 11:03.48 | *** mode/#asterisk [+o file] by ChanServ |
| 11:05.42 | Micc | so building asterisk against old dahdi then installing new dahdi seems to work. |
| 11:05.46 | Micc | but that makes no sense. |
| 11:05.57 | Micc | going to run configure in asteirsk again |
| 11:06.54 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 11:07.39 | Micc | sometimes I wonder why I go through all this. |
| 11:07.49 | angryuser | I am looking for the Yealink SIP-T26P tftp configuration file syntax, nothing on vendor site, anyone have it ? |
| 11:09.25 | Micc | angryuser, I haven't played with one of those phones yet. |
| 11:10.51 | angryuser | found ot |
| 11:10.52 | Micc | Our distributor sent me a new panasonic kx-tgp550 to play with. I wonder who else got them? It came with a interoperability checklist form to send back to them. |
| 11:10.52 | angryuser | it |
| 11:11.03 | *** join/#asterisk Tako-san (~Tako-san@p3192-ipad414osakakita.osaka.ocn.ne.jp) |
| 11:12.46 | Micc | angryuser, let me know how you like it. I might have to order one to play with. Our default cheap phone is an aastra 6730i, but I'd be open to trying the Yealink if it was easier to use for the end user. |
| 11:13.48 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 11:14.58 | angryuser | Micc, it is better then 30i that's for sure |
| 11:15.12 | angryuser | lcd is a bit small |
| 11:15.22 | angryuser | all the rest is simple |
| 11:16.08 | Micc | I like the 30is, except they are confusing to customers in some cases. |
| 11:16.14 | angryuser | Micc, config manual is here :http://www.google.fr/url?sa=t&source=web&ct=res&cd=3&ved=0CCcQFjAC&url=http%3A%2F%2Ftrack.sipfoundry.org%2Fsecure%2Fattachment%2F23242%2FAuto%2BProvision%2BManual%2Bversion%2B1.2.3.odt&rct=j&q=y000000000004.cfg&ei=FgH9S__1OOOL4ga9-IGiCw&usg=AFQjCNHGhvc5trOyEHmnNGEEpfz_iPdIxQ |
| 11:16.23 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 11:16.25 | DelphiWorld | hi all |
| 11:16.29 | angryuser | Dont ask me why |
| 11:16.36 | DelphiWorld | any config example for digium T120 card for dahdi? |
| 11:19.09 | DelphiWorld | tzafrir_laptop? |
| 11:19.14 | Micc | dangit! still some mismatch somewhere with dahdi structures. Failed go get span information on channel 24 (span 1): Inappropriate ioctl for device |
| 11:19.14 | tzafrir_laptop | busy |
| 11:19.34 | tzafrir_laptop | dahdi_genconf should generate a config example... |
| 11:21.00 | *** join/#asterisk michael-i (~michael-i@141.41.40.121) |
| 11:21.37 | *** join/#asterisk DJ_HaMsTa (~carlos@unaffiliated/dj-hamsta/x-2342346) |
| 11:21.38 | DJ_HaMsTa | I have a DID from les.net, any other better providers that i can try? |
| 11:21.52 | Micc | so I guess I will just have to live with the old dahdi, would a reboot maybe help? |
| 11:22.27 | DelphiWorld | DJ_HaMsTa: what country? |
| 11:22.35 | DJ_HaMsTa | USA |
| 11:23.21 | DelphiWorld | DJ_HaMsTa: http://www.flowroute.com |
| 11:23.26 | DelphiWorld | DJ_HaMsTa: http://www.voxbone.com |
| 11:23.30 | DelphiWorld | DJ_HaMsTa: http://www.ipkall.com |
| 11:23.58 | DJ_HaMsTa | ipkall is free ? |
| 11:24.14 | DelphiWorld | DJ_HaMsTa: yes |
| 11:24.21 | DJ_HaMsTa | oh lawrd |
| 11:24.45 | DJ_HaMsTa | and they give me a DID so i can add it to my ip phone SPA941 ? |
| 11:26.12 | Micc | sounds like a non did service. |
| 11:26.17 | Micc | just IP dialing |
| 11:26.26 | DJ_HaMsTa | thats what i have thought all along |
| 11:26.45 | DJ_HaMsTa | i did see a part that they offered DID but no idea how to set it up |
| 11:28.07 | DelphiWorld | i have allready a did from ipkall |
| 11:28.29 | DJ_HaMsTa | is it good quality ? |
| 11:28.59 | Micc | still inappropriate ioctl for device. |
| 11:29.00 | TSM | does the g729 codec licences that come with S4A allow you to do normal transcoding or are they tied directly to S4A |
| 11:29.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:29.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:30.07 | Micc | I'm not sure what I have to do to fix this inapporpiate ioctl for device problem, but I think its beyond me at this point. |
| 11:30.37 | Micc | I've done all I can think of and still no luck. I'm going to have to build older version of dahdi I think. |
| 11:31.36 | leifmadsen | Micc: a new version of DAHDI will be out very soon and that sounds familiar... |
| 11:31.48 | leifmadsen | have you tried the latest from the branches? |
| 11:32.08 | Micc | I was just trying 2.3.0 |
| 11:32.54 | leifmadsen | right, which is why I'm saying, "Have you tried the branches to see if your problem has already been fixed"? |
| 11:33.21 | Micc | no, which one would you suggest? |
| 11:33.22 | DJ_HaMsTa | DelphiWorld, http://phone.ipkall.com/ what am i supposed to put for SIP Phone number and proxy? |
| 11:33.32 | leifmadsen | new DAHDI is out soon, and I saw something about IO stuff in the draft release announcement |
| 11:35.11 | *** join/#asterisk LeandroAugusto (~leandro.a@189.90.54.130) |
| 11:35.51 | Micc | so should I try branch 2.3? |
| 11:36.08 | Micc | I only see 2.2 and 2.3 in branches at http://svn.digium.com/svn/dahdi/linux/branches |
| 11:36.09 | leifmadsen | yes, that is what I said :) |
| 11:36.22 | leifmadsen | and dahdi/tools/branches |
| 11:36.32 | Micc | ok. wasn't sure if it would have a new version number or not. |
| 11:37.05 | leifmadsen | well there is DAHDI 2.3.0+2.3.0 which means DAHDI linux 2.3.0 + tools 2.3.0 |
| 11:38.14 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 11:39.54 | Micc | would I need to rebuild asterisk too? |
| 11:42.13 | *** join/#asterisk Sweeper (~ppp@74.51.109.60) |
| 11:42.16 | Micc | if this doesn't work, i'll have to go back to what works and call it a night. |
| 11:42.25 | Micc | its almost 5am here. |
| 11:42.31 | Sweeper | so what's the preferred method of semi-complex datetime based dialing? |
| 11:42.45 | Sweeper | I have a support line with several oddball shifts |
| 11:43.05 | Sweeper | I'd like to define a range and an extension to dial when datetime falls in that range |
| 11:43.07 | DelphiWorld | hi leifmadsen |
| 11:43.14 | Micc | Sweeper, put it in a database where it can be changed easily. |
| 11:43.53 | DelphiWorld | leifmadsen: any dahdi example for E1 PRI span? |
| 11:44.12 | leifmadsen | DelphiWorld: I don't have any examples for you, sorry |
| 11:44.15 | Sweeper | k, that's not really what I'm looking for, what I'm looking for is the dialplan method of routing |
| 11:44.34 | Sweeper | storing the schedule is fairly independant of that... |
| 11:44.58 | leifmadsen | Sweeper: GotoIfTime() |
| 11:45.02 | Micc | oh, just use GotoIfTime then. |
| 11:45.07 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 11:45.08 | Sweeper | ah, great :D |
| 11:45.19 | Micc | I was thinking more complex. :) |
| 11:45.59 | Micc | I give my customers default preprogrammed hours of operation, but then give them extensions to override open/closed/default |
| 11:46.21 | Micc | because they always forget to tell me about some holiday or something. |
| 11:47.42 | Sweeper | gotcha. database support would be nice...eventually...but it'd take me as long to just set up the database connection and schema as it would to just use gotoiftime |
| 11:47.42 | Micc | leifmadsen, no good, same problem. but I could be missing a step. I'll be sure to read the release notes when the new version does come out to see if it addresses my problem. |
| 11:48.24 | Micc | leifmadsen, from what I read on http://docs.tzafrir.org.il/dahdi-linux/ its a dahdi structures mismatch in the kernel drivers. |
| 11:48.27 | leifmadsen | okie |
| 11:48.29 | Sweeper | prepends tt-weasels to outbound calls for a day |
| 11:48.31 | leifmadsen | ya perhaps |
| 11:52.54 | *** join/#asterisk n3hxs (~HAMming@173-162-253-153-NewEngland.hfc.comcastbusiness.net) |
| 11:53.56 | Micc | now I'm haivng a problem getting back to any working version of dahdi. |
| 11:54.04 | Micc | it seems to finiky. |
| 11:57.05 | Micc | I think the right order to build in is make dist-clean; make ; make install in dahdi, then dahdi restart then make dist-clean in asterisk and configure and make ; make install asterisk. |
| 11:57.11 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 11:57.13 | DelphiWorld | alarms=BLU/RED |
| 11:57.17 | DelphiWorld | while runing dahdi_scan |
| 11:57.21 | DelphiWorld | what is this status? |
| 11:58.37 | Micc | I was going to say something stupid, but I'm too tired for making jokes. |
| 11:59.10 | leifmadsen | Micc: ya sorry, I don't use hardware (pretty much SIP only) so I'm not sure how else to help |
| 11:59.47 | Micc | leifmadsen, at least I can call digium soon if I still can't get it working. |
| 11:59.56 | leifmadsen | ya that's what I'd do probably |
| 12:00.13 | leifmadsen | sometimes faster to try and get it working yourself unless you start spinning tired |
| 12:00.14 | leifmadsen | tires* |
| 12:00.17 | *** part/#asterisk gelo (~gelo@mx01.quobis.com) |
| 12:00.31 | Micc | I've had it a couple times tonight, but I kept changing things to see if I could figure it out. |
| 12:00.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:01.01 | Micc | I hate making something work, but not know why it works. |
| 12:01.07 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:01.12 | leifmadsen | ya :) |
| 12:01.45 | Micc | I like it when things just work too, but you know what I mean. |
| 12:01.45 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:02.46 | leifmadsen | guess there isn't a way to make a Polycom failover to another service provider with a different registration authentication? |
| 12:02.55 | Micc | ok, got it with 2.2.1 |
| 12:03.34 | leifmadsen | i.e. I'm registering line 1 with my Asterisk with X set of credentials, and then if Asterisk dies, I want my phone to register directly to my ITSP with Y set of credentials |
| 12:03.38 | leifmadsen | not sure I can do that |
| 12:04.32 | Micc | leifmadsen, I would be surprised if there isn't some way to make it do that. Those polycom config files are the craziest things. |
| 12:04.42 | leifmadsen | ya... hmmmm |
| 12:05.01 | leifmadsen | wonder if I can just do reg.1.server.2.auth.userID |
| 12:05.05 | leifmadsen | will have to look in the manual |
| 12:05.36 | [TK]D-Fender | leifmadsen: Should be able to. The cred and IP, etc are all completely separate. |
| 12:05.47 | Micc | I'm off to bed finally. Gotta get up in 2 hours. |
| 12:06.21 | leifmadsen | [TK]D-Fender: for example I see reg.1.server.1.expires and reg.1.server.2.expires |
| 12:06.53 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 12:07.00 | leifmadsen | it's just that I only see reg.1.address for example, so I wonder if reg.1.server.2.address is valid |
| 12:07.08 | [TK]D-Fender | leifmadsen: IIRC thats reg expiry... i suppose if you se that low enough and it fails to contact #1, it should logicall try to jump to #2 |
| 12:07.38 | [TK]D-Fender | set* |
| 12:07.40 | leifmadsen | right, that was just an example of something that does have reg.1.server.2 -- I just don't know if using something like reg.1.server.2.address is valid |
| 12:08.00 | leifmadsen | because on the server.2 failover I want it to use a totally different authentication |
| 12:08.01 | [TK]D-Fender | leifmadsen: yup |
| 12:08.25 | leifmadsen | yup what? |
| 12:08.34 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:09.10 | TSM | can new asterisk modules only be loaded when you do a restart |
| 12:09.16 | leifmadsen | no |
| 12:09.20 | leifmadsen | 'module load' |
| 12:09.30 | TSM | i cant see my new chan_skype module |
| 12:09.42 | [TK]D-Fender | TSM: is it in the folder with the right perms? |
| 12:09.44 | leifmadsen | [TK]D-Fender: yup what? |
| 12:11.06 | *** join/#asterisk gr0mit (~tim@90.155.53.9) |
| 12:11.22 | TSM | [TK]D-Fender: its in /usr/lib/asterisk/modules/ with 755 perms as the installer did it |
| 12:11.25 | [TK]D-Fender | leifmadsen: Actually.. scratch that.. only has sever location details, not full user auth |
| 12:11.40 | [TK]D-Fender | TSM: And when you try to load it manually? |
| 12:12.21 | TSM | [TK]D-Fender: module load chan_skype.so , it says nothing but i dont get the skype option in the cli |
| 12:12.55 | [TK]D-Fender | TSM: Have you registered your licenses, etc? |
| 12:13.32 | TSM | [TK]D-Fender: yup run the register util, it says that asterisk must be restarted but thought mabey there is a way around this |
| 12:13.54 | TSM | [TK]D-Fender: dont want to kick people off calls, unless i do a condrestart, does that exist? |
| 12:14.24 | [TK]D-Fender | TSM: "restart when convenient |
| 12:14.58 | TSM | [TK]D-Fender: ok, ile try |
| 12:19.05 | TSM | [TK]D-Fender: ahh seems that i have a x64 but the s4l install only thought i was running a 32 and did not install the chan_skype.so in the correct lib dir |
| 12:19.05 | *** join/#asterisk ecolitan (~aaron@li57-124.members.linode.com) |
| 12:21.19 | [TK]D-Fender | TSM: Please place the barrel firmly to the temple and stand on that plastic sheet over there, k? ;) |
| 12:21.32 | [TK]D-Fender | TSM: SILLY GOOSE |
| 12:21.54 | TSM | [TK]D-Fender: just copied the chan_skype.so and res_skypeforasterisk.so into the correct lib64 and then restarted asterisk and now it shows up |
| 12:22.12 | TSM | the installer needs fixing me thinks |
| 12:26.30 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 12:35.02 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 12:41.15 | *** join/#asterisk igorg (~igorg@net182.255.92-116.dynamic.omsk.ertelecom.ru) |
| 12:43.15 | *** join/#asterisk retentiveboy (~pdugas@atl.pra-corp.com) |
| 12:53.28 | *** join/#asterisk kartik (~koolkarti@117.207.82.244) |
| 12:54.54 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 13:07.31 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:13.30 | BarthezZ | hmm, I must be missing something... How can I get a account added using register => to an other context than default (for incoming calls from our uplink provider) |
| 13:14.07 | *** join/#asterisk gr0mit (~tim@90.155.53.9) |
| 13:16.48 | leifmadsen | BarthezZ: define a peer definition |
| 13:16.50 | Katty | [foo] |
| 13:16.54 | kaldemar | BarthezZ: sounds like the incoming calls don't match with any of your peer definitions. a register statement just lets your provider know where you are. peer definitions decide what contexts are used for calls. |
| 13:17.14 | leifmadsen | heh, 3 answers all saying the same thing :) |
| 13:17.25 | Katty | yeah but mine's the cutest. |
| 13:17.26 | Katty | :P |
| 13:17.26 | BarthezZ | hmm that's weird, got the peers added, but for some reason they don't seem to match in that case |
| 13:17.34 | kaldemar | we have different levels of verbosity. :) |
| 13:17.46 | leifmadsen | heads off to do bug tracker work |
| 13:17.57 | Katty | well fine, don't say hi )= |
| 13:18.03 | Katty | I"VE ONLY BEEN GONE FOR AN ENTIRE WEEK TO DALLAS |
| 13:18.06 | BarthezZ | hmm, i'll just add the extra variables in the register to the peer definition, maybe that helps |
| 13:18.10 | leifmadsen | Katty: hi :) |
| 13:18.25 | *** join/#asterisk gevaerts (~fg@rockbox/developer/gevaerts) |
| 13:18.35 | Katty | leifmadsen: hi (= |
| 13:18.39 | kaldemar | BarthezZ: look at "Naming devices" in the sample sip.conf. |
| 13:18.58 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
| 13:20.08 | BarthezZ | hmm, time to do sip debug to see what i'm missing in my peer definitions |
| 13:21.26 | *** join/#asterisk jasonjjohnsonjr (~jasonjjoh@asa1.jsmc.org) |
| 13:21.31 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:21.58 | jasonjjohnsonjr | Does anyone have any experience using Astribank USB channel banks? How do they compare with T1 connected CBs? |
| 13:22.25 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-vsdeuezywrlyejgc) |
| 13:22.32 | [TK]D-Fender | jasonjjohnsonjr: What are you exact requirements? |
| 13:24.33 | jasonjjohnsonjr | I have about 100 existing analog lines that I want to connect to my Asterisk Systems. We are trying to avoid recabling the rooms and replacing the phones so I was thinking of using channel banks. Trying to evaluate the different models. |
| 13:25.38 | BarthezZ | ; 2. Asterisk checks the IP address (and port number) that the INVITE |
| 13:25.38 | BarthezZ | ; was sent from and matches against any devices with type=peer |
| 13:25.46 | BarthezZ | what if I have 5 peers going to the same host? |
| 13:25.57 | BarthezZ | maybe that's why nothing matches, because everything matches 8)7 |
| 13:26.33 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:26.49 | kaldemar | if nothing matches, nothing really matches. if everything would match, you'd see one of them matching. |
| 13:27.06 | BarthezZ | hmm, than i'm quite clueless what i'm doing wrong |
| 13:28.01 | kaldemar | BarthezZ: pastebin your peer definitions and a CLI output of an incoming call with verbosity and sip debug enabled. |
| 13:28.15 | BarthezZ | yeah was working on that already :P |
| 13:28.23 | [TK]D-Fender | jasonjjohnsonjr: I'd probably sooner go with a SIP> FXS gateway for this |
| 13:28.47 | [TK]D-Fender | jasonjjohnsonjr: Wait... LINES? As in "to the telco"? |
| 13:28.54 | BarthezZ | just scrubbing out the passwords :P |
| 13:29.47 | BarthezZ | http://pastebin.com/RLZtppd5 @ kaldemar |
| 13:30.22 | jasonjjohnsonjr | [TK]D-Fender: No sorry if I was unclear. I have 100 rooms with analog phones that I need to connect to Asterisk. Our telco connectivity is PRI and already connected. |
| 13:30.54 | leifmadsen | [TK]D-Fender: he meant physical cables |
| 13:30.58 | [TK]D-Fender | jasonjjohnsonjr: Then SIP gateways it is |
| 13:32.06 | kaldemar | BarthezZ: your host definition doesn't match the ip address the call is coming from. |
| 13:32.30 | BarthezZ | hmm |
| 13:32.44 | BarthezZ | ah, the sip.sipnl.net has 3 ip addresses |
| 13:32.44 | *** join/#asterisk lantizia (~lantizia@217.154.146.153) |
| 13:33.05 | Gido-E | BarthezZ i have some servers, with those accounts to :-) |
| 13:33.22 | BarthezZ | ah, a fellow dutchman ^^ |
| 13:33.41 | Gido-E | BarthezZ yes. i wrote the pdf of voicecentrale.nl (dutch asterisk pdf book.) |
| 13:34.06 | jasonjjohnsonjr | [TK]D-Fender: SIP Gateway as opposed to channel banks? |
| 13:34.21 | BarthezZ | hmm kaldemar, if I would just add 3 host with just host=<ip> and context=Incoming would it than match everything to the good context? |
| 13:34.32 | BarthezZ | never heard of it Gido-E :p I'll check it out if it's any good ;) |
| 13:34.43 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 13:34.43 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:35.12 | [TK]D-Fender | jasonjjohnsonjr: Correct. No T1 card overhead, no USB potential flakeyness, inherent failover potential, reduced system loads an not need for funny drivers |
| 13:35.14 | leifmadsen | heh, I'm already seeing a table we used in our book in that book :) |
| 13:35.42 | Gido-E | leifmadsen i also, but it is in your book :-) |
| 13:35.44 | Katty | what causes allergies |
| 13:35.47 | kaldemar | BarthezZ: it will, if the ip address matches. remember you only can only set one host line per peer. |
| 13:36.16 | BarthezZ | yeah... but I'll just add 3 of them to just get the calls to the right context |
| 13:36.22 | BarthezZ | let's try and see what happens, thanks so far |
| 13:36.23 | Katty | does our body lack a certain hormone? |
| 13:37.03 | lantizia | Hey... we have a display (www.fop2.com) that can monitor QueuePaused being on or off... this is because QueuePaused is a part of Asterisk rather than additional AstDB variables that DND that don't actually exist (but we use it as a cheat to show if DND is on or off) |
| 13:37.08 | tzafrir_laptop | wonders what is that USB flakiness |
| 13:37.18 | lantizia | Can anyone think of in built variable like QueuePaused to be used in a similar way? |
| 13:37.31 | BarthezZ | Gido-E do you also work as asterisk guru on the dutch market? :p |
| 13:37.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:37.52 | [TK]D-Fender | lantizia: DND doesn't really exist |
| 13:37.55 | Gido-E | leifmadsen soon i wil write a chapter, you dont have :-). It's about a new FOP, only working good. |
| 13:38.00 | lantizia | [TK]D-Fender: I just said that |
| 13:38.20 | leifmadsen | Gido-E: that's good because we're not writing about any GUI stuff |
| 13:38.29 | leifmadsen | we're in the middle of writing the book from scratch |
| 13:38.32 | TSM | when you licence S4A does each licence enable a usage of the g729 codec, i cant see it working in the 'core show translations' |
| 13:38.50 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:39.14 | Baylink-work | leifmadsen: You guys need a techreader on that book? |
| 13:39.16 | BarthezZ | ah, seems to work kaldemar, let's see what happens when i also add the other 4 numbers/registrations |
| 13:39.46 | leifmadsen | Baylink-work: it will be available for OFPS at some point |
| 13:39.57 | leifmadsen | ~ofps |
| 13:40.18 | Baylink-work | Cool. Hadn't noticed that; must dive in. |
| 13:40.24 | Baylink-work | Naw; google's fine. |
| 13:40.26 | leifmadsen | infobot: ofps is the Open Feedback Publishing System by O'Reilly available at http://labs.oreilly.com/ofps.html |
| 13:40.27 | infobot | okay, leifmadsen |
| 13:40.49 | Baylink-work | I presume you have to be blessed to tell the bot stuff? |
| 13:40.49 | Gido-E | leifmadsen you work for or at Oreilly? |
| 13:41.01 | leifmadsen | I write for O'Reilly as an author |
| 13:41.09 | leifmadsen | Baylink-work: nope |
| 13:41.14 | leifmadsen | Baylink-work: anyone can do it |
| 13:41.17 | *** join/#asterisk LeandroAugusto (~leandro.a@189.90.54.130) |
| 13:41.20 | Baylink-work | Oh. Cool. |
| 13:41.39 | Baylink-work | Now, if I could just keep them straight. :-) (I think this bot uses ~, and the FreePBX bot uses ?) |
| 13:41.54 | leifmadsen | infobot uses ~ |
| 13:42.00 | leifmadsen | infobot: morning! |
| 13:42.01 | infobot | Good morning! |
| 13:42.13 | leifmadsen | infobot: how are you? |
| 13:42.13 | infobot | leifmadsen: you know how it is... |
| 13:42.16 | *** join/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 13:42.20 | leifmadsen | I sure do |
| 13:42.37 | leifmadsen | infobot: working hard, or hardly working? |
| 13:42.52 | BarthezZ | bows again for the great support in here :) |
| 13:43.03 | Baylink-work | I don't think they're using infobot. |
| 13:43.19 | Baylink-work | I don't see it in their user list |
| 13:43.33 | BarthezZ | hmm, but should I now file a bug report that a peer definition with a dns name with multiple records isn't matched? |
| 13:44.03 | leifmadsen | BarthezZ: no, it's not a bug |
| 13:44.07 | leifmadsen | it's a configuration issue |
| 13:44.13 | leifmadsen | setup 3 different peers |
| 13:44.18 | leifmadsen | for each IP address |
| 13:44.50 | BarthezZ | yeah, did that now, but now i should once in a while monitor when the provider changes ip addresses |
| 13:44.51 | leifmadsen | what you're expecting to happen isn't supported and thus would be a feature request |
| 13:45.07 | leifmadsen | get a provider that doesn't change IP addresses :) |
| 13:45.29 | *** part/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 13:45.36 | BarthezZ | I wish the customer took a better provider so i could just add 1 peer for the differen caller id's, not 5 (and more in the future) |
| 13:46.13 | BarthezZ | ah well, it works |
| 13:46.57 | leifmadsen | Gido-E: maybe it's my PDF reader, but this "asterisk voor programmeurs.pdf" I'm looking at has some serious styling issues. I'm looking at page 18 and it has missing newlines so dialplan examples look weird, and Verbose(2/Unrouted call handler) is not correct syntax |
| 13:47.29 | tuxx- | that sounds dutch. 'asterisk voor programmeurs' |
| 13:47.29 | tuxx- | :D |
| 13:47.33 | Gido-E | leifmadsen normally i open it with acroreaded. But i wil look into it right now |
| 13:47.41 | Gido-E | That is DUTCH |
| 13:47.48 | leifmadsen | tuxx-: it is :) |
| 13:47.48 | tuxx- | orly. |
| 13:47.54 | tuxx- | <- dutch 2 |
| 13:47.55 | tuxx- | xD |
| 13:48.09 | leifmadsen | I'm opening it with some PDF reader in Linux |
| 13:48.18 | Chainsaw | evince most likely, if you use Gnome. |
| 13:48.24 | leifmadsen | Document Viewer 2.28.1 |
| 13:48.44 | Chainsaw | *nod* That's evince. Which I find works on pretty much everything. Except credit card statements from HSBC. |
| 13:49.10 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 13:49.12 | *** join/#asterisk BANSAL (~bansal@117.207.82.30) |
| 13:49.59 | Gido-E | leifmadsen i see what you mean. Thnx for the tip. |
| 13:50.31 | Gido-E | i only mis good docu about the users.conf. Cant find it. Anny tips? |
| 13:51.20 | BarthezZ | just don't use it? Or well, I once read a big rant about users.conf disabling all the advanced features and just being too simple |
| 13:51.53 | Gido-E | BarthezZ, i just want the documentation. |
| 13:52.20 | BarthezZ | hehe |
| 13:55.35 | *** join/#asterisk ickmund (~magnus@cli-5b7ee15c.bcn.adamo.es) |
| 13:59.40 | TSM | has anyone used S4L, ive got it installed but its not logging in to any account |
| 13:59.42 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:59.45 | TSM | S4A |
| 13:59.49 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 13:59.58 | leifmadsen | TSM: the accounts are business panel accounts? |
| 14:00.08 | TSM | @leifmadsen: yes |
| 14:00.16 | leifmadsen | (and not the primary business panel account?) |
| 14:00.44 | TSM | @leifmadsen: well ive tried it with my skype username and another username, we are both managers on the panel |
| 14:01.06 | leifmadsen | you can't use the primary accounts there -- you have to create an account in the business panel for it |
| 14:01.46 | leifmadsen | S4A works for me on 1.6.2.7 |
| 14:02.07 | *** join/#asterisk grante (~PuPic@64.235.109.58) |
| 14:03.07 | TSM | @leifmadsen: nop that does not work, ive tried a non manager account and it will not login |
| 14:03.35 | leifmadsen | don't know what to tell you other than it's a business product so I think you get support for that if you call Digium |
| 14:07.24 | TSM | @leifmadsen: can i call them or only email? |
| 14:07.46 | leifmadsen | TSM: I think you can call, but I don't know what kind of support you get with S4A |
| 14:09.17 | *** join/#asterisk saftsack (~oliver@p579DD646.dip.t-dialin.net) |
| 14:09.42 | saftsack | hey, regarding your experience: waht is better? app_fax_digium or t38modem? |
| 14:15.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:15.50 | *** join/#asterisk x303 (~X303@28.28.102.97.cfl.res.rr.com) |
| 14:20.12 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:20.12 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:44.48 | pabelanger | Any phone recommendations: After I have re-invited a call from Asterisk, and RTP is between 2 SIP phones, I'm go 'on hold' with one phone. I want to stream MOH from Asterisk to the other. |
| 14:45.15 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:47.33 | *** join/#asterisk Rajmohan (~raj@122.164.76.94) |
| 14:48.53 | WIMPy | pabelanger: Tell the phone to use a moh server (never tried that) or disable reinvites. |
| 14:50.06 | Baylink-work | pabelanger: As handy as reinvite is, it reduces your call control possibilities -- and your security for that matter -- markedly. |
| 14:50.09 | pabelanger | WIMPy: Ya, that is what I usually do. But just had a conversation with somebody about Asterisk, and they may a reference to Asterisk being able to handle this. I have never tried it, working setting it up now. |
| 14:50.36 | Baylink-work | Nope. Once the phones have reinvited, * is out of the loop. Unless I badly misunderstood something. |
| 14:50.36 | pabelanger | Baylink-work: Agree, but this is a closed network. |
| 14:50.46 | Baylink-work | Ah. |
| 14:50.59 | [TK]D-Fender | pabelanger: AFAIK the reinvite only affects the audio stream. SIP should still go to the server which should force another reinvite back to provide MoH |
| 14:51.49 | pabelanger | [TK]D-Fender: yes, that is how it was explained to me. I was curious if anybody is doing that, and any success stories with SIP phones. |
| 14:52.21 | Katty | any good movies come out lately |
| 14:52.22 | WIMPy | That would have to be initiated by the phone, wouldn't it? |
| 14:53.52 | [TK]D-Fender | pabelanger: Do you have a failure? |
| 14:54.51 | pabelanger | [TK]D-Fender: No, I am testing it now. Like I said, was just looking for feedback. |
| 14:55.21 | [TK]D-Fender | pabelanger: FFS just place a call first :p |
| 14:57.42 | pabelanger | [TK]D-Fender: Working on it... |
| 14:59.20 | TSM | does S4A support g722, seems it chose the worse quality g729 over ulaw |
| 15:01.33 | [TK]D-Fender | I figured SFA would use SILK |
| 15:04.33 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:06.12 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 15:07.39 | ChannelZ | I'm not sure SFA understands silk |
| 15:08.17 | *** join/#asterisk grandpapadot (~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 15:08.27 | grandpapadot | Hey guys, in asterisk.conf, the load=0.9, is that 90% of CPU usage? |
| 15:10.07 | *** join/#asterisk jsmith (~jsmith@asterisk/training-and-documentation-guru/jsmith) |
| 15:10.07 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 15:10.25 | *** join/#asterisk Porks (~BARAD-DUR@unaffiliated/porks) |
| 15:10.53 | *** join/#asterisk BANSAL (~bansal@117.207.82.30) |
| 15:13.02 | *** join/#asterisk BANSAL (~bansal@117.207.82.30) |
| 15:13.30 | *** part/#asterisk Porks (~BARAD-DUR@unaffiliated/porks) |
| 15:13.34 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 15:13.54 | ZeXr0 | In the Asterisk CDR, is there a way to know if it's an Inbound or outbound call ? |
| 15:14.08 | Qwell | grandpapadot: no, it's a load of 0.9 |
| 15:17.40 | ZeXr0 | No one knows how to determine if a call is an Inbound call or an Outbound call from the information in the CDR ? |
| 15:17.41 | ChannelZ | ZeXr0: in a way all calls are 'inbound' to Asterisk from its point of view. You could figure it out by the calling number or called number, or maybe even context depending on your setup, etc |
| 15:17.43 | grandpapadot | Qwell, tnx. |
| 15:17.47 | *** join/#asterisk gaunt4000 (~d5b0a1d3@gateway/web/freenode/x-qgvmehskofsudpga) |
| 15:18.04 | ZeXr0 | ah okay |
| 15:18.06 | ZeXr0 | thanks ChannelZ |
| 15:18.11 | gaunt4000 | hi all |
| 15:18.44 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:18.54 | ChannelZ | in my case all incoming calls from the outside world go into the context 'incoming' and nothing else does. |
| 15:20.11 | gaunt4000 | Im working with asterisk 1.6.2 and im tring to work with "graphyc mode" and I want to know where this soft save extensions, so, I know that u can setup extensions in sip.conf and extensions.conf, but when u are working with graphic mode, where asterisk save this changes? someone can helps? |
| 15:20.54 | ChannelZ | ~freepbx |
| 15:20.55 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:21.04 | ChannelZ | ..I assume anyway.. |
| 15:21.19 | grandpapadot | Qwell - so on a quad core box, an asterisk load of 3.6 is basically 90% server load? |
| 15:21.34 | Qwell | no |
| 15:21.49 | Qwell | there is no such thing as an "asterisk load" |
| 15:22.05 | grandpapadot | Qwell: I mean system load |
| 15:22.05 | Qwell | read up on how load averages work |
| 15:22.17 | grandpapadot | Qwell: will do, tnnx |
| 15:23.22 | pabelanger | grandpapadot: http://blog.scoutapp.com/articles/2009/07/31/understanding-load-averages |
| 15:24.13 | grandpapadot | Thanks, pabelanger |
| 15:25.48 | Qwell | grandpapadot: disregard the "Multicore vs. multiprocessor" section there. It's wrong. |
| 15:26.31 | Qwell | (mostly because of Intel Hyperthreading) |
| 15:27.18 | grandpapadot | Right, I understand those aspects, that's a great article, it was what I assumed, so on a quad core box, I would likely set the load to 3.0, which will give me 25% headroom ... |
| 15:27.57 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:29.36 | *** join/#asterisk JayT (unstable@tor/regular/sid) |
| 15:29.51 | JayT | What is the key combo for broadvoice to allow people to see me on caller ID? |
| 15:30.19 | *** join/#asterisk ITgeek (~chadd@99-23-174-75.lightspeed.irvnca.sbcglobal.net) |
| 15:31.07 | grandpapadot | Qwell: if asterisk reaches it's load or concurrent call value as set in asterisk.conf, will it retain the ability to still place calls? |
| 15:31.38 | grandpapadot | Qwell: in the case of a call queue server, will it still ring the members even know it's not allowing inbound calls? |
| 15:32.26 | Baylink-work | grandpapadot: I believe you're assuming perfect cross-core distribution; that's not generally safe. |
| 15:32.42 | [TK]D-Fender | JayT: http://broadvoice.com/features_access_codes.html |
| 15:33.30 | Qwell | grandpapadot: Asterisk can still work with a load of 87. |
| 15:33.31 | grandpapadot | Baylink-work: right, I got that, but I'm seeing good distributed load now on this server, so I figured with 25% overhead i would be in good shape? |
| 15:33.55 | Qwell | CPU time would just be queued. You may or may not actually notice it. |
| 15:34.03 | grandpapadot | Qwell: Ah! |
| 15:34.12 | Baylink-work | Assuming that 25% isn't the amount by which your present load distribution cannot expand due to such failings... |
| 15:34.23 | Baylink-work | If you can actually *load* it to 4.0, then yes. |
| 15:34.24 | grandpapadot | Qwell: So not it's down to me understanding my kernel vs cpu's capabilities .. arg |
| 15:34.33 | Baylink-work | With a real world load, I mean, grandpapadot |
| 15:34.58 | Qwell | for instance, if you had 1000 processes all accessing a single file on a single disk... the entire rest of the system would work just fine. Those threads would be "waiting", but it wouldn't hurt anything else. |
| 15:35.00 | grandpapadot | Well, we're setting up to take the calls for the garth brooks final concert, and are expecting 10,000-20,000 calls per minute .. |
| 15:35.07 | Qwell | load *IS NOT* the same as "CPU usage" |
| 15:35.39 | grandpapadot | Qwell: got it, I'm with it, I was looking for a reasonable setting for the load= value to keep the servers from coming to their needs when they get blasted |
| 15:35.47 | grandpapadot | needs = knees |
| 15:36.06 | keith4 | is it worth trying out a Yealink sip phone? i've never heard of 'em |
| 15:36.21 | paulc | grandpapadot: Heavy call volume! Multiple boxes? SER fronted? (I'm interested because we have 26 boxes running 960 ports each, non-Asterisk, right now..) |
| 15:36.27 | Qwell | keith4: there's probably a reason you've never heard of them, eh? |
| 15:36.28 | keith4 | they look like linksys ripoffs |
| 15:36.29 | leifmadsen | keith4: is it a full phone or a USB device? |
| 15:36.34 | leifmadsen | ah |
| 15:36.43 | keith4 | http://www.voipsupply.com/yealink-sip-t20pex |
| 15:36.48 | leifmadsen | I've used a yealink USB adapter for a softphone. Worked good enough. |
| 15:36.54 | keith4 | Qwell: well, maybe they're just *brand new* ;-) |
| 15:37.07 | Qwell | they arent |
| 15:37.11 | grandpapadot | Yep, 100 servers, no SER in this deployment |
| 15:38.27 | paulc | grandpapadot: so how do you spread your calls across boxes? round robin DNS? or something cleverer? |
| 15:38.28 | JayT | thanks [TK]D-Fender I appreciate the help |
| 15:38.47 | keith4 | Qwell: good to know. thanks. |
| 15:39.09 | paulc | grandpapadot: and out of interest, how many calls per box before it maxes out - do you enforce a limit? or is that the whole point of the current conversation thread (I just got in, sorry) |
| 15:39.50 | [TK]D-Fender | JayT: Next time you might want to like.. you know... look at their own FAQ's yourself :p |
| 15:39.57 | grandpapadot | paulc: aparrently, there is a high end cisco router over at XO that is going to handle the distribution for us. |
| 15:40.00 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:41.08 | paulc | grandpapadot: Ah gotcha.. bit easier that way, for sure :) |
| 15:41.16 | grandpapadot | paulc: I'm hoping each box will handle about 500 calls, our normal production boxes run loaded about 250 and they are doing some transcoding. On the servers for the garth brooks event, no transcoding, g729 everywhere, I just go finished converting all of our custom audio go g729 |
| 15:41.24 | grandpapadot | go = to |
| 15:41.41 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 15:42.26 | paulc | grandpapadot: that's cool. I'm doing some playing here - I want to grab a couple of boxes and hit an asterisk box with a few hundred calls, see how it handles it.. I know on our current blades we get 800-900 calls to our legacy app with no problem |
| 15:45.13 | JayT | [TK]D-Fender: I was checking their website, I didn't find it. |
| 15:45.58 | *** part/#asterisk jsmith (~jsmith@asterisk/training-and-documentation-guru/jsmith) |
| 15:47.31 | [TK]D-Fender | JayT: Didn't take me 2 minutes to find it.... |
| 15:49.10 | *** join/#asterisk yabadabado (~root@static-213-115-44-229.sme.bredbandsbolaget.se) |
| 15:49.35 | yabadabado | ~docs |
| 15:49.36 | infobot | docs is, like, for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 15:50.12 | yabadabado | can somebody point me to all the recommended asterisk docs and reading list... |
| 15:51.26 | yabadabado | ~topic |
| 15:52.23 | yabadabado | the channel topic - whats wrong with it |
| 15:52.30 | leifmadsen | what do you mean? |
| 15:52.33 | leifmadsen | nothing is wrong with it |
| 15:53.34 | yabadabado | answer me leif |
| 15:53.34 | yabadabado | point me to official n unofficial asterisk docs |
| 15:53.38 | leifmadsen | I don't answer to demands |
| 15:53.52 | yabadabado | specially when it comes to integration with tdm exchanges, |
| 15:54.07 | yabadabado | i/clecs, nsps(isdn..) |
| 15:54.33 | *** join/#asterisk kartik (~koolkarti@117.207.82.244) |
| 15:54.47 | yabadabado | leifmadsen: um on the front line fighting to introduce asterisk in a telco - you gotta help |
| 15:54.49 | [TK]D-Fender | yabadabado: What "documentation"? :) |
| 15:55.04 | *** join/#asterisk willianmazzardo (~willian@server2.sysvoip.com.br) |
| 15:55.08 | willianmazzardo | tzafrir_laptop: hi |
| 15:55.10 | leifmadsen | technically, I don't have to do ANYTHING |
| 15:55.24 | tzafrir_laptop | willianmazzardo, what's up? |
| 15:55.24 | [TK]D-Fender | yabadabado: What "integration"? Could you be a little more vague perhaps? |
| 15:55.37 | willianmazzardo | i`m trying to install OSLEC with dahdi |
| 15:56.02 | willianmazzardo | but I had some errors when I use OSLEC in /etc/dahdi/system.conf |
| 15:56.05 | yabadabado | [TK]D-Fender: ISDN with QSIG against * |
| 15:56.29 | willianmazzardo | tzafrir_laptop: some logs: http://pastebin.org/283502 |
| 15:56.30 | [TK]D-Fender | yabadabado: There is no quality documentation AFAIK as to how much of the spec * follows |
| 15:56.55 | willianmazzardo | do you know how to fix this? |
| 15:57.02 | yabadabado | [TK]D-Fender: i need all the bookmarks for that |
| 15:57.24 | leifmadsen | lunch! |
| 15:57.24 | ccesario | tzafrir_laptop, the oslec_kernelorg patch can solve this....!?? (I remeber that you send me a fix to this) |
| 15:57.31 | yabadabado | i am in a moode to read read read and then like a lawyer fight fight fight until i can prove to them that * has it what you need |
| 15:57.35 | tzafrir_laptop | willianmazzardo, looks like the modules echo / oslec was not loaded. |
| 15:57.46 | tzafrir_laptop | How did you build it? |
| 15:57.49 | yabadabado | i heard dialogic and sangoma can be used for ss7 integration as well. (no i am not vague) |
| 15:58.05 | yabadabado | they are pretty solid i heard |
| 15:58.06 | [TK]D-Fender | yabadabado: http://www.google.ca/#hl=en&source=hp&q=asterisk+qsig+isdn&aq=f&aqi=&aql=&oq=&gs_rfai=&fp=c9e66e819996722 <--- bookmark away |
| 15:58.08 | willianmazzardo | tzafrir_laptop: I followed this tutorial: http://blog.paulsnet.org/?p=44 |
| 15:58.40 | willianmazzardo | the dahdi_echocan_oslec.ko is in /lib/modules/kernelversion/dahdi |
| 15:58.53 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:00.15 | yabadabado | [TK]D-Fender: Thanks for the google. |
| 16:00.57 | yabadabado | I wish pstn support in asterisk was tabulated in terms of supported protocols/interfaces and any propertietry hardware to be used |
| 16:01.47 | yabadabado | better hack a pc of some digiums' mark n sales dude |
| 16:01.57 | [TK]D-Fender | yabadabado: Nobody bothered to make you a pretty table. Go make one yourself |
| 16:02.04 | yabadabado | [TK]D-Fender: screw u |
| 16:02.19 | yabadabado | 7 years of chat chat and u dont even have one. |
| 16:02.19 | [TK]D-Fender | yabadabado: You keep asking for things that don't exist. What do you want? |
| 16:02.30 | yabadabado | i ll fig |
| 16:03.03 | yabadabado | thanks for your help everyone |
| 16:05.24 | grandpapadot | Hey guys, any real benefit running -p (psuedo RT) if asteris is the only app (beyond the base install) on a box? |
| 16:05.30 | grandpapadot | asteris = asterisk |
| 16:05.45 | *** join/#asterisk n3hxs (~HAMming@173-162-253-153-NewEngland.hfc.comcastbusiness.net) |
| 16:06.22 | grandpapadot | ah, nevermind .. |
| 16:06.25 | grandpapadot | ;p |
| 16:06.55 | willianmazzardo | :) |
| 16:08.01 | p3nguin | i_hax_u: /away |
| 16:08.06 | p3nguin | fuck |
| 16:08.13 | p3nguin | disregard |
| 16:08.15 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:08.33 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:09.53 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 16:10.00 | willianmazzardo | tzafrir_laptop: something? |
| 16:11.21 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
| 16:13.26 | *** join/#asterisk scalex000 (~chatzilla@190.166.189.12) |
| 16:13.31 | scalex000 | Hello, |
| 16:14.36 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 16:14.48 | scalex000 | hello, this warning its important http://pastebin.com/qCJ0V9fW, have solution? |
| 16:17.27 | Gido-E | scalex000 what is the problem? |
| 16:17.38 | Gido-E | it complains about the codecs |
| 16:18.22 | Gido-E | scalex000 if you want to trancode to g729, you need a license, you know that? |
| 16:18.29 | scalex000 | i get this warning first, [2010-05-26 12:16:03] WARNING[18092]: translate.c:268 ast_translator_build_path: No translator path from alaw to unknown |
| 16:18.35 | scalex000 | yes, I knew it |
| 16:19.16 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 16:19.20 | scalex000 | gido, what kind of license? |
| 16:19.33 | Gido-E | scalex000 check digium site. |
| 16:19.42 | scalex000 | I have one license |
| 16:19.59 | scalex000 | gido, I only use g729 for skype for asterisk |
| 16:20.07 | scalex000 | gido, let me check |
| 16:20.24 | Gido-E | on cli, start with g729 <tab> |
| 16:21.58 | *** join/#asterisk Mw3 (mw3@2a01:270:90dc::42) |
| 16:23.50 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.178.236.dsl.dyn.forthnet.gr) |
| 16:26.34 | lantizia | How can I get up the DAHDI version information? |
| 16:28.41 | Chainsaw | lantizia: In Asterisk itself? dahdi show version |
| 16:29.05 | lantizia | Chainsaw: it's ok I looked at /sys/kernel/modules/dahdi/version |
| 16:30.04 | Chainsaw | lantizia: *nod* That works too :) |
| 16:30.39 | lantizia | Chainsaw: I read that qozap was removed from dahdi as wcb4xxp does the trick... is there a list of dahdi supported cards? |
| 16:31.03 | Qwell | qozap was never in dahdi |
| 16:31.08 | Qwell | (or zaptel) |
| 16:31.19 | lantizia | OK but is wcb4xxp meant to work with Junghanns.NET cards? |
| 16:32.00 | *** part/#asterisk gevaerts (~fg@rockbox/developer/gevaerts) |
| 16:32.52 | lantizia | Chainsaw, Qwell ? |
| 16:35.07 | *** join/#asterisk knarfly (~vlad@c-98-242-237-166.hsd1.fl.comcast.net) |
| 16:35.19 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 16:35.35 | knarfly | anyone know about Yealink SIP-T20PEX VOIP phones.....? |
| 16:35.51 | lantizia | bbiab |
| 16:37.07 | knarfly | lantizia: is that buyer beware or something... |
| 16:37.47 | *** part/#asterisk willianmazzardo (~willian@server2.sysvoip.com.br) |
| 16:40.08 | [TK]D-Fender | knarfly: Yes |
| 16:41.17 | knarfly | okay cool I got burned with some Grandstream GXP 2000's the phone looked and worked great but the within days of each other they stopped working |
| 16:41.53 | *** part/#asterisk grandpapadot (~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 16:41.55 | knarfly | funny though my GX BT200 is still working fine and it's a cheaper phone than the GXP2000 |
| 16:42.42 | [TK]D-Fender | knarfly: "working" and being a "cheap piece of crap" are not mutually exclusive. |
| 16:42.48 | [TK]D-Fender | knarfly: its a question of standards |
| 16:43.43 | *** join/#asterisk Lantizia (~Lantizia@212.183.140.49) |
| 16:43.48 | Lantizia | ok im back |
| 16:44.18 | Lantizia | So... can the Junghanns.NET BRI cards work with DAHDI? |
| 16:44.42 | Lantizia | can anyone give me a head start in seeing a supported list of cards as i cant find one |
| 16:45.56 | Kobaz | grandstream phones are just horrible |
| 16:46.06 | Kobaz | bad audio quality, and the power supplies tend to fry after a year |
| 16:46.32 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.7 (2010/05/04), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.31 (2010/05/04), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.11 (2010/05/26) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 16:46.36 | *** join/#asterisk maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:46.48 | leifmadsen | libpri 1.4.11 has just been released. For more information please see the release announcement: http://www.asterisk.org/node/51360 |
| 16:47.11 | scalex000 | TK, have you heard about this case, I have connection between Nortel and BCM using SIP, when I dial a number that belong a mobile so I need to dial with +1 for go though i not the voice from Nortel i only listen a ring tone, why? |
| 16:47.53 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:48.09 | *** part/#asterisk maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
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| 17:15.03 | leifmadsen | shhh |
| 17:17.02 | knarfly | ok guys here it comes...Comcast made their announcement this morning. all customers will be limited to 250GB per month |
| 17:17.02 | knarfly | this is their first step into forcing us to use their VOIP as opposed to our own |
| 17:17.31 | knarfly | watch the other ISP's to follow suite shortly |
| 17:19.53 | Qwell | knarfly: you realize that a voip call running 24/7 for an entire month, is like nothing, right? |
| 17:20.31 | Qwell | by my crappy math, it's all of 800mbytes |
| 17:21.44 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 17:21.54 | russellb | i think your math fails |
| 17:22.13 | Qwell | (64k * 60 * 60 * 30) / 8 |
| 17:22.15 | russellb | but you didn't specify a codec :-) |
| 17:22.23 | russellb | you missed a * 24 |
| 17:23.07 | russellb | and it's 64k of the audio itself, you didn't account for the overhead |
| 17:23.09 | leifmadsen | IP overhead |
| 17:23.30 | leifmadsen | s/64k/96k/ |
| 17:23.38 | leifmadsen | that'll get you closer :) |
| 17:23.58 | Qwell | 21gb? there's no way |
| 17:24.10 | leifmadsen | I get like 1.296 GB |
| 17:24.20 | russellb | lol ... zomg math. |
| 17:24.22 | Qwell | you didn't *24 |
| 17:24.33 | leifmadsen | because I s/64k/96k/ |
| 17:25.44 | *** join/#asterisk saftsack (~oliver@p5DDCF672.dip.t-dialin.net) |
| 17:25.47 | knarfly | Qwell: with my wife on the phone with her sisters all day and night I can only imagine what my useage is |
| 17:25.50 | saftsack | hey, some patton smartnode users here? |
| 17:25.56 | leifmadsen | knarfly: it's not nearly as much as you think |
| 17:26.14 | leifmadsen | knarfly: if you used more than 500MB a month in usage for VoIP I'd be shocked |
| 17:26.26 | knarfly | yes but add in a couple of movies each day for the kids and my downloads from RapidSahre |
| 17:26.42 | Qwell | so your real complaint is that Comcast is limiting your porn downloads |
| 17:26.46 | leifmadsen | knarfly: so your theory about limiting you to 250GB because of VoIP is almost laughable. It's likely a cap due to over usage of data such as downloading movies |
| 17:27.11 | leifmadsen | knarfly: welcome to the world of tomorrow -- unlimited data is not sustainable. |
| 17:27.39 | leifmadsen | I limited to about 100GB or something like that with Rogers. I never hit my cap. |
| 17:27.45 | leifmadsen | and I work at home |
| 17:27.48 | knarfly | leifmadsen: you are probably right but either way it's Comcast starting to place limits on the amount you use the Internet before chargib you more |
| 17:27.59 | leifmadsen | s/Internet/internet/ |
| 17:28.05 | Qwell | knarfly: and these limits have been in place for well over a year. |
| 17:28.11 | Qwell | in place *and known* |
| 17:28.12 | leifmadsen | like I said, welcome to the world of tomorrow. Every service will be capped soon enough. |
| 17:28.44 | knarfly | leifmadsen: except for that oilwell service down in the Gulf of Mexico |
| 17:28.50 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 17:28.57 | leifmadsen | ok this conversation just left the track |
| 17:29.04 | leifmadsen | goes back to something productive |
| 17:29.06 | knarfly | but Obama's on his way there...he'll save us |
| 17:29.35 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 17:29.40 | *** kick/#asterisk [knarfly!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (your off-topicness has passed the maximum threshold. Please return when you can speak closer on-topic.) |
| 17:29.50 | *** join/#asterisk rallveird (~vegard@212.33.136.212) |
| 17:30.26 | [TK]D-Fender | And now for something completely different! </python> |
| 17:34.06 | TSM | would be nice if S4A had a full AGI implimentation |
| 17:34.55 | TSM | or can i just use the exec command to execute one line dialplan commands |
| 17:35.07 | *** join/#asterisk QaDeS_ (~mklaus@p54A1B0BF.dip0.t-ipconnect.de) |
| 17:35.16 | ZeXr0 | Hum... I have this error when I try to receive a call... When I make a call it's ok : chan_sip.c:8825 check_auth: username mismatch, have <107001>, digest has <s> |
| 17:35.33 | [TK]D-Fender | TSM: What does SFA have to do with AGI? |
| 17:36.58 | ZeXr0 | Anyone have some hint where I could start my research on this problem ? |
| 17:37.13 | [TK]D-Fender | TSM: I also see no CLI command for "exec" |
| 17:37.14 | leifmadsen | TSM: AGI is independent of the channel driver |
| 17:37.27 | TSM | [TK]D-Fender: there are loads of commands to get skype metadata but they are all dialplan commands, you cant access them via the CLI etc |
| 17:38.21 | [TK]D-Fender | TSM: So go make an exten to pop all of those answers into ASTDB, originate into it, then go pick up the results |
| 17:38.53 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:39.01 | TSM | [TK]D-Fender: crudy way around for metadata than can change all the time for the buddies |
| 17:39.21 | TSM | [TK]D-Fender: seems its the only way, just thinking it would have been nicer to do that via the CLI |
| 17:39.50 | [TK]D-Fender | TSM: SFA adds no CLI commands for this info, and no Manager events? |
| 17:39.57 | *** join/#asterisk kartik (~koolkarti@117.207.82.244) |
| 17:40.48 | TSM | [TK]D-Fender: seems its limited, i can see the buddies and their status, but i cant see their full display name, country etc.., im trying to make a phone directory of buddies that are registered to the account thats why i want all that data |
| 17:41.35 | [TK]D-Fender | TSM: Not sure you'd need this to be extremely real-time... |
| 17:42.08 | TSM | [TK]D-Fender: not realy no, but it would be easier |
| 17:42.26 | [TK]D-Fender | TSM: Agreed... see if Digium has a request form for this stuff... |
| 17:42.38 | [TK]D-Fender | TSM: as you are a client it might get more consideration |
| 17:46.33 | Naikrovek | launches Portal |
| 17:50.36 | saftsack | hi, has someone an image.bin for a smartnode 4552? |
| 17:51.22 | *** join/#asterisk Coolrazor (~Tim@71.95.209.242) |
| 17:51.31 | leifmadsen | saftsack: seems like something you would get from the manufacturer |
| 17:52.15 | saftsack | yes, still asked there. but i don't know if i will get an answer from there before the telephone system has to run again here. |
| 17:53.01 | saftsack | leifmadsen, they offer just relative updates on their website, but no incremental bin for the bootloader. |
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| 18:14.04 | Katty | sprawls |
| 18:14.12 | Katty | dear lord |
| 18:14.16 | Katty | why did i eat mexican food |
| 18:15.37 | *** join/#asterisk x303 (~X303@28.28.102.97.cfl.res.rr.com) |
| 18:16.02 | Qwell | Katty: because you enjoy flavor |
| 18:17.56 | ZeXr0 | And you really don't like the other end |
| 18:18.20 | *** join/#asterisk kartikm (~koolkarti@117.207.82.244) |
| 18:18.47 | Katty | Qwell: yesh |
| 18:18.51 | Katty | ZeXr0: that statement did not parse. |
| 18:18.55 | Katty | ZeXr0: i don't think i want to know tho |
| 18:20.02 | *** join/#asterisk kartik (~koolkarti@117.207.82.244) |
| 18:20.15 | ZeXr0 | Katty: You don't want to ;) |
| 18:20.58 | Katty | k |
| 18:21.07 | *** join/#asterisk grungies1138 (~grungies1@cpe-66-74-197-34.san.res.rr.com) |
| 18:21.12 | grungies1138 | Hi kids! |
| 18:21.33 | Katty | hello kiddo |
| 18:21.45 | grungies1138 | I have a registered SIP Trunk and SIP Softphone. I can dial out, but I get disconnected after 30 seconds. |
| 18:21.55 | grungies1138 | running 1.6 |
| 18:21.57 | grungies1138 | no GUI |
| 18:22.28 | Katty | what does the cli say |
| 18:22.44 | grungies1138 | no errors. |
| 18:22.54 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 18:23.19 | Katty | erm |
| 18:23.21 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 18:23.30 | Katty | okay let me explain how this works |
| 18:23.31 | Katty | to help you |
| 18:23.36 | Katty | we have to /see/ what's going on |
| 18:23.52 | Katty | so unless you pastebin all the relavent info... |
| 18:24.03 | Katty | we might as well be staring at a brick wall |
| 18:24.47 | thehar | builds a brick wall to stare at |
| 18:24.51 | thehar | like this one? |
| 18:25.05 | Katty | yes dear |
| 18:25.06 | Katty | just like that one |
| 18:25.11 | thehar | oh i see |
| 18:25.12 | thehar | hehe |
| 18:25.39 | Katty | mkay mister smart tail |
| 18:25.45 | thehar | ;) |
| 18:25.52 | beek | hugs Katty. |
| 18:25.55 | thehar | been a while since i've poked my head around in here |
| 18:26.12 | grungies1138 | I understand that, but not sure where to look forst |
| 18:26.26 | Katty | forst? |
| 18:26.37 | beek | Between first and farst. |
| 18:26.41 | grungies1138 | first. on the phone too. |
| 18:26.55 | beek | Katty -- is the critter cam on? |
| 18:27.10 | Katty | just pastebin all relavent info, including the cli |
| 18:27.24 | Katty | beek: mmm, no i dont think i turned it on this morning |
| 18:27.33 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:30.09 | beek | will check the critter cam later... |
| 18:33.32 | *** join/#asterisk kartikm (~koolkarti@117.207.82.244) |
| 18:34.14 | Defraz | Could someone look at this and tell me if I am getting this right. It is supposed to hack the +1 off if it is there and the just 1 off if it is there. |
| 18:34.16 | Defraz | http://pastebin.com/S4DxCY2U |
| 18:34.20 | Katty | beek: they haven't eaten much in the last couple months |
| 18:34.41 | Katty | beek: gotta refill the feeders again this weekend, tho |
| 18:34.52 | Katty | beek: so about 75lbs of food in... 3 months, rather than 1 month |
| 18:35.17 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 18:38.13 | [TK]D-Fender | Defraz: lol... so not happening :) |
| 18:38.22 | beek | Katty -- the presumption is that they can fend for themselves bettery this time of year. |
| 18:39.24 | Defraz | really? |
| 18:39.37 | [TK]D-Fender | Defraz: See in #freepbx |
| 18:40.05 | Katty | beek: well. |
| 18:40.14 | Katty | beek: that and their not trying to maintain body heat all hours of the day and night |
| 18:41.32 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 18:46.23 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 18:48.49 | grungies1138 | sorry guys. What 'relevant information' would you like? sip.conf? Extensions.conf? |
| 18:48.54 | pabelanger | Odd question: could /dev/dahdi/channel be created under a different path for Asterisk, I would assume no |
| 18:50.49 | p3nguin | defraz: ExecIf($["${CALLERID(num):0:2}" = "+1"]|Set|CALLERID(num)=${CALLERID(num):2}) |
| 18:52.30 | grungies1138 | <PROTECTED> |
| 18:52.46 | p3nguin | Asterisk does not play well with comfort noise. |
| 18:52.57 | Defraz | p3nguin: thanks. |
| 18:53.14 | grungies1138 | I couldn't find it in X-Lite |
| 18:53.26 | grungies1138 | would that cause it to disconnect? |
| 18:53.27 | p3nguin | defraz: Use this instead of the GotoIf and Set commands. |
| 18:57.41 | [TK]D-Fender | p3nguin: Already taken this into #freepbx with him... |
| 18:57.48 | p3nguin | ah |
| 18:59.25 | grungies1138 | anyone know how to disable Comfort noise in X-Lite |
| 18:59.29 | grungies1138 | ? |
| 18:59.57 | p3nguin | Do you see any setting about voice detection or silence suppression? |
| 19:00.30 | p3nguin | If you do, disable it, since that's where the comfort noise is coming from. |
| 19:01.21 | grungies1138 | only thing I found is Noise reduction? disabled it and still getting the error. |
| 19:03.13 | *** join/#asterisk sat-man (~jlupresto@c-174-52-20-94.hsd1.ut.comcast.net) |
| 19:03.43 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:03.44 | grungies1138 | I may have found it |
| 19:04.37 | grungies1138 | yeah I think I got it. |
| 19:04.46 | grungies1138 | It was a client issue. sorry to bother you guys |
| 19:04.47 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 19:05.38 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 19:07.32 | grungies1138 | <PROTECTED> |
| 19:08.44 | [TK]D-Fender | grungies1138: Show us your peer. |
| 19:08.51 | [TK]D-Fender | ~pb |
| 19:08.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 19:08.53 | [TK]D-Fender | ^^^ |
| 19:09.11 | grungies1138 | sip.conf and voicemail.conf? |
| 19:09.18 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 19:09.31 | [TK]D-Fender | grungies1138: Show us your peer. <----------------------- |
| 19:09.57 | grungies1138 | still trying to get the terminology right. sip.conf part right? |
| 19:10.53 | [TK]D-Fender | YES |
| 19:11.41 | *** join/#asterisk scalex000 (~chatzilla@190.167.156.68) |
| 19:12.08 | scalex000 | hi I do not understand this notice :[2010-05-26 15:09:31] NOTICE[18170]: chan_sip.c:11156 check_auth: Correct auth, but based on stale nonce received from '""<sip:3247@200.88.120.181>' |
| 19:12.24 | *** join/#asterisk QaDeS (~mklaus@p54A1A8B3.dip0.t-ipconnect.de) |
| 19:13.34 | grungies1138 | http://pastebin.com/M0WfrRdz |
| 19:15.53 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:16.11 | [TK]D-Fender | grungies1138: indeed you have NOT specified a MAILBOX for that peer, just like it said |
| 19:20.15 | grungies1138 | got it. thanks Fender |
| 19:22.56 | *** join/#asterisk kartik (~koolkarti@117.207.82.244) |
| 19:25.50 | *** join/#asterisk kartik (~koolkarti@117.207.82.244) |
| 19:29.33 | *** join/#asterisk Zairus (~charrit69@66.76.60.213.dynamic.mundo-r.com) |
| 19:30.42 | Zairus | hi, I want to retrieve the ${EXTEN} var after a Goto(custom-context,s,1), is it possible? |
| 19:31.30 | Zairus | if I use the ${EXTEN} variable it is stored as 's' instead the real extension, eg: 1550 |
| 19:31.53 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 19:32.10 | ZeXr0 | I think I read somewhere that you need to set the EXTEN in a variable first, and then retrieve that one. |
| 19:32.50 | [TK]D-Fender | Zairus: That variable represents the exten you are ON. |
| 19:33.18 | [TK]D-Fender | ZeXr0: If you want to refer to something older, then you'll ahve to assign its value to something else and check that |
| 19:33.55 | ZeXr0 | [TK]D-Fender: I explain really bad, but that's what I meant :P |
| 19:34.09 | Zairus | ZeXr0 I thought on so store in the astdb but I have no way to pass the var name to the other context |
| 19:34.34 | ZeXr0 | hum You can set it in a variable that is not in AstDB |
| 19:34.37 | p3nguin | Set(oldexten=${EXTEN}) before the Goto, then check the value of ${oldexten} after. |
| 19:35.22 | Zairus | [TK]D-Fender ok. It's true, but Is there any variable that follow the goto? in the macros there is ${MACRO_EXTEN}, perhaps another one? |
| 19:35.42 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 19:35.58 | [TK]D-Fender | Zairus: No. |
| 19:36.03 | p3nguin | anything you set that doesn't get reset will work fine. |
| 19:37.12 | *** part/#asterisk sat-man (~jlupresto@c-174-52-20-94.hsd1.ut.comcast.net) |
| 19:37.29 | Zairus | and if I set a global variable it should be independent for other extensions or there is a choice of one extension pick the oldexten var? |
| 19:37.45 | Zairus | I can explain it better |
| 19:37.56 | [TK]D-Fender | Zairus: No reason for this to be global |
| 19:38.07 | [TK]D-Fender | Zairus: Just set a regular variable and check it. |
| 19:38.16 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-177-17.mia.bellsouth.net) |
| 19:38.59 | Zairus | if I use this variable ( Set(oldexten=${EXTEN}) ), it will be set as global or local variable to use for each call? |
| 19:39.38 | [TK]D-Fender | Zairus: Variables are by default local to the CALL you are in |
| 19:39.43 | *** join/#asterisk Lyma (~lyma@unaffiliated/lyma) |
| 19:40.16 | Zairus | ok, a lot of thanks |
| 19:43.10 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 19:43.59 | *** join/#asterisk Greek-Boy (~Greek-B0y@41.188.154.137) |
| 19:44.46 | Zairus | another question, Is there a way to prioritize queues? I want to have 2 independen queues, some agents logued in the 2 queues simoultaneusly... |
| 19:45.15 | Zairus | the queue 1 has a list of callerid that should be attended first than the queue 2 |
| 19:46.40 | Zairus | if all the agents are busy and in the queue 3 it has 3 calls waiting and it enter a new call in the queue 1... then the call in queue 1 should be attended first and the 3 calls waiting in the queue 2 should stay waiting... |
| 19:46.45 | Zairus | any idea? |
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| 20:15.53 | Micc | I have a customer using a Cisco-SIPGateway/IOS-12.x, and they keep trying to register with the correct auth information but the to/from contains their phone number and not their sip peer username. Is there something I should do to assign all the numbers to their account to auth correctly or should they be sending the username in the to/from header when registering? |
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| 20:19.58 | p3nguin | micc: Stop using "friend" and don't worry about the username. type=peer does not care about the username. |
| 20:23.32 | Micc | p3nguin, it is set to peer. I did try friend too, but its set to peer and still getting Registration from '<sip:425250xxxx@pbx1.nwdigits.com>' failed for '63.228.xxx.x' - No matching peer found |
| 20:23.58 | p3nguin | What is the peer name it is trying to use? |
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| 20:24.24 | Micc | badventures is the peer name in sip.conf |
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| 20:27.05 | lhz | asterisk send 503 to the phone, is it possible to send something else? (404 or 486) |
| 20:27.49 | p3nguin | micc: [badventures] ? |
| 20:28.02 | Micc | yes |
| 20:28.15 | p3nguin | micc: Does the phone ever try to send that name? |
| 20:29.02 | Micc | looks like calls are working now, but registration was failing. |
| 20:29.11 | Micc | If I ignore registration it probably works. |
| 20:29.20 | p3nguin | You don't need to register to be able to make calls from the device. |
| 20:29.52 | [TK]D-Fender | checkout time, BBIAB |
| 20:30.23 | p3nguin | A sip debug usually shows what peer name the phone is sending. I would want it to match the name of the peer. |
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| 20:39.06 | Micc | the only place it shows up in the register packet is here: Authorization: Digest username="badventures",realm="nwdigits.com",uri="sip:pbx1.nwdigits.com:5060",response="1434e6fdb90471fc96baaca2d9f5f0d5",nonce="3a4517c4",algorithm=MD5 |
| 20:39.34 | Micc | The only place I set the realm is in general section of sip.conf |
| 20:39.45 | Micc | I wonder if I need to associate that peer with that realm. |
| 20:40.40 | saftsack | someone who uses t.38 with a patton smartnode and *? |
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| 20:54.57 | p3nguin | micc: When you make a call to that phone, does your extension use something like Dial(SIP/badventures)? |
| 20:55.40 | Micc | no, Dial(SIP/4252505540@badventures) |
| 20:56.01 | p3nguin | Oh, badventures is not a phone? |
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| 21:00.31 | Micc | can I put host=dynamic and host=IP in sip.conf? |
| 21:00.40 | Micc | or host=dynamic,x.x.x.x ? |
| 21:00.49 | p3nguin | not both in the same peer. |
| 21:00.58 | Micc | I guess that would defeat the whole idea there. |
| 21:01.05 | p3nguin | If your device is sending registrations, you must have host=dynamic. |
| 21:01.21 | Micc | it is sending registrations. |
| 21:01.39 | p3nguin | If you have host=address-or-hostname, turn off registrations. |
| 21:03.01 | Micc | it is attempting to register as badventures and as phone numbers. I'll have them turn off number registrations. |
| 21:04.19 | p3nguin | You've not answered if badventures is a single phone/device. |
| 21:04.45 | Micc | Its a SIP gateway. Cisco trunk device. |
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| 21:05.27 | Micc | user agent is Cisco-SIPGateway/IOS-12.x |
| 21:05.45 | p3nguin | Why are we talking about phone numbers, then? |
| 21:07.07 | p3nguin | The device needs a peer name. The peer registers if needed, then you can route calls to the peer from Asterisk. If calls come from the peer, they will (hopefully) be routed properly in Asterisk. |
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| 21:17.34 | sizzers | Leif, please help. I see you dealt with this in the forums a bunch... but there's no clear resoution: "doing dnsmgr_lookup" |
| 21:17.37 | sizzers | It's SO annoying |
| 21:17.45 | sizzers | and I can't upgrade to the beta |
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| 21:18.29 | Gary_B | hi, anyone know kinds of companies that have high landline to mobile spend |
| 21:19.52 | sizzers | has anyone managed to solve this issue? "doing dnsmgr_lookup" |
| 21:20.25 | Micc | p3nguin, thats what I assumed too, but this cisco guy is setting it up and I keep seeing him trying to register phone numbers and extensions. |
| 21:20.57 | p3nguin | micc: Do the phones register to that Cisco device or to Asterisk? |
| 21:23.39 | p3nguin | s/Do the phones/Are the phone supposed to/ |
| 21:30.00 | Micc | as far as I know the phones should be registering to the cisco sip gateway. |
| 21:30.35 | p3nguin | Okay, then you shouldn't care (on Asterisk) what the phones are doing. |
| 21:31.11 | p3nguin | Asterisk sends a call to that other device as a peer and that device makes the decision what to do with the call. |
| 21:31.50 | Micc | its a cisco uc540 |
| 21:32.03 | p3nguin | You're on the same network, right? |
| 21:32.29 | Micc | no, the uc540 is behind nat. but I don't see a problem getting packets through. our asterisk server is not behind nat. |
| 21:32.57 | p3nguin | Does the IP address that you use to reach that device ever change? |
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| 22:38.17 | hluesea | hello channel, i have some troble on queue... Exactly when an agent on a call coming call trying to agent again. And this increase the missed calls for the agents. queue.conf is here ==> http://pastebin.com/a41ZuZji is anyone help me for the solving this problem ? |
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| 22:49.17 | [TK]D-Fender | hluesea: Hundreds of garbage commented out lines and you don't even have a CALL to show us |
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| 23:07.30 | Micc | I don't really care about the cisco uc540 registering weird extensions at this point. Its registering the main peer fine. But now when calls are placed to it the uc540 looks like its rejecting because the sip address we are sending is the external IP, but its checking it against the internal ip. Is there anyway to tell asterisk to use the internal IP in the sip invite headers? |
| 23:07.43 | Micc | I guess it would only need to be the to header. |
| 23:07.55 | Micc | otherwise the packet is getting to the device through the firewall just fine. |
| 23:09.34 | Micc | yeah it looks like just the To sip address in the invite has the external IP address, which it doesn't like. I'm sure there is a way to fix it on their side, but it would be nice to know if there is an asterisk trick to do it on my side too. |
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| 23:33.16 | Micc | I think I was wrong, I need not change the To header but the invite request URI. |
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| 23:46.15 | autojack | does anyone have a US number I can dial that always rings busy? having a hard time finding one that works. |
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