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| 00:30.01 | drmessano | Hmmm |
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| 00:48.04 | carrar | moof |
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| 01:10.31 | Hectaman | Can calls be routed based on distinctive ring? |
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| 01:17.55 | jpsharp | Yes, if you're using a DAHDI compatible card. |
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| 01:38.29 | gboudreau | hi guys. |
| 01:38.42 | gboudreau | what would be the easiest way to share a phone between multiple persons. |
| 01:39.09 | gboudreau | each person would have his extension, and when the phone would ring, the callerid would show something that would tell who the call is for |
| 01:39.28 | gboudreau | like maybe prefixing the real caller id with the extension that the incoming caller dialed. |
| 01:39.33 | WIMPy | What kind of phone? |
| 01:40.00 | gboudreau | snom 300 |
| 01:40.13 | WIMPy | Use multiple accounts. |
| 01:40.54 | gboudreau | so nothing to configure or any special dial plan to create in asterisk. just connect the phone to all the accounts I created in sip.conf ? |
| 01:41.13 | WIMPy | That's what I'd do. |
| 01:41.29 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 01:41.49 | gboudreau | ok. good. I'll check how many identities the phone supports. I'd need 4 per phone. |
| 01:41.49 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 01:42.04 | WIMPy | Just treat them as if they were different phones, even if they all get registered by the same device. |
| 01:43.22 | gboudreau | 4 identities, says the data sheet. perfect! :) |
| 01:43.37 | gboudreau | well, that was simpler than I expected. |
| 01:43.50 | gboudreau | thx for the pointer |
| 01:48.47 | gboudreau | on a not really related note... would there be somewhere a good guide on securing an asterisk install ? like bullet points on what should be checked, to insure everything is properly locked down. or even better, a web service that would poke my server to look for known bad configurations? |
| 01:49.01 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 01:51.06 | WIMPy | There's a readme.bestpractices that should cover most points. |
| 01:55.37 | gboudreau | ./README-SERIOUSLY.bestpractices.txt nice. will check that out. thx again. |
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| 02:33.27 | *** join/#asterisk Yourname` (~whocares@unaffiliated/yourname/x-837320) |
| 02:33.49 | Yourname` | Hello, Retransmitting nat unauthorized means firewall issues for sure? |
| 02:34.09 | *** part/#asterisk gboudreau (~gboudreau@mail2.danslereseau.com) |
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| 02:36.21 | jpsharp | No. |
| 02:36.48 | Yourname` | jpsharp: What is it in majority of cases? |
| 02:36.58 | jpsharp | Or rather, chances no. Need to see the whole debug message to message to besure. |
| 02:38.53 | Yourname` | jpsharp: http://pastebin.ca/2199540 |
| 02:40.45 | jpsharp | That's definitely not firewall. There's two way communications between Asterisk & the SIP client. |
| 02:40.59 | jpsharp | You've got a misconfiguration in authentication/usernames/passwords. |
| 02:41.47 | Yourname` | jpsharp: Let me make sure.. |
| 02:42.02 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 02:42.13 | jpsharp | Your client is sending the INVITE, asterisk is saying "You're not authorized", the client is saying "ACK", then asterisk is saying again "No, you're still not authorized". |
| 02:42.30 | Yourname` | lol |
| 02:42.50 | jpsharp | I'm assuming 192.168.1.252 is Asterisk and 1.15 is your client. |
| 02:43.20 | Yourname` | jpsharp: Yup |
| 02:43.49 | jpsharp | Yep, check your usernames & passwords & authentication then. There's no firewall issue. |
| 02:44.19 | Yourname` | jpsharp: Username/pwd is good. Which part of "authentication" is left? |
| 02:45.20 | jpsharp | It looks like asterisk is expecting authentication, but the client doesn't know to send it. |
| 02:46.27 | Yourname` | You mean realm? |
| 02:48.17 | Yourname` | nvmd, one sec |
| 02:49.36 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 02:50.43 | Yourname` | Whoa, you're right. It says "Forbidden (bad auth" in the debug when I'm trying to register from the client. |
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| 02:53.24 | Yourname` | jpsharp: Is there a way to check what the password is being sent as? It keeps saying bad auth, and I don't see why.. I changed the password to something VERY SIMPLE as well... |
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| 03:13.53 | Hectaman | jsharp: thanks |
| 03:18.00 | Yourname` | jpsharp: It worked out, had to remove the password .. and somehow, it worked. Thanks for the guidance! |
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| 05:37.27 | MACscr | ok, so i swear on my polycom 330 when I would be on the phone and someone else called my extension, my second line would light up. This way i could put my current call on hold and except the the other call. This doesnt appear to be happening anymore. Any idea what settings I should be looking at? Think its more a phone setting or a system setting? |
| 05:40.19 | MACscr | That wasnt just regular call waiting is it? |
| 05:40.47 | ChannelZ | probably |
| 05:41.23 | MACscr | guess i need to enable that and test. Odd that the feature is now off though for my extension if that was the case |
| 05:43.13 | MACscr | lol, wth. It was that simple |
| 05:45.36 | ChannelZ | Yay! |
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| 05:54.37 | kaldemar | MACscr: reg.1.lineKeys="1" reg.1.callsPerLineKey="1" in the phone config do matter. |
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| 06:16.43 | v0lZy | lo |
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| 07:51.53 | bulkorok | hi... is it possible to put the uniqueid in the AMI Event FaxStatus after finished fax sending?! |
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| 08:59.10 | beebeeep | hey folks, did anybody tried to access AstDB using external programs? |
| 09:00.28 | fenrus | its a Berkeley DB.. |
| 09:23.05 | beebeeep | fenrus: i know, looks like reading is not a problem, but i'm not sure about writing - looks like * is caching db file in some way - i.e. if you delete it, 'database show' will show whole db |
| 09:39.26 | kaldemar | astdb is sqlite in 10 and newer versions. |
| 09:48.22 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 10:03.06 | rolandow | is it possible that old dect repeaters are functioning without configuring them? |
| 10:03.46 | rolandow | it seems that i can have only two concurrent calls, so i think the signal is going through the repeater that are installed here for the old (analog) dect system |
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| 11:23.13 | WIMPy | That was a pretty strange thing. IIRC you either need to pair the repeater with the base for transparent operation or you have to pair the mobile with the base through the repeater. |
| 11:28.26 | rolandow | so it shouldn't do that by itself |
| 11:31.47 | *** join/#asterisk last1 (~last1@79.113.142.153) |
| 11:33.21 | last1 | I'm registering from my mobile phone to my Asterisk server and I can receive calls just fine |
| 11:33.31 | last1 | I can also make calls but after ~20 seconds the connection dies |
| 11:34.11 | last1 | the error on the phone is: Call 0 is DISCONNECTED [reason=408 (Request Timeout)] |
| 11:34.28 | WIMPy | rolandow: I don't think it can. |
| 11:35.42 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 11:36.32 | rolandow | hm... then it surprises me that i received a "busy signal here" when only two concurrent calls were going on .. the dect station should be able to handle 4. |
| 11:37.13 | WIMPy | rolandow: Do you have to enable that, perhaps? |
| 11:38.39 | rolandow | can't find the option in the web interface though .. |
| 11:39.25 | WIMPy | I've seen an "yes, I have more than the minimum DSL bandwith" checkbox on those kind of things. |
| 11:39.56 | WIMPy | Although most seem to be limited to two calls anyway. |
| 11:40.23 | WIMPy | Probably because they re-used parts from BRI stations. |
| 11:41.19 | *** join/#asterisk irroot (~gregory@2002:294f:dde2:1:b5ce:5681:7113:dadc) |
| 11:41.30 | WIMPy | Hi irroot! |
| 11:42.09 | *** part/#asterisk irroot (~gregory@2002:294f:dde2:1:b5ce:5681:7113:dadc) |
| 11:43.34 | *** join/#asterisk pbxMan (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 11:43.40 | pbxMan | hello |
| 12:02.48 | *** part/#asterisk deo (~dnepangue@222.127.13.226) |
| 12:03.19 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 12:03.23 | v0lZy | hey guys |
| 12:03.26 | v0lZy | Erm.. some help |
| 12:03.31 | v0lZy | For some reason ic ant make outgoing calls |
| 12:04.09 | v0lZy | Dial("SIP/16-00000024", "SIP/386801000@SoftNET,,T") in new stack |
| 12:04.09 | v0lZy | <PROTECTED> |
| 12:04.09 | v0lZy | <PROTECTED> |
| 12:04.09 | v0lZy | <PROTECTED> |
| 12:04.24 | v0lZy | what would this issue be? |
| 12:05.14 | rolandow | your trunk is out of lines? |
| 12:05.55 | v0lZy | hmm |
| 12:05.57 | v0lZy | misconfigured i suppose |
| 12:06.00 | v0lZy | i can get incoming calls just fine |
| 12:06.03 | v0lZy | cant make outgoing ones |
| 12:07.03 | rolandow | how can it suddenly be misconfigured.. was it working before? |
| 12:07.06 | rolandow | did you change anything? |
| 12:07.52 | v0lZy | never had it working before |
| 12:07.58 | v0lZy | i suspect my default context |
| 12:08.01 | v0lZy | but i dont know |
| 12:08.07 | v0lZy | i have a pbx on which it works |
| 12:08.12 | v0lZy | now i switchedips fixed the firewall and nada |
| 12:10.41 | Roelt | registration fails? |
| 12:12.03 | beebeeep | v0lZy: sip set debug peer SoftNET |
| 12:12.05 | v0lZy | no |
| 12:13.00 | v0lZy | hm |
| 12:13.57 | v0lZy | http://pastebin.com/SFA11hRp |
| 12:15.06 | WIMPy | 404 Not found |
| 12:15.18 | WIMPy | Looks like the called number doesn't exist. |
| 12:15.28 | WIMPy | Probably it's in the wrong format. |
| 12:15.32 | v0lZy | hm |
| 12:15.33 | v0lZy | but its not |
| 12:15.48 | v0lZy | im doin SIP/blahblah@SoftNET,,Tt |
| 12:15.59 | beebeeep | mb you should ask your provider for correct dialing plan |
| 12:16.09 | beebeeep | or just try various formats |
| 12:16.10 | v0lZy | i have it working in the other pbx... |
| 12:16.33 | beebeeep | Reason: Q.850;cause=1;text="Unallocated (unassigned) number" |
| 12:17.32 | WIMPy | That's even more clear, yes. |
| 12:17.51 | v0lZy | so ... my dialings wrong |
| 12:17.52 | v0lZy | ok.. sec |
| 12:18.57 | v0lZy | im supposed to send in national id format |
| 12:19.00 | v0lZy | 38618100100 for example |
| 12:20.09 | v0lZy | hmzz |
| 12:21.48 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:22.25 | v0lZy | must be my dialplan |
| 12:22.31 | v0lZy | i can dial |
| 12:22.38 | v0lZy | 0018100100 without a problem |
| 12:22.40 | v0lZy | but not |
| 12:22.45 | v0lZy | 0040233688 |
| 12:22.56 | WIMPy | >Probably it's in the wrong format. |
| 12:23.28 | WIMPy | 00? What kind of format is that? |
| 12:25.36 | v0lZy | http://bpaste.net/show/qOCuFda5iLpQc2lXhWyd/ |
| 12:25.42 | v0lZy | this is my dialplan |
| 12:25.58 | v0lZy | I press 0 to dialout WIMPy, then i enter the 040233688 number |
| 12:26.45 | WIMPy | Ok, but you obviousely don't send the first 0 to your provider. |
| 12:26.58 | v0lZy | its supposed to call |
| 12:27.00 | v0lZy | ${EXTEN:2},386,${CALLERID(num)} |
| 12:27.06 | WIMPy | Ok. |
| 12:27.20 | v0lZy | n,Dial(SIP/${ARG2}${ARG1}@${ARG6},,T) |
| 12:27.30 | WIMPy | I see you cut the 0 if you have one 0 prefix. |
| 12:27.47 | WIMPy | But on the national/international version, you cut all zeros. |
| 12:27.58 | v0lZy | so its dialing SIP/38640233688@SoftNET right ... ${EXTEN:2} |
| 12:28.21 | WIMPy | You will have to leave the others in place. How else is your provider to know you're not calling a local number? |
| 12:28.37 | v0lZy | WIMPy: my provider wants the full country code |
| 12:28.39 | v0lZy | for all calls |
| 12:28.42 | v0lZy | if i make a local call |
| 12:28.56 | v0lZy | its 386CityCodePhone |
| 12:29.09 | v0lZy | if i call outside my city... its 386CityCodePhone |
| 12:29.24 | v0lZy | if i call into another ocuntry, its countrycodecitycodephone |
| 12:29.24 | v0lZy | etc |
| 12:29.26 | v0lZy | so |
| 12:29.28 | WIMPy | I starting to see through your dialplan. |
| 12:29.48 | v0lZy | 0 018100100 works as expected |
| 12:29.56 | v0lZy | 0 040 233 688 doesnt work |
| 12:30.31 | v0lZy | ihave no idea why |
| 12:30.35 | v0lZy | 0 018100100 |
| 12:30.41 | v0lZy | 0 040233688 |
| 12:31.09 | WIMPy | So your area code is just 1? |
| 12:31.45 | v0lZy | yeah |
| 12:31.59 | v0lZy | capital city |
| 12:32.11 | v0lZy | its actually 01 but the 0 is to dial upwards |
| 12:32.12 | v0lZy | so |
| 12:32.18 | v0lZy | if i call somone in my city |
| 12:32.19 | v0lZy | i omit 01 |
| 12:32.23 | v0lZy | for example |
| 12:32.30 | WIMPy | Ja, ok. |
| 12:32.48 | v0lZy | 580 93 16 is my number... from my city i just call taht. if i move to anotehr city i have to dial 01 580 93 16 |
| 12:32.56 | v0lZy | 040 233 688 is a mobile number |
| 12:32.59 | WIMPy | That seems to look ok. |
| 12:33.04 | v0lZy | from my phone i should dial |
| 12:33.24 | v0lZy | 0040233688 because my dialplan is supposed to require users to press 0 before dialing out |
| 12:34.20 | v0lZy | according to askoziapbx where this work |
| 12:34.21 | v0lZy | s |
| 12:34.22 | v0lZy | 3861+0|XX. |
| 12:34.22 | v0lZy | 386+00|XX. |
| 12:34.22 | v0lZy | 000|X. |
| 12:34.26 | *** part/#asterisk MACscr (~Adium@c-98-214-103-147.hsd1.il.comcast.net) |
| 12:34.36 | v0lZy | local, national international. |
| 12:35.32 | v0lZy | + - adds a prefix (i.e. "1+555" matches "555" and passes "1555" to the provider) |
| 12:35.32 | v0lZy | | - removes a prefix (i.e. "1|555" matches "1555" but only passes "555" to the provider) |
| 12:36.11 | WIMPy | Maybe you should change the end of your NoOps fomr ${ARG1} to ${ARG2}${ARG1} as you dial to see if that's really in the correct format. |
| 12:36.16 | leifmadsen | I've never seen that syntax for pattern matching... o>O |
| 12:36.28 | v0lZy | its askoziapbx, dont worry |
| 12:36.31 | v0lZy | WIMPy: ill try |
| 12:36.35 | leifmadsen | ah that's what I figured |
| 12:38.29 | v0lZy | 38640233688 |
| 12:38.31 | v0lZy | thats what i get |
| 12:39.11 | WIMPy | Which is correct as far as I git it. |
| 12:39.13 | WIMPy | got |
| 12:40.13 | WIMPy | So it loks like they either gave you wrong information or there is some other issue at their end. |
| 12:40.35 | v0lZy | well according to askozia ... im matching 00XX. adding 386 and cutting 00 |
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| 12:42.59 | v0lZy | hm |
| 12:43.03 | v0lZy | according to askozia where it works |
| 12:43.04 | v0lZy | _0XX.,n,Dial(SIP/3861${EXTEN:1}@SIP-PROVIDER |
| 12:43.29 | v0lZy | _00XX.,n,Dial(SIP/386${EXTEN:2}@SIP-PROVIDER |
| 12:43.49 | v0lZy | _000X.,n,Dial(SIP/${EXTEN:3}@SIP-PROVIDER |
| 12:44.54 | v0lZy | i have |
| 12:44.57 | v0lZy | Gosub(AppProvider1,OutgoingLocal,1(${EXTEN:1},3861, ... |
| 12:45.25 | v0lZy | Gosub(AppProvider1,OutgoingNational,1(${EXTEN:2},386, |
| 12:45.57 | v0lZy | Gosub(AppProvider1,OutgoingInternational,1(${EXTEN:3},,... |
| 12:46.19 | v0lZy | so when it gets put together ... its the same |
| 12:46.26 | v0lZy | 3861${EXTEN:1} |
| 12:46.33 | v0lZy | 386${EXTEN:2} |
| 12:46.42 | v0lZy | and ${EXTEN:3} |
| 12:47.07 | v0lZy | funny thing i cant call 00038640233688 |
| 12:50.42 | v0lZy | i called them |
| 12:50.45 | v0lZy | and they see my number as |
| 12:50.46 | v0lZy | 016 |
| 12:50.52 | v0lZy | instead of 015809316 |
| 12:50.58 | v0lZy | callerID appears to be the problem |
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| 12:53.23 | WIMPy | that's a different story. |
| 12:53.44 | WIMPy | Try to finde the difference beween the call that works and the one that doesn't. |
| 12:54.53 | v0lZy | the one that works works because im calling my provider |
| 12:54.59 | v0lZy | and they let it pass since its inside their network |
| 12:55.02 | v0lZy | got the issue |
| 12:55.04 | v0lZy | will fix. |
| 12:56.34 | v0lZy | btw |
| 12:56.35 | v0lZy | if i use |
| 12:56.44 | v0lZy | Set(CALLERID(all)=) |
| 12:56.51 | v0lZy | will it set all parameters to the same value? |
| 12:56.59 | v0lZy | or is it like first parameter is name,s econd parameter is number etc? |
| 12:57.11 | WIMPy | Don't use (all). |
| 12:57.16 | v0lZy | num and name? |
| 12:57.20 | v0lZy | just num? |
| 12:57.26 | v0lZy | any sense in passing name? |
| 12:57.52 | WIMPy | That does the old Name <num> thing, but it may be more readable if you seperate them. |
| 12:58.23 | WIMPy | It might work when you calls other customers of the same provider. |
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| 13:01.54 | v0lZy | yeah i was sending as name num thing |
| 13:02.00 | v0lZy | but it didnt work apaprently |
| 13:02.05 | [TK]D-Fender | <PROTECTED> |
| 13:02.35 | v0lZy | Set(CALLERID(all)=Abc <123>) |
| 13:02.37 | v0lZy | would that be ok? |
| 13:02.41 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 13:02.45 | v0lZy | or i reckon is hould set each individually |
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| 13:04.35 | [TK]D-Fender | you should |
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| 13:07.46 | v0lZy | can i use more than one setvar parameter per context in sip.conf? |
| 13:08.16 | WIMPy | yes |
| 13:08.49 | v0lZy | ah, ok, know what to do then |
| 13:08.51 | v0lZy | thanks |
| 13:18.20 | v0lZy | gotta run now |
| 13:18.20 | v0lZy | bye |
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| 13:23.36 | CommaCrazy | hi all, question has anyone had a problem with a Linksys IP Phone SPA-942 where it starts buzzing when the handset is picked up and where if the speaker button is pressed it reboots. |
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| 13:24.07 | CommaCrazy | Also I tried doing the factory reboot but with no use because it reboots before it does the factory reboot |
| 13:24.37 | WIMPy | Don't connect the etherkiller to the handset port :-) |
| 13:24.42 | WIMPy | Sounds bad. |
| 13:24.56 | WIMPy | Have you tried another PSU? |
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| 13:50.43 | bitglue | does anyone know of a good source of information on statistical analysis of inbound call centers? |
| 13:51.35 | bitglue | i've read all the wikipedia articles on the underlying concepts, and found some online calculators that spit out numbers, but i'd like to fill the gap between, and understand how to apply the math myself. |
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| 14:02.30 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 14:05.24 | longst | I am wondering how I could use Asterisk Manager GetVar action fetch data from Asterisk RealTime |
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| 14:08.30 | Qwell | from realtime? |
| 14:09.03 | Qwell | realtime doesn't "do" things, so I'm not sure how you could do something "from" it. |
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| 14:17.08 | bitglue | longst: I think GetVar gets channel variables, which have only an indirect relationship to realtime. What exactly are you trying to accomplish? |
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| 14:20.34 | navaismo | mooorning! |
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| 14:30.05 | longst | For example, if I would like to query data from realtime in Asterisk Manager. according to Asterisk Manager API there is no dedicated action to query realtime data. I am wondering if would be possible in someway use getvar to query asterisk realtime data |
| 14:33.12 | *** join/#asterisk Diffen (~diffen@c-83-233-59-209.cust.bredband2.com) |
| 14:33.17 | danfromuk | Hi, anyone know if there are any SIP phones that have open source firmware? We've been trying to get shared line appearance working on a panasonic kx-ut133x but its quite buggy. Panasonic tech support aren't high enough in the development chain to be able to assist. |
| 14:34.32 | ectospasm | not likely to find a phone that has open source firmware, unless it's open hardware... |
| 14:35.31 | Diffen | Evning. If I have three extensions in a queue and simultainous ringing on. Is it possible to see how many "missed" calls an extension have? Is that something you can find in the queue.log file? |
| 14:35.41 | danfromuk | shame. |
| 14:36.00 | danfromuk | leifmadsen: are you available at the moment? |
| 14:36.13 | WIMPy | danfromuk: DIY |
| 14:36.19 | Katty | the world has gone insane. |
| 14:36.25 | danfromuk | DIY? |
| 14:36.39 | WIMPy | danfromuk: Build your own phone. |
| 14:37.09 | [TK]D-Fender | Katty, No, it's been there a long long time ..... it occasionally remembers it for brief intervals however. |
| 14:37.14 | danfromuk | Wouldnt have a clue where to start and certainly wouldnt be worth the cost. |
| 14:37.28 | bitglue | longst: i think the answer is "no". There may be ways to get the data you want, but GetVar is not it. Perhaps if you tell us specifically what you are trying to do, we can tell you how to do it. |
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| 14:37.47 | WIMPy | danfromuk: Take a RaspberryPI and some time. |
| 14:37.51 | *** join/#asterisk seik0 (3eb61f96@gateway/web/freenode/ip.62.182.31.150) |
| 14:37.53 | bitglue | Diffen: if I understand what you are asking, yes. AGENTNOANSWER, iirc. |
| 14:37.56 | [TK]D-Fender | danfromuk, remember what I told you about this : colossal waste of time. |
| 14:38.03 | Katty | [TK]D-Fender: mister fender... |
| 14:38.12 | specing | danfromuk: arm9 + linux + 320x200 rgb display + keypad + case |
| 14:38.12 | [TK]D-Fender | danfromuk, Polycom's you CAN change the colour on though. |
| 14:38.14 | Katty | [TK]D-Fender: why can't you say why yes...yes it is |
| 14:38.23 | Katty | [TK]D-Fender: why must you constantly disagree with everything |
| 14:38.24 | bitglue | Diffen: actually RINGNOANSWER: http://www.asterisk.org/astdocs/node271.html |
| 14:38.33 | Diffen | bitglue thanks |
| 14:38.36 | Katty | [TK]D-Fender: please go look up the word CHARM. |
| 14:38.37 | specing | WIMPy: RPI is a toy, you can't do much serious on it |
| 14:38.41 | [TK]D-Fender | Katty, Because that would mean supporting a lie :) |
| 14:38.43 | Katty | [TK]D-Fender: and do be more charming. |
| 14:38.51 | Katty | Qwell is very charming. |
| 14:38.54 | Katty | leifmadsen is charming |
| 14:39.07 | seik0 | hi everybody. If someone could, may you give links to how call transfers works (internall - when and which channels are created and dropped) & |
| 14:39.12 | danfromuk | [TK]D-Fender: yes, you've said it a few times and i have said that its an important sale and therefore worth the time. |
| 14:39.14 | Katty | it would not hurt you to develop that lovely trait. |
| 14:39.32 | WIMPy | specing: Should be more than adequate for a phone and the price is good. |
| 14:39.49 | [TK]D-Fender | Katty, Ferengi Rules of Acquisition # 48 :The bigger the smile, the sharper the knife |
| 14:39.54 | specing | WIMPy: Im not sure you have enaugh IOs for a dialpad |
| 14:40.20 | specing | [TK]D-Fender: lol |
| 14:40.26 | [TK]D-Fender | danfromuk, Polycom <- IP 650's it is |
| 14:40.36 | [TK]D-Fender | danfromuk, and not "open", just "configurable" |
| 14:40.47 | longst | I am trying to use Asterisk Manager API query data from realtime. From Asterisk CLI, we could use realtime load <realtime-family> <key> <matched key value> query realtime I am wondering how I could make this type of query from Manager API. |
| 14:41.11 | [TK]D-Fender | danfromuk, What are these sites going to use for connectivity? |
| 14:41.25 | WIMPy | specing: Isn't there plenty of I/O? I always mix up all these gadgets. |
| 14:41.32 | *** join/#asterisk jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 14:42.15 | seik0 | so, could anyone help? |
| 14:42.48 | [TK]D-Fender | seik0, Your question is a bit too vague |
| 14:43.07 | [TK]D-Fender | seik0, there are multiple types of transfers |
| 14:44.16 | seik0 | [TK]D-Fender mm, I want to know how attended call transfer works |
| 14:44.35 | specing | WIMPy: afaik no |
| 14:44.35 | [TK]D-Fender | seik0, there are multiple kinds. What kind of phone? Triggered how? |
| 14:45.52 | danfromuk | [TK]D-Fender: Adsl |
| 14:46.03 | seik0 | [TK]D-Fender triggered by sending dtmf to asterisk |
| 14:46.06 | [TK]D-Fender | danfromuk, ITSP you mean |
| 14:46.20 | [TK]D-Fender | seik0, What phones? |
| 14:46.46 | seik0 | [TK]D-Fender sip |
| 14:46.49 | danfromuk | [TK]D-Fender: yes. all connected via sip (or any other IP method if it a suitable method is available) |
| 14:46.58 | [TK]D-Fender | danfromuk, since there is no such thing as really distinct channels I'm not sure how you intend to break that into "lines". for SLA. |
| 14:47.17 | [TK]D-Fender | danfromuk, The rabbit hole gets DEEPER..... |
| 14:47.46 | Diffen | bitglue Hmm I dont see the calls that are RINGNOANSWER if someone picks up the call. Then I just see that call in the queue.log. Is there anyway to set the level of detail that are shown in queue.log? |
| 14:47.46 | pabelanger | longst: why are you using AMI to get access to realtime? Why not just connect to the database directly? |
| 14:48.16 | danfromuk | [TK]D-Fender: rabbit hole? Actually, Asterisk seems to be handling SLA very nicely over SIP. Theres just a few bugs in the Panasonic firmware. |
| 14:48.17 | seik0 | [TK]D-Fender in fact, it's just start point, then, I need to understand what is the difference for asterisk when transfer is made through Flash (transfer key) on phones, attached to analog switch |
| 14:48.24 | danfromuk | - To take a call off hold, I need to press the Shared Line Key twice? When I press it once, it says "Busy" on the screen. |
| 14:48.30 | danfromuk | When a call is placed on hold, the line key remains solid red so there is no way to identify HELD calls from ACTIVE calls |
| 14:48.34 | danfromuk | The Shared Line key is red even if it is in-use by the local user (it should be green if possible when in use by the local user) |
| 14:48.37 | danfromuk | Thats all |
| 14:49.04 | danfromuk | They are bugs because ive checked the SIP debug for each bug and confirmed it |
| 14:49.52 | [TK]D-Fender | danfromuk, that double "busy" is the phone's doing to suggest you don't really want to disturb that "person" |
| 14:50.14 | [TK]D-Fender | This isn't a "bug" this is the phone trying to warn you before annoying people. |
| 14:50.41 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 14:51.04 | seik0 | [TK]D-Fender in the case of analog switch, transfer is made through that switch (it's heard by MoH not specific to asterisk) and, I think, asterisk is not acknowledged of the fact of transfer made |
| 14:51.53 | [TK]D-Fender | seik0, What is this analog switch you are bringing into the process? You are continuing to be vauge and I am not seeing a contrete goal in this. |
| 14:52.23 | [TK]D-Fender | vague* |
| 14:52.48 | danfromuk | [TK]D-Fender: interesting. Any idea if its possible to disable that feature? |
| 14:53.10 | [TK]D-Fender | danfromuk, It's phone-based... check the manual. I've never heard of ANYONE using the ones you've got. |
| 14:53.31 | longst | pabelanger: for some reason we don't want to directly expose database. I wish queries go through Asterisk Realtime, moreover, in this way it also avoid to develop connections for different databases. |
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| 14:54.24 | danfromuk | [TK]D-Fender: waiting for panasonic to get back to me about the other bugs. maybe they'll have a hidden way to turn it off. they've shown be a few hidden sections to the web based config. |
| 14:54.36 | [TK]D-Fender | danfromuk, What bugs? |
| 14:55.01 | danfromuk | When a call is placed on hold, the line key remains solid red (on all phones) so there is no way to identify HELD calls from ACTIVE calls. |
| 14:55.35 | danfromuk | I've checked the sip logs, and all phones receive a sip packet showing that the call has been placed on hold/picked up |
| 14:55.35 | seik0 | [TK]D-Fender another try ), let's have two analog phones, connected to panasonic kx-tda100 phone system, which connected to asterisk |
| 14:55.55 | [TK]D-Fender | danfromuk, dif you get that SIP DEBUG prrof I told you to get days ago? |
| 14:56.10 | [TK]D-Fender | seik0, Connected to * HOW? |
| 14:56.10 | danfromuk | Yes. One moment. |
| 14:57.36 | seik0 | [TK]D-Fender you make me think ) |
| 14:58.07 | danfromuk | [TK]D-Fender: http://pastebin.com/ajcCqSKj |
| 14:59.20 | danfromuk | It seems that <param pname="+sip.rendering" pvalue="no"/> identifies that a call is on hold, and without that it means that the call is currently talking. |
| 14:59.22 | bitglue | Diffen: well, if someone answers, then I wouldn't expect to see RINGNOANSWER. RINGNOANSWER means that the ACD attempted to call a queue member, and the member did not answer. I'm confused about what you are asking. |
| 14:59.31 | [TK]D-Fender | danfromuk, "state" says the same... |
| 14:59.47 | bitglue | Diffen: or do you want to know calls that were made to members, but they didn't answer, because someone else did? |
| 15:00.11 | [TK]D-Fender | danfromuk, And may not be a "bug". Polycom doesn't indicate "ringing" last I checked..... could be an incomplete featureset. |
| 15:00.17 | seik0 | [TK]D-Fender * uses Zap interface to communicate with panasonic |
| 15:00.37 | [TK]D-Fender | danfromuk, And the words "good luck" come to mind in getting a change like that done if the state you're loking to accomodate isn't supported. |
| 15:01.29 | [TK]D-Fender | seik0, So you have hook-switch to do transfers, or DTMF. what is the real question about them? |
| 15:01.45 | danfromuk | [TK]D-Fender: I wonder if i can patch asterisk to send a different state. its already sending a sip packet to all the phones when a call is placed on hold. |
| 15:02.12 | danfromuk | [TK]D-Fender: did you see my message above regarding the <param pname= tag? |
| 15:02.13 | [TK]D-Fender | shrugs |
| 15:02.33 | seik0 | [TK]D-Fender hook-switch is that configured in features.conf? |
| 15:02.38 | [TK]D-Fender | danfromuk, That param didn't give me the impression that you alluded to |
| 15:02.41 | seik0 | [TK]D-Fender or what do u mean? |
| 15:03.01 | danfromuk | [TK]D-Fender: i agree but i wonder why its there when the call is held, and missing when the call is picked up |
| 15:03.03 | *** part/#asterisk eureka^ (~eureka@s5.azn.so) |
| 15:03.23 | danfromuk | What would be the perform of the sip packet if not to update all the phones about the hold state? |
| 15:03.24 | seik0 | [TK]D-Fender maybe i'm still not clear? |
| 15:03.25 | [TK]D-Fender | seik0, No, that's set in chan_dahdi.conf or zapata.conf depending on version |
| 15:04.30 | [TK]D-Fender | danfromuk, Perhaps just to reinforce for ones that weren't around for previous state changes and so that phones already in the right state don't have to pay attention. |
| 15:05.41 | bmoraca_work | Qwell: just FYI, 10.6.0-digiumphones with DPMA 1.2.0 on that machine that I was having trouble with works. |
| 15:06.19 | danfromuk | [TK]D-Fender: perhaps. however its certainly not hopeless. |
| 15:06.40 | [TK]D-Fender | danfromuk, No, hope is the very last thing they take away from you :) |
| 15:06.43 | seik0 | [TK]D-Fender so, we have both configured hook-switch and #-transfer (in features) |
| 15:07.02 | [TK]D-Fender | seik0, waste of time. DTMF transfers = ass. Analog on a whole = ass |
| 15:08.56 | Qwell | bmoraca_work: of course it does... |
| 15:09.04 | Qwell | bmoraca_work: I'll try to lab it up here today. |
| 15:09.05 | bmoraca_work | yep |
| 15:11.54 | bitglue | i'd like to implement something so when a user registers from a new handset, i can send a message to the handset previously registered as that user to log out. Is there an event handler or such I can create in asterisk to do this? |
| 15:12.03 | seik0 | [TK]D-Fender when one side is on analog uses transfer through hook asterisk know nothing about that transfer? |
| 15:12.40 | [TK]D-Fender | seik0, The DAHDI channel driver signals to * for this. |
| 15:12.50 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:13.17 | seik0 | [TK]D-Fender thanks for help, i learned alot ), now i'm off |
| 15:13.17 | Diffen | bitglue exactly. i want to know how many calls each agents get and that are not answered. so during a day i can see that agent 1 have missed 30 calls out of 35. |
| 15:13.36 | seik0 | [TK]D-Fender i can't use dahdi for now ( |
| 15:13.40 | [TK]D-Fender | bitglue, nope. You could probably use a similar watcher like fail2ban to catch the registration and keep track of the predecessor yourself. |
| 15:14.04 | [TK]D-Fender | seik0, chan_zap, etc. |
| 15:14.04 | [TK]D-Fender | ^^^same thing |
| 15:14.15 | bmoraca_work | Qwell: i guess the next challenge is whether or not it works on another machine with 10.6.0-digiumphones first time without any trouble |
| 15:14.19 | danfromuk | [TK]D-Fender: just fyi, asterisk hints show that asterisk SLA considers the call HELD and therefore should probably update the phones BLF state. |
| 15:14.20 | bmoraca_work | i'll check that here this morning |
| 15:14.56 | bitglue | Diffen: what strategy are you using for your queue? |
| 15:15.19 | [TK]D-Fender | danfromuk, Possible... time to wait on Panasonic. Hopefully you are using this time to test multiple Aastra's as well... their presence implementation seems pretty solid. |
| 15:15.28 | Diffen | round robin |
| 15:15.30 | *** part/#asterisk seik0 (3eb61f96@gateway/web/freenode/ip.62.182.31.150) |
| 15:15.34 | Diffen | or ring all |
| 15:16.29 | danfromuk | [TK]D-Fender: i'll see if there are any demo models available at the distributor. Do you know if any models have at least 24 flexible keys? |
| 15:17.19 | *** join/#asterisk jmls1 (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 15:17.21 | [TK]D-Fender | danfromuk, yeah you were really hoping for no side-car too IIRC... |
| 15:17.26 | bitglue | Diffen: well, with round robin, you should see RINGNOANSWER each time a member does not answer, because only one is called at a time. However with ringall, you only get RINGNOANSWER if no one answers. Is that not what you expect? |
| 15:17.35 | danfromuk | [TK]D-Fender: correct |
| 15:17.44 | leifmadsen | danfromuk: Digium phones have flexible line keys with paging |
| 15:17.50 | [TK]D-Fender | danfromuk, Yuo can paginate your way though that one. Digim's D70 might be a fit as well. Bu no phone I've seen has that much in 1 shot |
| 15:17.53 | Qwell | leifmadsen: Heck yeah they do. |
| 15:17.59 | bitglue | Diffen: ie, if someone calls the queue, and five agents ring (ringall), one answers, the other four didn't "miss" the call. They just didn't answer it before the agent who did. |
| 15:18.02 | [TK]D-Fender | danfromuk, "Would you like fries with that, sir?" |
| 15:18.11 | *** join/#asterisk jpcansa (~JP@ip122-130-64-186.ct.co.cr) |
| 15:18.23 | *** part/#asterisk WolWid (WolWid@pD9ED49B2.dip.t-dialin.net) |
| 15:18.25 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 15:18.29 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:18.35 | leifmadsen | paging as in, switching which line keys are active with left/right keys, not overhead paging |
| 15:18.55 | Diffen | bitglue ok so thats why its not showing in the log under RINGNOANSWER.. I see. |
| 15:19.12 | [TK]D-Fender | danfromuk, Your project specification is to put it lightly, a flaming pile of shit. So nauseatingly specific they may as well not ask for a bid. Only 1 model out there is likely to be exactly what they want and are deluding themselves into believing they'll find something else exactly ilke it. |
| 15:19.31 | [TK]D-Fender | danfromuk, or in short : |
| 15:19.35 | [TK]D-Fender | ~wglwat |
| 15:19.35 | infobot | wglwat is, like, well, good luck with all that |
| 15:19.43 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:19.49 | danfromuk | [TK]D-Fender: i spend most of my time trying to get faulty firmware to work properly. Spent hours trying to get a snom300 to work through a NAT to find out that it was a firmware bug. |
| 15:20.03 | [TK]D-Fender | danfromuk, Hope this process doesn't waste too much of your business' attentions and resources. |
| 15:20.04 | bitglue | Diffen: if you need to know who was ringing, even when someone else answers on the same attempt, i do know the console log will show "foo is ringing" if you turn verbose up to 3. Maybe you can parse that? Or, you can look in queue.log and see which agents were active, and just assume they were ringing for each call. |
| 15:20.29 | danfromuk | [TK]D-Fender: i like a challenge. |
| 15:20.35 | jpcansa | hi, i receive several E1s on my asterisk box on building A and i want to provide E1 terminations on building B, i have ethernet between A and B over fiber, what equipment can i use on building B to serve E1s?? |
| 15:21.16 | [TK]D-Fender | danfromuk, There is a difference between "challenge" and "fool's errand". Enjoy running that razor's edge :p |
| 15:21.35 | WIMPy | jpcansa: RAD has (or had) a quad E1 over Ethernet soloution. But I'd probably do it myself with a PC instead. |
| 15:21.38 | bitglue | "fool's challenge"? |
| 15:22.11 | danfromuk | [TK]D-Fender: you are an extremely negative person. From what I can see, asterisk is working fine and it appears to be a simple bug. |
| 15:22.23 | jpcansa | WIMPy, hi , its me again, can i do that with the Optimux-108? |
| 15:22.38 | [TK]D-Fender | jpcansa, You want to spit out E1 inside of building B arriving by ethernet from building A? |
| 15:22.53 | danfromuk | Does anyone know which asterisk source files are responsible for SLA? |
| 15:23.00 | [TK]D-Fender | danfromuk, Oh no.. not just THIS issue. ALL of the pieces around what the finished picture would take :) |
| 15:23.18 | [TK]D-Fender | danfromuk, This is as Jon Stewart would say, a "clusterfuck" |
| 15:23.19 | jpcansa | [TK]D-Fender, yes |
| 15:23.53 | [TK]D-Fender | jpcansa, Fonbridge RedPhone |
| 15:23.58 | [TK]D-Fender | j(sp?) |
| 15:24.01 | WIMPy | jpcansa: What EXACTELY do you want to provide? |
| 15:24.05 | *** part/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 15:24.12 | jpcansa | WIMPy, voice channels |
| 15:24.24 | [TK]D-Fender | WIMPy, take in E1 from A, spit across ethernet to B, tun BACk to E1. Basically an E1 bridge |
| 15:24.35 | WIMPy | jpcansa: Yes, that looks like the device I was thinking of. |
| 15:24.53 | jpcansa | WIMPy, but asterisk on A will do billing |
| 15:24.56 | danfromuk | [TK]D-Fender: We've got quite a stable hosted pbx platform with a control panel developed in house. Managed to get it working exactly as specified. We like building pictures. |
| 15:25.01 | WIMPy | jpcansa: Sounds like you could just put an Asterisk box at B. |
| 15:25.13 | Diffen | bitglue hmm i will try to see if we can work it out with round robin. Then it shows who have missed the call. A lot easier then mixing up with the console log and so on |
| 15:26.07 | danfromuk | [TK]D-Fender: everythings possible with money, and this client has basically given us a blank cheque to deliver the solution they want that uses sip. so if that means we need to rewrite asterisk or deliver a desktop application that displayed virtual shared lines, so be it. |
| 15:26.16 | WIMPy | jpcansa: Or if you want it fully transparent, you can use L1oIP, available from the kernel. |
| 15:26.25 | bitglue | Diffen: if the expectation is that your agents will always answer, that sounds like a good solution. There's also an option to kick agents that don't answer out of the queue. |
| 15:26.28 | jpcansa | WIMPy, what card will I use at the asterisk box on B? |
| 15:26.29 | Qwell | danfromuk: I like building pictures too. See http://tinyurl.com/8q5lx5o |
| 15:26.47 | [TK]D-Fender | danfromuk, Don't take my personal advice and assessment of the value of the fight you're up against as a personal slight. You seem stable in your approach, resourceful, and thorough. you may still be screwed but you in no way appear to be an "idiot". You are however in for a very rough ride and just want to make sure you have a good view of everything you're up against. |
| 15:27.10 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:27.10 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:27.13 | WIMPy | jpcansa: If you only want to do voice with Asterisk: Any. If you want a transparent connection, one that is supported by Linux. |
| 15:27.54 | danfromuk | Qwell: +1 |
| 15:28.08 | [TK]D-Fender | danfromuk, I just hope it doesn't cost you to much in the fight. And any positive changes you can cooerce these vendors into adapting into their products I suppose should only be a good thing for everyone else at the very least. |
| 15:28.18 | WIMPy | jpcansa: Do yu want to do anything an A or do you really only want to forward the line? And what wil be connected at B? |
| 15:28.43 | jpcansa | WIMPy, can i provide E1 instead of receive with a digium Te card? |
| 15:29.04 | WIMPy | yes |
| 15:29.04 | Qwell | jpcansa: yes |
| 15:29.33 | Diffen | bitglue yes i know that. the problem with that is that then the agent doesnt know if he is kicked out of the queue and then there will be even less calls to that agent :) |
| 15:29.37 | WIMPy | AFAIK all PRI cards can be used both directions. |
| 15:29.50 | Diffen | bitglue thanks for the information and i will look further in this matter :) |
| 15:30.05 | jpsharp | Yes. It is up to the software to provide PRI termination, not the card. |
| 15:30.07 | jpcansa | WIMPy, yes, billing |
| 15:30.19 | jpsharp | The card just digests the Layer 1 & 2. |
| 15:30.37 | danfromuk | [TK]D-Fender: It really tells something about a company if they can quickly alter their products to fit customer demand. We have a few developers whose job it is to quickly add features to our hosted pbx on demand. Polycom have recently gone down in my expectation. I asked if they could add a feature and was basically told where to go. |
| 15:30.51 | danfromuk | [TK]D-Fender: at least panasonic are in dialog with me |
| 15:31.01 | WIMPy | jpcansa: A location for billing doesn't seem to make much sense. What else is going on? |
| 15:31.22 | danfromuk | [TK]D-Fender: thats the problem with business. smaller companies are ignored, and large companies are too big to care to make a good product. |
| 15:31.39 | [TK]D-Fender | danfromuk, You have my condolences for the complications this project came bundled with... |
| 15:32.14 | danfromuk | [TK]D-Fender: polycom was a separate issue |
| 15:32.31 | jpcansa | WIMPy, the provider can only serve me the E1s at building A, then i have to re-provide to others buildings B,C and D |
| 15:32.38 | [TK]D-Fender | danfromuk, Is your prospective customer really inflexible with their specifications? |
| 15:33.13 | [TK]D-Fender | danfromuk, Have you put any effort to give them everything else and let sell them on a less troublesome solution? |
| 15:33.32 | WIMPy | jpcansa: 1:1? |
| 15:33.49 | jpcansa | WIMPy, not necesarilly |
| 15:34.06 | WIMPy | Ok, that's a statement. |
| 15:34.11 | jpcansa | jaja |
| 15:34.33 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:34.47 | jeffspeff | i'm looking for suggestions for failover methods |
| 15:34.52 | jpcansa | WIMPy, they want E1s but for sure they will not use all channels |
| 15:34.52 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 15:35.01 | WIMPy | Then just terminate them at A and generate new ones at the other locations. If you only want to do voice, you can use whatever you want between them. |
| 15:35.04 | jeffspeff | my boxes are geographically seperated |
| 15:35.20 | autofsckk | hi, i need a little help installing an openvox card, anybody who have done this? |
| 15:35.51 | WIMPy | autofsckk: What drivers do they use? |
| 15:36.04 | autofsckk | gimme a sec |
| 15:36.06 | danfromuk | [TK]D-Fender: yes, trust me, i tried to explain. the problem is, the director is my ex-boss. its a software development company, so they believe that anything is possible with a bit of programming |
| 15:37.10 | jpcansa | WIMPy, so i think the better solution cost/effective would be something like the optimux-108 instead of another asterisk ? |
| 15:37.15 | WIMPy | autofsckk: And what kind of interface? |
| 15:37.40 | WIMPy | jpcansa: With a multiplexer you can only do 1:1. |
| 15:37.50 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-133.mobile.uci.edu) |
| 15:38.01 | WIMPy | With Asterisk or LCR you can share the lines. |
| 15:38.20 | jpcansa | WIMPy, ohhhhh |
| 15:38.41 | jpcansa | WIMPy, LCR? |
| 15:38.56 | WIMPy | Linux Call Router |
| 15:38.58 | [TK]D-Fender | danfromuk, Believe? Today's lesson : Telephone is NOT "faith-based". :p |
| 15:39.07 | jpcansa | oh got it |
| 15:43.48 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 15:43.52 | plundra | Har hoppat av vid Tekniska nu. |
| 15:44.02 | plundra | Wrong channel! :-) |
| 15:44.47 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 15:44.47 | Qwell | Tekniska would be an interesting genre of music. |
| 15:45.16 | WIMPy | :-) |
| 15:46.16 | roe | Has anyone found a decent hardware vendor for rackmount, fanless servers to be used with asterisk? |
| 15:46.34 | Qwell | If it's in a rack, who cares if it has a fan? |
| 15:47.00 | roe | because I don't want to worry about it dying and overheating |
| 15:47.20 | roe | the small 1U short depth stuff with fans generally have cheapo fans |
| 15:47.23 | Qwell | This is why systems have redundant components. |
| 15:47.46 | roe | sure, and when deploying a phone system for 1000 users we're talking about a different class of machine |
| 15:47.57 | roe | I"m talking about an 8-user SIP phone with 2 analog trunks |
| 15:47.58 | Qwell | I concur. |
| 15:48.11 | roe | I don't want to spend 2K on a server |
| 15:48.23 | WIMPy | Roe: Sounds like a job for the standard plasticrouter. |
| 15:48.36 | roe | what standard plasticrouter? |
| 15:48.49 | WIMPy | Countless ones. |
| 15:48.54 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a15c6.pool.mediaWays.net) |
| 15:49.20 | WIMPy | But if you want something looking nice in a rack, I look at mini-itx.com. |
| 15:49.31 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:50.22 | jpcansa | WIMPy, so with any kind of multiplexer, fonbridge and that king of equipments i can only do 1:1 ? |
| 15:51.09 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 15:51.21 | autofsckk | WIMPy: this is the interface http://pastebin.com/XG5j52LZ |
| 15:51.35 | WIMPy | The fonebridge connects to Asterisk. |
| 15:52.00 | Qwell | autofsckk: Why don't you read the documentation they provide? |
| 15:52.09 | Qwell | oh, right. |
| 15:52.16 | WIMPy | A multiplexer doesn;t care about the contents. So unless you connect that to a physical port at A, it's will only move the interface from one location to anotehr. |
| 15:52.39 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:54.18 | WIMPy | autofsckk: That ID seems bogus. But indeed they seem to provide some good documentation. And can even be used with either dahdi or mISDN. |
| 15:54.55 | jpcansa | WIMPy, i got it, thanks a lot for your info |
| 15:56.03 | danfromuk | Is there an asterisk developers IRC channel? |
| 15:56.19 | Qwell | danfromuk: #asterisk-dev |
| 15:56.20 | WIMPy | #asterisk-dev |
| 15:56.51 | danfromuk | Thanks |
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| 16:10.53 | gusto | hey |
| 16:10.56 | gusto | what's new? |
| 16:11.44 | wdoekes | the heap allocator |
| 16:11.48 | wdoekes | bad-um-ts |
| 16:15.44 | gusto | what heap allocator? |
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| 17:02.12 | *** join/#asterisk recourse (~recourse@li339-197.members.linode.com) |
| 17:02.50 | recourse | hey all, running Asterisk 1.6.1.1 - everything working ok but just started to get these lines on the console - "TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/laxgsx1-09b090b0] |
| 17:03.10 | recourse | for every call that is placed. the calls work and all but wanted to find out whats causing this and having no luck with the google. |
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| 17:05.21 | MLNoah | is anyone available that can help me with some res_xmpp issues? |
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| 17:15.44 | MLNoah | I cannot seem to get sendtodialplan=yes to do anything with res_xmpp.so in Asterisk SVN-branch-11-r371592M. Is anyone available that could help me out? |
| 17:19.33 | jeffspeff | anybody tried clustering 2 * boxes? |
| 17:20.37 | jeffspeff | I'm trying to figure out something for high-availability and failover |
| 17:20.40 | bmoraca_work | Qwell: just fyi, an install of 10.6.1-digiumphones works with 1.2.0 first try. so it looks like something in 10.7.0-digiumphones that's causing the hang. let me know if you need any other info about my build to help isolate the issue |
| 17:22.15 | bmoraca_work | Qwell: spoke too soon..."visual voicemail" causes asterisk to crash |
| 17:26.30 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 17:32.09 | ks3 | jeffspeff: I've got an openSIPS proxy in front of a handful of Asterisk boxes |
| 17:32.43 | jeffspeff | ks3, but that allows openSIPS to be a single point of failure doesn't it? |
| 17:33.12 | ks3 | jeffspeff: I've there weren't a pair of them in an active / passive setup, yes... :) Guess I should have specified. |
| 17:33.23 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 17:33.24 | ks3 | Same with MySQL and NFS backends |
| 17:34.09 | jeffspeff | why use openSIPS instead of Asterisk for the registration proxy? |
| 17:35.36 | leifmadsen | because OpenSIPS will scale much better for number of registrations |
| 17:35.59 | leifmadsen | it can also be powerful in distributing the calls amongst the asterisk boxes |
| 17:36.40 | ks3 | You can also have multiple registrations per user with it, which is occasionally helpful. |
| 17:36.46 | jeffspeff | leifmadsen, is there an updated version of http://leifmadsen.com/sites/default/files/Why_Cluster_An_Introduction_to_Asterisk_Clustering_and_Database_Integration_AstriCon_2008_LMadsen.pdf ? |
| 17:37.53 | leifmadsen | latest version is in ATDG |
| 17:39.13 | pabelanger | Proxies are fun |
| 17:39.50 | rbowles | Kamailio anyone ? :-) |
| 17:39.51 | jeffspeff | so is that type of active/passive setup appropriate for geographically seperated failover servers? |
| 17:42.00 | Katty | hi. |
| 17:46.01 | wonderworld | evening |
| 17:49.14 | pabelanger | How do you say kamailio btw? |
| 17:49.28 | pabelanger | ka-mail-ee-oh? |
| 17:51.13 | Katty | so what's a popular sip winders phone these days |
| 17:51.47 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-121-214-153-226.lnse3.lon.bigpond.net.au) |
| 17:56.04 | jeffspeff | Katty, x-lite |
| 17:56.24 | jeffspeff | it's recently become horrible bloat-ware, but it's the most reliable i've found |
| 17:56.32 | navaismo | O_o |
| 17:56.34 | jeffspeff | others have severe issues with dtmf and such |
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| 18:00.27 | navaismo | blink blink zoiper |
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| 18:07.35 | Katty | i've used xlite and zoiper both. |
| 18:07.43 | Katty | just been awhile |
| 18:08.59 | navaismo | use linphone |
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| 18:11.10 | MLNoah | has anyone successfully set up asterisk so that they can send unsolicited XMPP to asterisk and have the server do something and send an XMPP response back? |
| 18:13.40 | leifmadsen | MLNoah: Asterisk 11 supposedly does that now |
| 18:13.48 | leifmadsen | out of band messaging |
| 18:13.59 | leifmadsen | I have not yet documented it for ATDG 4e, but it's on the list of things to do |
| 18:15.03 | MLNoah | I can't get it to work. |
| 18:15.19 | MLNoah | i have sendtodialplan=yes in jabber.conf, and context=xmpp-incoming |
| 18:15.40 | MLNoah | i see the message hit xmpp-incoming,s,1 but then it doesn't continue executing after there |
| 18:17.06 | leifmadsen | MLNoah: us xmpp.conf and res_xmpp on Asterisk 11 |
| 18:17.35 | leifmadsen | at least for chan_motif that's what you're suppose to use -- and I think xmpp.conf is supposed to essentially match jabber.conf |
| 18:18.11 | MLNoah | didn't see the xmpp.conf file in the /samples build, only jabber.conf -- but I guess it is there. maybe the jabber.conf should be taken out of the samples for 11? |
| 18:18.39 | leifmadsen | MLNoah: it wouldn't be taken out even if it was deprecated, which I've provided a patch to do |
| 18:18.40 | leifmadsen | but it's not in yet |
| 18:20.06 | MLNoah | I see. even with the config file renamed to xmpp.conf, same behavior. Debug 5 shows: |
| 18:20.27 | MLNoah | [Aug 31 14:19:14] DEBUG[6964]: res_xmpp.c:2988 xmpp_pak_message: XMPP client 'testaccount' received a message |
| 18:20.27 | MLNoah | [Aug 31 14:19:14] DEBUG[6964]: res_xmpp.c:3029 xmpp_pak_message: Deleted 0 messages for client testaccount from JID nengelberth@jabber.metalink.net |
| 18:20.27 | MLNoah | [Aug 31 14:19:14] DEBUG[6954][C-00000000]: pbx.c:4410 pbx_extension_helper: Launching 'Verbose' |
| 18:20.27 | MLNoah | [Aug 31 14:19:14] DEBUG[6964]: res_xmpp.c:3494 xmpp_client_receive: XML parsing successful |
| 18:20.27 | MLNoah | <PROTECTED> |
| 18:20.27 | MLNoah | [Aug 31 14:19:14] DEBUG[6954][C-00000000]: pbx.c:6065 __ast_pbx_run: Extension s, priority 1 returned normally even though call was hung up |
| 18:22.10 | MLNoah | and i've verified with dialplan show xmpp-incoming that there are, in fact, more priorities after s,1 |
| 18:23.12 | *** join/#asterisk teff (~teff@client-80-1-162-104.bsh-bng-011.adsl.virginmedia.net) |
| 18:24.07 | *** join/#asterisk teff (~teff@client-80-1-162-104.bsh-bng-011.adsl.virginmedia.net) |
| 18:24.58 | teff | hi there, does anyone have any experience with configuring repro ( http://www.resiprocate.org/About_Repro ) to intergrate with asterisk please? |
| 18:29.22 | leifmadsen | MLNoah: well don't just rename it.. make sure yu have all the right settings because xmpp.conf uses res_xmpp.so, not res_jabber.so |
| 18:29.44 | leifmadsen | although output looks like you're using res_xmpp |
| 18:29.47 | leifmadsen | MLNoah: additionally.... |
| 18:29.48 | leifmadsen | ~pb |
| 18:29.49 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 18:31.17 | *** join/#asterisk gonewage (~gonewage@173-9-239-197-Illinois.hfc.comcastbusiness.net) |
| 18:31.49 | MLNoah | ah, ok, sorry for the flood. |
| 18:34.45 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 18:34.51 | MLNoah | I've been using res_xmpp.so the entire time. module load doesn't show res_jabber.so as being available. |
| 18:34.58 | leifmadsen | right |
| 18:35.32 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 18:35.39 | leifmadsen | maybe show the dialplan you're using |
| 18:36.03 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 18:37.11 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 18:37.59 | MLNoah | leif > http://pastebin.com/CpGvvXd9 |
| 18:39.17 | leifmadsen | MLNoah: Verbose() on lines 2 and 3 might been a verbose number |
| 18:39.20 | leifmadsen | Verbose(2, |
| 18:39.22 | leifmadsen | not just Verbose( |
| 18:39.39 | MLNoah | though it doesn't matter what I have for the dialplan. it will execute s,1 and only s,1 (no matter what it is). I've tried with a Goto() in and it shows the verbose console output like it's jumping, and then show it hung up on goto-extension, 0 |
| 18:39.54 | leifmadsen | maybe it's designed only to execute the first line |
| 18:40.20 | leifmadsen | GoSub() doesn't work either? |
| 18:40.29 | leifmadsen | well if Goto() doesn't, GoSub() likely won't |
| 18:41.10 | MLNoah | correct, GoSub() doesn't work either |
| 18:41.35 | MLNoah | Debug: "Extension message, priority 0 returned normally even though call was hung up" |
| 18:41.44 | leifmadsen | try answering the call |
| 18:42.29 | MLNoah | hits the answer applicatoin, then "Spawn extension (xmpp-incoming, s, 1) exited non-zero on 'Message/ast_msg_queue' |
| 18:42.53 | leifmadsen | I'm out of ideas and need to focus on this vagrant stuff, GL |
| 18:43.07 | MLNoah | ok. thanks for trying. |
| 18:44.45 | Corydon76-work | If the application at any priority returns -1, that signals hangup |
| 18:45.16 | Corydon76-work | If that's a problem, and you didn't write the application, you can surround the application call with TryExec() |
| 18:47.20 | Corydon76-work | Oddly, my reading of the Verbose() app suggests that it can never return -1. |
| 18:47.45 | *** join/#asterisk portugues (~Portugus@187.115.25.87) |
| 18:49.15 | Corydon76-work | I'd suggest a 'core set debug 3' before you invoke, and ensure that debug is going to console in logger.conf |
| 18:50.54 | MLNoah | corydon > i have debug output available from my xmpp receiving stuff -- asterisk is hanging up even on applications like Answer() or Goto(foo,1) or Gosub(bar,foo,1) |
| 18:51.26 | Corydon76-work | That tends to suggest that the problem is in the channel itself |
| 18:51.58 | MLNoah | meaning something above my paygrade? *smirk* |
| 18:52.31 | MLNoah | yeah, even wrapping Answer() or whatever in TryExec, it's still hanging up before it gets to priority 2 |
| 18:53.23 | Corydon76-work | I have little to no knowledge of how the XMPP stuff works |
| 18:53.37 | *** join/#asterisk Cadey (56ab68a5@gateway/web/freenode/ip.86.171.104.165) |
| 18:54.43 | Cadey | Hi guys - Anyone no much about B Channel Negotiation with Dhadi ? |
| 18:55.20 | Corydon76-work | Cadey: you mean libpri? |
| 18:55.30 | Cadey | yeah sorry :) |
| 18:55.46 | Corydon76-work | Just ask your question |
| 18:55.54 | Corydon76-work | ~ask |
| 18:55.54 | 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. |
| 18:57.09 | Cadey | Well I know a bit about ISDN and Q931/Q921 and I understand that the D channel can negotiate (and does) a B channel to use if the intialy requested B channel becomes unavaliable during the initial call setup |
| 18:57.31 | Cadey | but I am looking to find out more information on the messages that get sent to negotiate the B channel |
| 18:58.09 | Cadey | This is my issue :) |
| 18:58.25 | Corydon76-work | Have you considered going to ITU.int and downloading the Q.931 specification? |
| 18:58.36 | Cadey | We have 55 PBX's setup where both asterisk and the telco use Top down |
| 18:59.12 | Cadey | however, one PBX is having issues because the telco is sending calls down channels that are in use and we are sending back a call complete 90 with reason 44 channel in use |
| 18:59.40 | Cadey | I thikn the problem is on there side because the other 54 locations all work fine |
| 18:59.54 | Corydon76-work | Basically, the way it works is Asterisk says "I wanna use channel A", and the net side says "No, you're going to use channel B", and the user side either accepts the channel the net side says or it can hangup. |
| 19:00.16 | Corydon76-work | That's it in a nutshell. |
| 19:00.35 | Cadey | I understand its "best practice" to send calls bottom up and receive top down but thats more for POTS PSTN not PRI, ISDN's stack has the ability to move channels and so glare is more a less non exsistant |
| 19:01.29 | Cadey | but this telco are basicly not listening to reason and now im having to explain why its not actualy a problem for both the telco and asterisk to use top down channel hunting |
| 19:01.35 | Corydon76-work | Glare can still occur when a PRI is nearly full, and there's a competition between an incoming and outgoing call to claim the last channel |
| 19:01.57 | Cadey | yeah which in this case isnt, the PRI has never even got past 30% capacity |
| 19:02.42 | Cadey | we response with a 44 because the telco is sending calls to active channels - no other telco (including them on another site with an identicaly setup asterisk box) yet still they clame its our side which is at fault |
| 19:03.04 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 19:03.22 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-58-165-46-63.lns1.lon.bigpond.net.au) |
| 19:03.27 | Corydon76-work | Best way to demonstrate otherwise is to obtain a pri debug |
| 19:03.34 | Cadey | we have :( |
| 19:03.38 | Cadey | we even went through it |
| 19:03.56 | Cadey | this is what they did in a nut shell |
| 19:04.00 | Corydon76-work | Are any calls getting through? |
| 19:04.34 | Cadey | Requested channel a, we send 44 because its in use. They re-send on channel b but thats also in use so we send another 44 and then instead of moving down and going to channel c they move back to channel a |
| 19:04.38 | *** join/#asterisk bitglue (~pfrost@207.8.235.242) |
| 19:04.39 | Cadey | which is really quite odd |
| 19:04.45 | Cadey | they already know a is in use |
| 19:04.55 | Corydon76-work | Now would probably be a good time to involve your hardware vendor and get their support to debug the issue with the telco |
| 19:05.05 | Cadey | we did :) |
| 19:05.12 | Cadey | we had a digium tech on hand |
| 19:05.19 | Cadey | and they still wouldnt listen |
| 19:05.26 | Cadey | 5 damn hours on a call last night i spent :( |
| 19:05.47 | Corydon76-work | and you're deescalating the problem to here, because you think we're going to be able to do something that the hardware tech could not? |
| 19:06.03 | Corydon76-work | Your option right now is to get another telco, honestly. |
| 19:06.26 | Corydon76-work | Free market. If the telco won't play ball, get another. |
| 19:06.30 | Cadey | no no |
| 19:06.45 | Cadey | I was wondering if there was anyone who could help me explain the ISDN messaging because i dont claim to be an expert - I know it quite well but there is much I dont |
| 19:06.47 | *** join/#asterisk aossama (~ao@41.232.120.114) |
| 19:07.00 | Corydon76-work | That's what a Digium tech can do |
| 19:07.07 | Corydon76-work | Use your ticket. |
| 19:07.22 | Cadey | :) |
| 19:07.32 | Cadey | if you dont or cant help thats cool - I was only asking :D |
| 19:07.59 | Cadey | its good to have chats in IRC i find because others read them and also learn |
| 19:08.02 | Corydon76-work | Your best bet for figuring out a solution is the Digium support system. |
| 19:08.17 | *** join/#asterisk PhoenixMage (~Phoenix@101.160.172.248) |
| 19:08.39 | Corydon76-work | But honestly, it sounds like the telco has a dickhead who isn't willing to do anything but play the blame game |
| 19:08.54 | Cadey | yeah I know :( |
| 19:09.05 | Corydon76-work | and in that situation, the only fix is to go to a competitor |
| 19:09.39 | sruffell | Couldn't changing the channel assignment order in libpri, while not a *fix*, at least let this work a little more often? |
| 19:09.51 | sruffell | making it an option to use top down or bottom up, etc.. |
| 19:10.08 | Cadey | the fact he wouldnt even achknowledge the fact everything was fine at another location with them as the telco on an asterisk system with identical server's/pri's and termination eqiptment really shocked me |
| 19:10.25 | Cadey | sruffell : thats what we did in teh end |
| 19:10.35 | Cadey | changed outs to send bottom up |
| 19:10.49 | sruffell | ahh…so you're customers aren't complaining….you're just looking for something a little more long term? |
| 19:11.22 | Cadey | but the problem will return at high load and be even harder to debug because of how random the occurances become |
| 19:11.30 | Corydon76-work | The other thing that will help this is to use a multi-span PRI |
| 19:11.36 | Cadey | oh they are complaing because they are getting fast busy on incomming calls |
| 19:12.04 | Cadey | we have a two span PRI |
| 19:12.09 | Cadey | 47 usable |
| 19:12.10 | Cadey | 2 D chan |
| 19:12.13 | Cadey | all in G1 |
| 19:12.16 | Cadey | group 1 |
| 19:12.29 | Corydon76-work | Uh, if you have 2 D channels, that should mean 46 usable |
| 19:12.46 | Qwell | Corydon76-work: NFAS |
| 19:12.51 | Cadey | 1-23, 26-47 so yes sorry :) |
| 19:12.55 | Cadey | 46 total |
| 19:12.56 | Qwell | err, nevermind. |
| 19:12.58 | Corydon76-work | You mean "g1", not "G1" |
| 19:13.04 | Corydon76-work | Qwell: yes, nfas |
| 19:13.11 | Cadey | G1 bottom up, g1 top down |
| 19:13.14 | Qwell | didn't realize he actually said 2 D |
| 19:13.15 | Cadey | so its now using G1 |
| 19:15.52 | Cadey | but anyway trying to fight my corner now because the business is kicking off the fast busy was the fact we are both using top down and its really not |
| 19:16.01 | Cadey | all the other 54 sites use the same and its fine :( |
| 19:17.09 | Corydon76-work | Cadey: I think you've got the groups backwards. Lowercase goes in ascending order, uppercase in descending order |
| 19:17.28 | Cadey | G is bottom up which is 47 back ? |
| 19:17.32 | Cadey | as in bottom channel |
| 19:17.34 | Cadey | ? |
| 19:17.53 | Corydon76-work | G starts with the last channel in the group, g starts with the first channel in the group |
| 19:18.01 | Corydon76-work | Numbering as we do in Asterisk |
| 19:18.44 | Cadey | we mean the same things, my termanoligy may be off :) |
| 19:18.47 | Corydon76-work | Most people would consider 1 to be the bottom and 47 to be the top |
| 19:18.57 | Cadey | :) |
| 19:19.12 | leifmadsen | would |
| 19:19.13 | Cadey | but "decending" suggests bottom as much as "1" in a number range :D |
| 19:19.27 | leifmadsen | decending from the top to the bottom.... |
| 19:19.34 | leifmadsen | top being high number |
| 19:19.37 | Corydon76-work | In descending order, 1 would be the last channel allocated |
| 19:19.54 | Corydon76-work | Assuming G1 and not R1. |
| 19:20.08 | Cadey | yeah so in decending the "bottom" is the first one going back up the range because the channel number decends each time |
| 19:20.22 | Cadey | but anyone :D |
| 19:20.29 | Cadey | I dont really care about that part :D |
| 19:20.30 | Corydon76-work | I thought you said the telco was going top down |
| 19:20.44 | Cadey | yeah top as in 1 to 47 |
| 19:21.01 | Corydon76-work | Top down for most of us would be from 47 to 1 |
| 19:21.13 | Cadey | :) |
| 19:21.19 | Cadey | can we move off this one heheh |
| 19:21.41 | Corydon76-work | If you're confused as to what it means, it may be the reason why you're getting glare. |
| 19:21.57 | Cadey | arg, semantics |
| 19:22.07 | Cadey | I know that decending and G is |
| 19:22.09 | Corydon76-work | And the telco guy may not be a dick, he may just think you're nuts |
| 19:22.23 | Cadey | possibly :) |
| 19:22.35 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:22.48 | Corydon76-work | So yeah, could very well be relevant |
| 19:24.08 | Cadey | only if youre really anal and get stuck on one word in an entier explination as to what is going on which includes the numbers to ratify what "top down" was supposed to mean |
| 19:24.51 | Cadey | which tbh, techies do a lot :) |
| 19:30.00 | *** join/#asterisk jsjc (~Adium@41.146.78.188.dynamic.jazztel.es) |
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| 20:05.43 | *** join/#asterisk lorsungcu (~lorsungcu@65.103.31.36) |
| 20:05.52 | lorsungcu | hi |
| 20:06.08 | lorsungcu | i've got an issue with odbc, i think |
| 20:06.24 | lorsungcu | asterisk hangs for a few seconds whenever it accesses it |
| 20:06.29 | lorsungcu | including reload |
| 20:19.51 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 20:23.25 | *** join/#asterisk Jeff_warren (~Jeff_warr@d24-204-241-182.home4.cgocable.net) |
| 20:23.28 | *** join/#asterisk ks3 (ks3@2600:3c03::f03c:91ff:feae:97e3) |
| 20:23.55 | Jeff_warren | I am having an issue with the directory not matching by last name |
| 20:25.06 | Jeff_warren | first name is fine |
| 20:25.31 | adeel|work | is there a way to insert the a-leg call-id as an X- header into one of the SIP packets on the b leg? |
| 20:29.20 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 20:31.14 | bitglue | so, I can, by setting REDIRECTING, cause Asterisk to emit a Diversion: header in SIP invites. And I can pay my SIP trunk provider to allow that Diversion header, and do something with it. What exactly do they do with it? How would a user on a cell phone or personal landline use this information? |
| 20:32.41 | Jeff_warren | I installed AseterisNow 2.0.2 |
| 20:35.52 | lorsungcu | bitglue, whatre you trying to do? |
| 20:36.50 | bitglue | lorsungcu: well, i'm trying to make it so that if someone has their handset forwarded to an external number (probably their cell), they can know who's calling, and in the case of a queue call, know that it's from a queue, and which one. |
| 20:37.21 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ikhdcynmelolipht) |
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| 20:38.15 | lorsungcu | ah yeah' |
| 20:38.24 | bitglue | i think my carrier will let me set callerid to anything I want, so I guess i can Set(CALLERID... and have the number displayed on the cell phone be that of the caller, not the office number. But, that doesn't tell them that the call is from a queue. I could play a prompt when the cell phone is answered, but I was hoping for a more elegant solution. |
| 20:38.49 | lorsungcu | most cell carriers wont pass caller id name |
| 20:38.52 | lorsungcu | reliably anyway |
| 20:38.57 | bitglue | yeah, so i'm learning. |
| 20:38.58 | lorsungcu | so what were you thinking? |
| 20:38.59 | bitglue | pretty lame. |
| 20:39.07 | lorsungcu | also |
| 20:39.19 | lorsungcu | i've found a prompt in situations like that actually works pretty well |
| 20:39.37 | bitglue | but i'm also wondering what this $100 feature that allows the Diversion header gets me |
| 20:39.41 | *** part/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 20:39.50 | bitglue | it must do something neat enough to make it worth $100, right? |
| 20:39.58 | lorsungcu | mind if i ask what carrier? |
| 20:40.01 | bitglue | Windstream |
| 20:40.06 | lorsungcu | hmm |
| 20:40.18 | lorsungcu | haven't used their SIP |
| 20:41.11 | bitglue | i guess ISDN has a way to signal that a call has been redirected, so i guess if i were placing calls over ISDN, i could access that information? |
| 20:41.26 | bitglue | but that's probably not of much use to any consumer telephone |
| 20:41.27 | lorsungcu | i know with some carriers, they expect that the diversion header included in order to change CID to anything not on the trunk group. |
| 20:41.40 | lorsungcu | but i dont think you should/need to in your case |
| 20:41.54 | bitglue | yeah, see I was under the impression they would want that, too, and that's why I had to pay $100. |
| 20:42.02 | Katty | battens down the hatches |
| 20:42.35 | lorsungcu | Katty: MN is ~80, breezy, and partly cloudy |
| 20:42.45 | lorsungcu | just fyi |
| 20:42.47 | lorsungcu | :D |
| 20:42.52 | Katty | youuu suck. |
| 20:42.57 | Katty | i have a lovely red BLOB heading this way |
| 20:43.02 | bitglue | but the way i read http://tools.ietf.org/id/draft-levy-sip-diversion-08.txt, Diversion: specifies the original destination of the call, before it was forwarded |
| 20:43.05 | Katty | and tornado warnings! weee |
| 20:43.17 | lorsungcu | http://www.accuweather.com/en/us/minnesota/weather-radar |
| 20:43.20 | lorsungcu | love it. |
| 20:43.20 | *** join/#asterisk bchia (~Adium@nat/digium/x-tatfuxqkryejenrv) |
| 20:43.33 | bitglue | so really, nothing to do with caller id. |
| 20:44.27 | bitglue | maybe i can try calling them and see if they understand their own features. probably not, though. |
| 20:44.41 | lorsungcu | eh, they've been good to me |
| 20:44.54 | lorsungcu | i'd drop that idea though |
| 20:45.01 | lorsungcu | as you really wont get anywhere |
| 20:45.22 | lorsungcu | since cell phones CID limitations are the problem here |
| 20:45.49 | bitglue | so, you'd set the caller number to the number of the original caller, and play a prompt to indicate the queue? |
| 20:45.54 | lorsungcu | right |
| 20:46.05 | lorsungcu | although i've had customers ask that we use the number of the office |
| 20:46.22 | lorsungcu | so they can use that number as part of some free calling group |
| 20:46.25 | lorsungcu | or something |
| 20:46.42 | bitglue | hmm |
| 20:46.50 | lorsungcu | and really, when you're on your cell, who cares what the incoming CID is, unless these are sales agents that have repeated interactions with the same peolle |
| 20:47.01 | lorsungcu | or users in their houses |
| 20:47.16 | lorsungcu | or remote office |
| 20:47.46 | bitglue | well, in my case, the "agents" are equally likely to receive calls directed to them as from a queue. You wouldn't really want to take a technical support call if you are driving, but if someone is trying to reach you directly, you might. |
| 20:48.12 | lorsungcu | so these are technicians? |
| 20:48.22 | bitglue | i should say "at a friend's house", because talking while driving is bad. |
| 20:48.26 | bitglue | some of them, yes. |
| 20:48.35 | bitglue | we have all kinds :) |
| 20:48.39 | lorsungcu | probably dont need caller id, and it would make more sense to just use the office CID |
| 20:48.46 | lorsungcu | when calling from the queue |
| 20:49.26 | lorsungcu | if you were to do that, you wouldnt _really_ need a prompt, although i guess it wouldnt hurt |
| 20:49.40 | bitglue | i have a prompt anyway, so cell phone voicemails don't suck calls out of the queue |
| 20:49.59 | lorsungcu | i see |
| 20:50.00 | bitglue | but currently it doesn't contain anything but "you have a call, press 1 to accept" |
| 20:50.15 | lorsungcu | what does ti say? |
| 20:50.46 | bitglue | Background(followme/no-recording); Background(followme/options) |
| 20:51.40 | Katty | Qwell: you lucky ducky.... |
| 20:52.01 | Katty | Qwell: huntsville is right between the two major lines of storms |
| 20:52.11 | lorsungcu | bitglue: so what are you waiting for them to input |
| 20:52.33 | bitglue | lorsungcu: 1 (accept) or 2 (reject) |
| 20:53.22 | lorsungcu | so you are asking them to press 1 |
| 20:53.27 | bitglue | yes |
| 20:54.04 | lorsungcu | but that isn't sufficient |
| 20:54.06 | lorsungcu | ? |
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| 20:54.28 | bitglue | well, i'd just have to modify it to say "you have a call from SOMEQUEUE. Press 1 to accept." |
| 20:54.51 | lorsungcu | sure |
| 20:55.06 | bitglue | just a bit of a bummer that they have to answer, then listen before they know the queue. But maybe no way around that, given this advanced cell phone technology. |
| 20:55.15 | lorsungcu | rather than all that |
| 20:55.26 | lorsungcu | maybe a recognizeable sound as soon as the call is answered |
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| 20:56.37 | bitglue | really it's a bit stupid that an iPhone has fewer features than a cheap SIP handset in this respect. |
| 20:57.08 | lorsungcu | do not like the iphone. |
| 20:57.48 | lorsungcu | you could build an app that asterisk would alert when a call was being sent to that phone |
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| 22:05.10 | lorsungcu | so swift() just stopped recognizing digits while it was speaking |
| 22:05.22 | lorsungcu | just skips straight to the next priority |
| 22:05.46 | lorsungcu | definitely sees that I pressed something |
| 22:05.48 | lorsungcu | then just moves on |
| 22:10.14 | lorsungcu | its so terrible... |
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| 22:13.02 | flujan | hello guys, this have been merged into 1.8 branch? |
| 22:13.03 | flujan | https://reviewboard.asterisk.org/r/760/diff/#index_header |
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| 22:24.46 | lorsungcu | wtf and now it's back to working |
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