| 00:01.21 | citywok | What do people use for solutions for their receptionists? I just finished my reception console application to replace sidecars: http://www.lighttouchsoft.com/ |
| 00:03.10 | novceguru | WIMPy: yeah, I'm wondering if I would be best going like, FXO --> gate --> FXS --> TCP/IP --> FXS --> FXO into the controller unit |
| 00:03.15 | citywok | Also, if anybody has any feedback i'd be happy to hear it |
| 00:04.48 | novceguru | you blast me about asking an obscure question then plug your own software that's specific to a KX-UT670? :) |
| 00:05.07 | easymac | what a faggot, amirite? |
| 00:05.29 | citywok | lol, i didn't blast you, i actually answered your question ;) |
| 00:05.44 | easymac | Nah, you were just an internet nerd virgin about it. |
| 00:05.54 | easymac | Your faggoty smiles don't save your retarded ass, either. |
| 00:06.22 | citywok | easymac: you may want to scroll back and re-read the conversation. your hatred is a bit misplaced. |
| 00:08.09 | *** part/#asterisk easymac (uminac@pdpc/supporter/student/easymac) |
| 00:08.31 | citywok | that's one way to apologize. thanks easymac! |
| 00:18.51 | greenwolf | wow the #ubuntu channel has the most retarded channel ops |
| 00:19.06 | greenwolf | i was banned for saying "shit' one time no warning |
| 00:19.07 | greenwolf | crazy |
| 00:19.15 | greenwolf | sorry i just had to bring that up :) |
| 00:24.11 | citywok | lol |
| 00:24.46 | greenwolf | its like i mean seriously |
| 00:24.52 | greenwolf | really you got nothing better to do |
| 00:24.53 | WIMPy | We call our admins shit. |
| 00:24.58 | greenwolf | lol |
| 00:25.02 | WIMPy | Well, actually SH:IT. |
| 00:25.08 | greenwolf | but i was asking a question about wireless cards |
| 00:25.15 | greenwolf | and i said this shit is giving me a hard time |
| 00:25.27 | greenwolf | next thing i know im kicked/banned and got a private msg from the operator |
| 00:25.39 | greenwolf | he was like no swearing me take this very seriously |
| 00:25.58 | greenwolf | bro he wont even let me back in..i told him to grow the fuck up man |
| 00:26.18 | greenwolf | ive never seen such immature arrogance in my entire life...it truly amazes me |
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| 00:51.49 | [TK]D-Fender | "i told him to grow the fuck up man" = clearly the thing to say to the person who just banned you for swearing. |
| 00:57.18 | citywok | lol, yea, not a great way to go about getting unbanned |
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| 01:15.05 | phix | hey gang! |
| 01:15.17 | citywok | hey |
| 01:16.03 | phix | oh nice, Ill take a citybeef and a citychicken thnx! and some airplane lessons! |
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| 04:17.07 | LiuYan | I use SIP realtime with SQL Server 2000 database server, when SIP client un-registered, asterisk server will crash with segfault error. If I changed database server to PostgreSQL, then all works fine. |
| 04:18.33 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 04:20.02 | citywok | LiuYan: what version of Asterisk? |
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| 04:26.09 | ChannelZ | I would say a valuable lesson has been learned |
| 04:30.31 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 04:33.27 | LiuYan | citywok: 1.8.12, I tried both x86 and x64 version, |
| 04:36.17 | citywok | ChannelZ: what's that? don't use sql2000? |
| 04:36.26 | citywok | LiuYan: does it happen with newer versions of SQL? |
| 04:38.22 | LiuYan | I haven't try yet, I will install a SQL Server 2008 Express to see what will happened |
| 04:55.32 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:03.09 | LiuYan | citywok: I just tried on SQL Server 2008 R2 Express SP1, but no luck, i got same result. |
| 05:03.36 | LiuYan | error message: *** glibc detected *** asterisk: malloc(): smallbin double linked list corrupted: 0x0000000006612510 *** |
| 05:04.39 | LiuYan | with some backtrace and memory map information followed |
| 05:07.32 | LiuYan | pastebin: http://pastebin.com/a6zahUzD |
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| 05:21.30 | citywok | LiuYan: I didn't look at it, but it sounds like you may have found yourself a bug worth posting on the tracker |
| 05:22.32 | citywok | Make sure to follow the bug posting guidelines and get a coredump/backtrace it with gdb |
| 05:23.59 | LiuYan | ok, thank you |
| 05:50.55 | *** part/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
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| 06:36.18 | schmidts | good morning |
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| 07:06.55 | oej | Morning folks |
| 07:07.34 | citywok | morning |
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| 07:40.05 | gavimobile1 | im a little lost. the asterisk definitive guide is talking about using GoSub and Macro() in scenarios where they both can be used to do the same thing. which one should I use? |
| 07:40.42 | wdoekes2 | gosub |
| 07:40.46 | wdoekes2 | macro is obsolete |
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| 07:42.02 | gavimobile1 | wdoekes2: since when? |
| 07:42.14 | wdoekes2 | long |
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| 07:42.19 | gavimobile1 | is one more flexable than the other? |
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| 08:57.27 | arekm | hi, is there a way to load balance between channels? I have for example SIP/a, SIP/b, SIP/c, SIP/d and want outgoing calls to go via these with load balacing, so the channels get +- equal number of calls? |
| 08:58.13 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 08:58.14 | arekm | putting these in the same group= seem to cause calls to go via first available only |
| 09:00.25 | arekm | hm, round-robin |
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| 09:00.52 | DagF | Hello |
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| 09:33.00 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 09:34.14 | as001 | Hello I need to get information about sip channels. I tried with sip show channel and it gives me some call-id but with that call-id I can not use sip show channel call-id because call-id is much greater than sip show channels give. How can I get list of call-ids which i can use in sip show channel command ? |
| 09:35.10 | as001 | sip show channels gives just 15 characters of callid |
| 09:37.13 | kaldemar | as001: from "core show channel <channel>" |
| 09:44.13 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 09:44.51 | wdoekes2 | as001: tab completion or the concise option |
| 09:45.27 | wdoekes2 | oh wait, there is no concise for 'sip show' |
| 09:45.38 | as001 | yes I know I can do that with tab in but I want to parse output in script. |
| 09:45.45 | as001 | *in cli |
| 09:45.54 | wdoekes2 | then using AMI is recommended |
| 09:46.01 | wdoekes2 | CLI output can change at any time |
| 09:46.09 | as001 | I need full callid i can use in sip show channel command |
| 09:46.46 | as001 | hm.ok |
| 09:47.07 | wdoekes2 | (but you could take the SIPCALLID= from the core show channel XYZ |
| 09:49.57 | as001 | core show channels gives me: SIP/TelekomSIP-00001 and i need SIP/TelekomSIP-00001 in core show channel |
| 09:50.19 | as001 | sorry i need SIP/TelekomSIP-000018d6 in core show channel |
| 09:50.34 | kaldemar | core show channels concise |
| 09:50.57 | as001 | yes thats it thanks :) |
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| 11:11.35 | qakhan | can any one tell me how to install pkg-config on centOS |
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| 12:35.30 | Greenlight | Afternoon folks. Quick question that I'm hoping is simple but can't seem to find an answer on line. My SIP provider doesn't authenticate incomming calls, all I know is that they'll come from one of a few hundred IP's. How can I configure my trunk in sip.conf to cover this? |
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| 12:36.21 | bulkorok | trace |
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| 12:47.17 | Greenlight | Does no one else have a situatuion like this for incomming calls, there must be a solution ? |
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| 12:51.51 | WIMPy | Greenlight: I had to create 36 peers only. |
| 12:52.03 | WIMPy | But that's unfortunatly the way it is. |
| 12:52.48 | Greenlight | Damn, so seperate peers for each IP, can't even have multiple host= lines for the one peer? |
| 12:53.32 | WIMPy | nope |
| 12:53.48 | Greenlight | Oh well time to write a script. Thanks anyways :) |
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| 12:54.11 | zamba | how can i get rid of the "doing dnsmgr_lookup for" messages? |
| 12:54.35 | WIMPy | The worse thing is that you have to regularly check th elist of IPs or the calls won't match if your ITSP changes something. |
| 12:55.10 | Greenlight | I really hope they dont change stuff too often |
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| 13:38.30 | pokus | hi is possible set for asterisk different host ip for peer? i want two group users. one is on private netork and one is on public network. when peer from private make call i want create connection: asterisk_ip1 -> peer1_ip when public, it will be: asterisk_ip2 -> peer2_ip |
| 13:38.34 | pokus | its possible? |
| 13:40.35 | pokus | or install 2 asterisk? one for public one for private network? |
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| 14:03.20 | treborsux | I have a Openvox A1200P Ver 4. I put it in my system and I cant make calls out. It will recieve a call but only on channel 1 and it is scratchy and over driven. Then if another call comes in it rings but there is no sound. Any attempt to call out gives error about number. So I bought another card from another supplier same thing. My original card in the machine another a1200p works just fine. Could there be |
| 14:03.43 | treborsux | There is a card in there working now. But I have 2 other cards both doiung the same thing wrong. |
| 14:03.59 | treborsux | COuld driver be dirrent for newer version? |
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| 14:15.34 | treborsux | Nevermind all 3 are ver 4 |
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| 14:15.53 | treborsux | I tried to put this card in as second 12 card same machine |
| 14:16.07 | treborsux | then tried to run it alone in place of the one that works. |
| 14:16.22 | treborsux | how could both cards be broke the same exact way? |
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| 14:23.05 | treborsux | what file tells polycom to use light for voicemail indicator or not? |
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| 14:23.16 | treborsux | .join #openvox |
| 14:23.55 | treborsux | IS there anyone here i have a strange issue with a couple of dahdi cards |
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| 14:24.51 | treborsux | Anyone please? |
| 14:24.58 | treborsux | .join #freepbx |
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| 14:25.12 | riddlebox | hey all, is there a point where asterisk stops scaling? like how many registrations can it handle on one server? Or is it basically everything depends on the specs of the server? {hopefully that makes sense} |
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| 14:34.25 | schmidts | riddlebox when you use asterisk 10 this limit is much higher than you should handle on one server but it sure depends on the specs of a server |
| 14:36.17 | newtonr | riddlebox: http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations has some good info |
| 14:38.37 | riddlebox | thanks, I am just looking into it, for grins. I have heard about using opensips+asterisk for real big installations but cant really find anything talking about why you need to or when you need to. |
| 14:41.02 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 14:44.17 | schmidts | riddlebox the problem you will have with a very large installation. if this one server dies you will loose a lot of concurrent calls and a lot of users will be offline until its rebootet |
| 14:44.37 | schmidts | with a sip proxy in front you can load balance between two or more servers and stay online all the time |
| 14:45.32 | *** join/#asterisk The_Ball (~The_Ball@122.150.108.38) |
| 14:46.19 | schmidts | it really depends on your hardware and what you want to do with your asterisk. An ivr service for very short calls, like 10 seconds can hold up to several thousand concurrent calls without any problems but if this is your only server without any fallback, what happens if this server dies? |
| 14:49.59 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 14:50.13 | Dovid | does anyone have a number in the US with early meadia that i can test against? |
| 14:51.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 14:55.15 | angler | Dovid, I believe the main number for UPS does early media until you press a DTMF |
| 14:58.00 | Dovid | angler: Do you know the numner? |
| 14:58.30 | Dovid | 800-pick-ups answers right away |
| 15:00.30 | Dovid | ah here we go: ...800-742-5727 |
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| 15:19.20 | ZenMaster | Morning gentlemen. |
| 15:19.26 | ZenMaster | Three pay checks this week. |
| 15:19.57 | ZenMaster | I think it will be a good week. |
| 15:20.11 | ZenMaster | Hopefully these are not the famous last words. |
| 15:20.59 | Qwell | ZenMaster: Did you bring enough to share with the whole class? |
| 15:23.08 | ZenMaster | Yes. |
| 15:23.17 | ZenMaster | Give me your bank numbers and I will. |
| 15:23.30 | ZenMaster | :X |
| 15:24.38 | [TK]D-Fender | OpenVox : When failure IS an option. |
| 15:26.14 | ChannelZ | "Dear Email Owner, This E-mail is to officially inform you that your E-mail Address have been verified and pronounced as the lucky Winner Of £1,000,000.00 GBP, in the 2012" |
| 15:26.18 | ChannelZ | Seems legit. |
| 15:26.29 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-tbmmhmkrcmsralqt) |
| 15:27.11 | ZenMaster | I am an African princess. However our banks are not working. I just inherited 10,000,000,000, African clams. I need your account and I will leave you 10%. |
| 15:28.42 | Dovid | lol |
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| 15:50.35 | bn-7bc | ChannelZ: Did the mail actually say "Dear E-mail owner"? damed spammers are lazy, note even the will to do a regex to insert yore name lol |
| 15:50.56 | bn-7bc | s/note/not |
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| 15:55.02 | Greenlight | There must be a more elegant and manageble way to handle incomming calls than over 150 seperate peers; one for each IP address.... |
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| 16:01.30 | pabelanger | Greenlight: templates |
| 16:02.15 | Greenlight | Explain |
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| 16:09.39 | p3nguin | greenlight: Are you talking about a single provider with 150 addresses? |
| 16:17.34 | Greenlight | Yea |
| 16:18.10 | Greenlight | Incomming calls are unauthenticated, but from one IP in their ranges |
| 16:19.08 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:19.27 | Greenlight | I'm now thinking AGI scripts to compare the IP's or a virtual server to trunk the calls as IAX, but seems a bit of a hack |
| 16:20.14 | Greenlight | I would have thought providers giving for example /26 ranges is quite common |
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| 16:24.52 | p3nguin | I don't know why ITSPs need to do that crap. One IP address should be enough. |
| 16:27.39 | Greenlight | You'd have thought yes - seperate media/rtp servers I understand, but one signalling server and maybe a backup would surely suffice /sigh |
| 16:30.40 | Greenlight | Laters folks |
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| 17:09.38 | *** join/#asterisk fubada (~fubada@ool-4573155d.dyn.optonline.net) |
| 17:09.41 | fubada | hi |
| 17:09.56 | fubada | [TK]D-Fender: i took your advice, ditched gui, read the asterisk wiki |
| 17:10.19 | fubada | i have some issues with incoming calls, may i provide ya with a dialplan and log? |
| 17:12.56 | talntid | fubada, pastebin it all, and any of us can take a look and help ya :) |
| 17:13.00 | fubada | ok |
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| 17:13.10 | fubada | its super simple, im using Goto and Im losing the argument |
| 17:14.12 | fubada | http://pastie.org/3945642 |
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| 17:14.42 | fubada | talntid: two questions about the above, how do i properly pass the extension on line 4 |
| 17:14.58 | fubada | and how do i implement the invalid handler on lines 6-9 |
| 17:15.07 | talntid | the extension that was dialed is what you want? |
| 17:15.11 | talntid | ${EXTEN} |
| 17:15.34 | fubada | yes, but my local extensions are not my DID, so Im doing ${EXTEN,-4} |
| 17:15.44 | talntid | gotcha |
| 17:16.08 | fubada | its not ringing my extensions, and instead does |
| 17:16.08 | fubada | <PROTECTED> |
| 17:16.26 | fubada | should be Goto (LocalSets,1417) |
| 17:16.59 | fubada | dialplan log http://pastie.org/private/aoiackcogfkouyklbk9ww |
| 17:17.10 | talntid | right after you do the Set, on line 3.... make line 4 do a NoOP or a Verbose, and display the FilteredExtenstion |
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| 17:18.23 | fubada | talntid: im not sure of how the verbose syntax looks |
| 17:18.37 | talntid | sure you do, you use it on line 2 |
| 17:18.44 | fubada | right |
| 17:18.44 | fubada | oops |
| 17:18.48 | talntid | ;] |
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| 17:19.53 | fubada | <PROTECTED> |
| 17:19.56 | fubada | blank |
| 17:20.21 | p3nguin | I feel like I already went over this with you. |
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| 17:20.50 | fubada | p3nguin: ive ditched the gui and im building it all by hand from scratch |
| 17:21.40 | p3nguin | Good. Now review how I instructed you to make this happen. |
| 17:21.48 | fubada | so my apologies if its something that we've already covered, im trying to fix that |
| 17:21.56 | p3nguin | Unless it wasn't you, in which case I will tell you. |
| 17:22.01 | fubada | i believe you suggested i use ${EXTEN,-4} |
| 17:22.10 | p3nguin | No. |
| 17:22.14 | p3nguin | That is invalid. |
| 17:22.26 | p3nguin | ${EXTEN:-4} |
| 17:22.33 | fubada | right right :) |
| 17:22.54 | p3nguin | That is equal to the last four characters of the extension used to get there. |
| 17:24.07 | p3nguin | I see your call isn't matching any peer entry or you've misconfigured the peer entry. |
| 17:24.23 | p3nguin | It ended up in the unauthorized calls, where it clearly doesn't belong. |
| 17:24.51 | fubada | the context for my sip prvider is set to unauthenticated |
| 17:25.14 | p3nguin | Why? Why would a known provider be unauthenticated? |
| 17:26.03 | p3nguin | Try not to create headaches. They will come naturally with time. |
| 17:26.57 | fubada | im not really sure, i followed the docs in section http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ICR-SECT-1.html#InternetCallRouting_id289964 |
| 17:27.30 | p3nguin | partially |
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| 17:28.01 | fubada | I changed the context to 'incoming' for my sip provider |
| 17:28.09 | p3nguin | That says to me that you should set the GENERAL context to unauthenticated. |
| 17:28.15 | fubada | but im not sure where that needs to be included |
| 17:28.27 | fubada | LocalSets needs to include incoming? |
| 17:28.30 | p3nguin | But when you create an actual peer for a known host, you should give it an appropriate context. |
| 17:29.05 | p3nguin | No. Not unless phones in the LocalSets context need to be able to dial extensions within the incoming context. Usually this isn't the case. |
| 17:29.25 | *** join/#asterisk navaismo (~navaismo@201.123.94.189) |
| 17:29.46 | p3nguin | <fubada> I changed the context to 'incoming' for my sip provider <----- Good. Now create extensions inside that context which will match calls from the provider. |
| 17:30.48 | fubada | i simply renamed [unathenticated] to [incoming] in extensions.conf |
| 17:30.48 | p3nguin | Bad. |
| 17:30.48 | fubada | heh |
| 17:30.48 | p3nguin | Now you need to make another unauthenticated context. |
| 17:30.51 | fubada | i dont want to maintain LocalSets and PublicExtensions |
| 17:30.56 | p3nguin | Since that is a used context, it needs to exist. |
| 17:31.22 | p3nguin | What kinds of extensions can be found in the LocalSets context? |
| 17:31.23 | fubada | http://pastie.org/3945710 |
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| 17:31.30 | p3nguin | What kinds of extensions can be found in the PublicExtensions context? |
| 17:31.51 | fubada | LocalSets has desk extensions, 1417, 1426...its the last 4 digits of my public DID |
| 17:32.09 | fubada | so you suggest my full DID is in PublicExtensions? |
| 17:32.21 | p3nguin | Why would the incoming context print UNAUTHENTICATED REQUEST for calls hitting it? |
| 17:32.40 | fubada | taking that out |
| 17:32.41 | p3nguin | You aren't even thinking at all when you're doing this stuff. |
| 17:32.52 | fubada | i was just following the guide heh |
| 17:32.57 | p3nguin | Not really. |
| 17:33.27 | p3nguin | Create a context for unauthenticed calls. Set it for the general context. |
| 17:33.40 | p3nguin | s/unauthenticed/unauthenticated/ |
| 17:34.04 | fubada | im not sure what to populate in this context |
| 17:34.21 | p3nguin | Create an appropriate context for calls from providers. Set it for the ITSP peers' context(s). |
| 17:34.35 | fubada | i only hvae one sip provider, so technically there are no unauthenticated requests |
| 17:34.48 | p3nguin | The unauth context can have zero or more extensions which will match calls from people you do not know and do not have peer entries for. |
| 17:34.50 | fubada | my sip port is open exposed to my only sip provider |
| 17:35.10 | fubada | s/open/only |
| 17:35.13 | p3nguin | Leave it empty for now. |
| 17:35.40 | fubada | ok, thank you |
| 17:35.41 | p3nguin | In your context used for calls from the provider, configure real extensions, either explicit or patterns. |
| 17:36.00 | p3nguin | For only NANP, use _NXXNXXXXXX |
| 17:36.28 | p3nguin | For improper NANP with a country code prefixed, use _1NXXNXXXXXX |
| 17:36.45 | p3nguin | Do you have other numbers that your provider will call on your box? |
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| 17:37.09 | fubada | no, i believe its always _1NXXNXXXXXX |
| 17:37.38 | p3nguin | Then you can create one extension with that pattern and it will match all calls. |
| 17:37.51 | fubada | http://pastie.org/3945754 |
| 17:37.54 | fubada | something like this? |
| 17:38.04 | p3nguin | To parse out the last four digits, use ${EXTEN:-4} |
| 17:38.25 | fubada | no reason to filter i guess |
| 17:39.03 | p3nguin | There is no reason to filter it, but it would still get what you wanted. |
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| 17:39.42 | p3nguin | You usually only need to filter things with a pattern which accepts things you didn't control. |
| 17:39.54 | p3nguin | Since you are using an appropriate pattern, you controlled it already. |
| 17:40.05 | fubada | got it |
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| 17:40.51 | p3nguin | You could skip lines 3 and 4 and just use same => n,Goto(LocalSets,${EXTEN:-4},1) |
| 17:40.59 | fubada | right thats how i have it |
| 17:41.34 | p3nguin | So two priorites: one to show the call hit the extension and one to send it to another extension in another place. |
| 17:42.03 | fubada | right |
| 17:42.12 | p3nguin | That should be enough to get a call from the provider and into an extension in another context. |
| 17:42.18 | fubada | working as expected |
| 17:42.23 | p3nguin | What happens at THAT extension I have not yet seen. |
| 17:42.48 | fubada | it works http://pastie.org/private/zaju2dqdr53buvvm7qcna |
| 17:42.52 | fubada | like that |
| 17:43.09 | fubada | straight from the docs |
| 17:46.28 | p3nguin | So that clears up the problem you were having? |
| 17:47.03 | fubada | yes that solves the incoming issues, im also in need of an outgoing dialplan, which is not well covered in the wiki |
| 17:47.23 | fubada | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id242479.html |
| 17:47.32 | fubada | is this the best section of the book for this? |
| 17:48.56 | p3nguin | http://pastebin.com/Piqv4Egj outbound example starts at line 119. |
| 17:49.26 | fubada | nice, thank you |
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| 17:50.14 | fubada | lines 120 & 121, which trunk do they take? |
| 17:50.24 | p3nguin | This is SIP; there is no trunking. |
| 17:50.40 | fubada | i mean, theres no SIP handler specified there |
| 17:50.48 | fubada | like tere is in 122/123 |
| 17:50.55 | fubada | how do you dial those? |
| 17:51.22 | p3nguin | 120 and 121 clearly use Goto(). |
| 17:51.43 | p3nguin | Both go to line 122 when matched. |
| 17:51.58 | fubada | thank you |
| 17:52.21 | p3nguin | I even put commenting as to the purpose. |
| 17:52.55 | talntid | http://paste.ubuntu.com/999439/ |
| 17:53.05 | talntid | I know I am doing that wrong - it just feels wrong. |
| 17:53.23 | talntid | but, I'm not sure what the best way of going about this is? I think I am thinking too deepy into this |
| 17:53.43 | talntid | in regards to the after recording, wait for exten 1,2,3, then hangup after 4 seconds if nothing |
| 17:53.44 | fubada | p3nguin: if i have to support a leading 9, just append 9 to every line and strip 2 isntead? |
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| 17:54.28 | p3nguin | Just don't do it. This isn't 1986. |
| 17:54.46 | fubada | eh the problem is that my extensions are all over the plae |
| 17:54.48 | fubada | place |
| 17:54.56 | fubada | some begin with a 3 |
| 17:54.56 | p3nguin | Time to correct them. |
| 17:55.08 | p3nguin | That's okay. |
| 17:55.31 | p3nguin | Unless you have one extension that matches another extension, there isn't a problem. |
| 17:55.36 | fubada | ok cool |
| 17:56.11 | p3nguin | You can dial both extensions 314 AND 3145551212 and they go to completely different places. |
| 17:56.27 | fubada | gotcha |
| 17:57.32 | fubada | so for example im trying to call 12038075556, as soon as I dial 1203, i get another dialtone |
| 17:57.42 | p3nguin | Sounds like a misconfiguration. |
| 17:57.52 | fubada | i think it tries to call 1203 as an extension (which doesnt even exist) |
| 17:58.15 | p3nguin | If you think the call is being placed into the system, read the verbose CLI output to see what it matched. |
| 17:58.57 | fubada | To: <sip:1203@192.168.60.12;user=phone>;tag=as1e6082d1 |
| 17:59.13 | p3nguin | So 1203 is the name of a device? |
| 17:59.32 | p3nguin | barfs and then goes to get more food so he can barf again. |
| 17:59.35 | fubada | i dont have such a device, i only have 1417 and 1426 at this time |
| 17:59.51 | fubada | i use phone MAC addr for device names in sip.conf |
| 18:00.06 | p3nguin | You must have dial plan that is trying to send calls to 1203 for some reason. |
| 18:00.09 | p3nguin | core set verbose 3 |
| 18:00.15 | p3nguin | Then call it again. |
| 18:01.58 | fubada | not seeing any useful output |
| 18:02.26 | p3nguin | If an extension is executed, there should be output saying so. |
| 18:02.30 | fubada | verbosity is at 3, sip debugging i on |
| 18:02.31 | fubada | is |
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| 18:02.38 | p3nguin | sip debugging need not be enabled. |
| 18:03.26 | fubada | http://pastie.org/3945856 |
| 18:03.36 | fubada | the only line that prints when i dial 1203 is that == Using SIP RTP CoS mark 5 |
| 18:04.42 | p3nguin | If you don't see extension 1203 being executed, then I guess there is no reason for any of what you're saying to actually occur. |
| 18:06.07 | fubada | if i dial an existing extension, i see the log that you are describing |
| 18:06.24 | fubada | howver, 1203 is not a registered extension, and no log occurs |
| 18:06.39 | fubada | however sip still tries to place a call |
| 18:07.05 | p3nguin | What kind of phone are you using? |
| 18:07.10 | fubada | polycom |
| 18:07.24 | fubada | ip 550 sip |
| 18:07.32 | p3nguin | You must have a digit map entry that matches 1203 and thinks it is a complete phone number. |
| 18:08.09 | fubada | http://pastie.org/3945876 |
| 18:08.23 | fubada | i didnt set this up so im not entirely sure but theres the digit map from the phone |
| 18:08.40 | fubada | can that be cleared entirely? |
| 18:08.41 | p3nguin | It's the first pattern. |
| 18:09.07 | p3nguin | I would devise a more appropriate digit map. |
| 18:09.17 | fubada | how about no digitmap at all |
| 18:09.30 | fubada | can I let asterisk handle that? |
| 18:09.40 | p3nguin | I think that will add lots of delay before sending the call after any number has been dialed. |
| 18:09.48 | fubada | got it |
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| 18:10.24 | p3nguin | I think I would add a small delay after the [0-8]xxx so that it does not accept it instantly. |
| 18:10.44 | p3nguin | Maybe 2 seconds? |
| 18:11.08 | p3nguin | Then I would adjust the parts which begin with a 9. |
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| 18:12.32 | fubada | ok |
| 18:12.41 | fubada | i think its still better than dialing 9 |
| 18:12.52 | fubada | (although all my devices will need to be changed) |
| 18:14.02 | fubada | i have a separate config line in the phone called Digimap Timeout: 3|3|3|3|3|3 |
| 18:24.38 | RubyRails | i have a quick, hopefully not too stupid, question... |
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| 18:25.00 | RubyRails | all our calls come into one context that hands it all off to adhesion |
| 18:25.01 | RubyRails | exten => _.,1,NoOp() |
| 18:25.02 | RubyRails | same => n,Set(ADHEARSION_URL=agi://localhost:4573) |
| 18:25.03 | RubyRails | same => n,AGI(${ADHEARSION_URL}) |
| 18:25.34 | RubyRails | how would i make it so that before that, any calls directed to one specific phone number (DID, not extension) go to a different context? |
| 18:26.15 | pabelanger | RubyRails: stop using exten => _., and start defining specific DIDs |
| 18:26.34 | WIMPy | First of all you should take the warning you're getting about using _. seriousely. |
| 18:27.20 | WIMPy | And your DIDs are your extensions. |
| 18:27.30 | RubyRails | there's literally thousands of DIDs that come through there - our adhearsion app handles them |
| 18:27.49 | RubyRails | wait, DIDs == extensions? |
| 18:28.20 | RubyRails | so if we rented the phone number 555-867-5309 from our VOIP provider, |
| 18:28.42 | RubyRails | extend => 5558675309 would catch it when somebody called that number? |
| 18:28.56 | WIMPy | Probably. |
| 18:29.08 | WIMPy | Depends on the format they send, but probably. |
| 18:29.18 | pabelanger | RubyRails: welcome to maintaining an asterisk box. If you don't want to many handle them in a dialplan, create an AGI application to do it |
| 18:31.21 | RubyRails | k thanks guys... |
| 18:31.26 | RubyRails | got a long road ahead of me :p |
| 18:31.45 | carrar | Be sure to take along a towel |
| 18:32.05 | RubyRails | lol, im a hoopy frood |
| 18:33.03 | carrar | Think about getting rid of _. and replacing it with something BETTER! |
| 18:34.45 | RubyRails | what's the logic behind that? I can see how _. could be dangerous if using the dial plans, but if the app written in adhearsion has logic to handle all kinds of calls, does it matter? |
| 18:35.15 | carrar | I would think _X. is safer |
| 18:35.20 | WIMPy | It's most likely to cause trouble because it matches the special extensions. |
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| 18:35.37 | RubyRails | ah ok |
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| 18:36.14 | carrar | _. will match first |
| 18:36.22 | carrar | and you won't match your number |
| 18:36.26 | carrar | that youw ant to catch |
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| 18:36.44 | RubyRails | ooh, even if the phone number comes first in the context? |
| 18:36.55 | carrar | read up pattern matching |
| 18:36.57 | RubyRails | haha, i'd just put that in and reloaded, was wondering why it wasn't working... |
| 18:36.59 | carrar | look at ! too |
| 18:37.03 | WIMPy | Order doesn't matter. |
| 18:37.42 | talntid | http://pastebin.com/5sJxZT27 |
| 18:37.54 | talntid | Trying to get the options to work... suggestions? |
| 18:38.07 | talntid | it plays the human/machine properly, but then doesn't have the options |
| 18:38.36 | RubyRails | wait, but looking at the asterisk book, it says that the most specific is matched first |
| 18:38.52 | RubyRails | isn't 5558675309 more specific than either _. or _x. ? |
| 18:38.58 | WIMPy | That's only kind of true AFAIK. |
| 18:39.14 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 18:39.37 | WIMPy | But you can alsows go for includes to create an order. |
| 18:42.34 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 18:43.42 | RubyRails | ok, changed the all-our-calls from _. to _X. - all seems to be well |
| 18:43.51 | RubyRails | now off to lunch, and then back to learn about includes... |
| 18:43.59 | carrar | get the book |
| 18:44.01 | RubyRails | thanks for the help WIMPy and carrar |
| 18:44.02 | WIMPy | No need |
| 18:44.04 | carrar | ~thebook |
| 18:44.04 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:44.14 | carrar | or read it online |
| 18:44.15 | WIMPy | Numbers will come prior ti _X. |
| 18:44.30 | WIMPy | to |
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| 18:44.43 | RubyRails | hmm... |
| 18:45.10 | kaldemar | includes are matched in the order they appear in a context. it doesn't matter if a latter context has a better match. |
| 18:45.37 | WIMPy | ... and after the context itself. |
| 18:45.38 | RubyRails | okay cool... that sounds like what i need |
| 18:47.48 | kaldemar | extensions in a context always come first and includes after them. |
| 18:52.38 | talntid | http://pastebin.com/a5uQvhgG |
| 18:53.02 | talntid | the last line... not sure why that won't goto humn.. i tried goto(humn,1), s, n... |
| 18:54.11 | talntid | [May 21 11:51:54] WARNING[27149]: pbx.c:4815 __ast_pbx_run: Channel 'SIP/vitel-outbound-00000025' sent into invalid extension 'humn' in context 'fanout_call', but no invalid handler |
| 18:56.38 | carrar | your trying to go outside of your current extension |
| 18:56.43 | carrar | I'm guessing |
| 18:57.13 | carrar | You need to tell it what extension in addition to the label |
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| 19:39.10 | fubada | p3nguin: on your outgoing example voipms-outbound, i dont see how you set callerid |
| 19:39.13 | fubada | i guess you dont? |
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| 19:50.41 | talntid | https://wiki.asterisk.org/wiki/display/AST/Background+and+WaitExten+Applications |
| 19:51.00 | talntid | is it just me, or in that dialplan, is the timeout-handler,1) line incorrect... ? |
| 19:51.20 | Qwell | what's wrong with it? |
| 19:51.25 | Qwell | oh, that one |
| 19:51.51 | talntid | :) |
| 19:52.01 | ChannelZ | (in other words, yes) |
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| 19:52.12 | talntid | yeah, it just looked out of place... |
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| 19:53.29 | Qwell | talntid: add a comment to that page |
| 19:54.16 | talntid | Qwell, I could add a comment pointing out that line is wrong, but I am not sure what it should read, as I am a lowly noob. |
| 19:54.31 | Qwell | "This line is wrong: " |
| 19:54.36 | talntid | roger that |
| 19:55.39 | talntid | heh, now that I registered, I can't even view that page |
| 19:55.40 | talntid | Not Permitted |
| 19:55.41 | talntid | You are not permitted to perform this operation. |
| 19:55.48 | Qwell | are you logged in? |
| 19:55.51 | talntid | yeah |
| 19:56.13 | Qwell | O.o |
| 19:56.14 | p3nguin | fubada: You can set it per phone via sip.conf peer entry, or we can explore other options if that isn't sufficient. |
| 19:56.18 | talntid | when logged in, I can't even -view- the page. |
| 19:56.36 | SeRi | <PROTECTED> |
| 19:56.51 | p3nguin | Just got in. |
| 19:57.01 | SeRi | I finally have time to sit and get caught up with life |
| 19:57.06 | talntid | when I log out, I can see it again |
| 19:57.14 | SeRi | new job is for sure sucking the life out of me |
| 19:58.28 | p3nguin | fubada: I actually set the caller id number in the peer entry to be the internal name/number, then use a variable for each device for external caller id numbers. |
| 20:00.53 | p3nguin | fubada: I'll write up an example of that for you... |
| 20:01.04 | fubada | thanks, im tryijng to solve it by having globals in extensions.con |
| 20:01.06 | fubada | like this: |
| 20:02.17 | fubada | http://pastie.org/private/vemvoasohrmujbfe55zq |
| 20:02.40 | fubada | but Set(CALLERID(num)=${CID_${CALLERID(num)_NUM}} doesnt work as expected |
| 20:03.53 | fubada | im sure you have a much better way ;) |
| 20:09.00 | p3nguin | fubada: For each device, configure a line in its peer entry with its real callerid number: setvar=externalCID=3145551212 |
| 20:09.17 | fubada | I use a template for my peers |
| 20:09.58 | p3nguin | fubada: Then in dial plan, before the Dial(), set the caller id number: exten => _1NXXNXXXXXX,n,Set(CALLERID(num)=${IF($[${EXISTS(${externalCID})}]?${externalCID}:3145486554)}); |
| 20:10.30 | p3nguin | fubada: Doing it this way, if you have not configured the externalCID for a device, it will fall back to a default number of 3145486554. |
| 20:10.33 | p3nguin | for example. |
| 20:10.44 | fubada | thank you for that |
| 20:11.25 | p3nguin | Using templates should not affect the outcome when done correctly. |
| 20:11.40 | fubada | http://pastie.org/3946476 |
| 20:11.48 | fubada | is how my peers look in sip.conf |
| 20:12.00 | fubada | can I still set the externalCID on each? |
| 20:12.13 | *** join/#asterisk jsjc (~Adium@119.Red-2-136-85.dynamicIP.rima-tde.net) |
| 20:13.02 | p3nguin | I see a few things I don't like: every phone uses the same password, you aren't using dtmfmode of rfc2833, and you're allowing both alaw and ulaw. |
| 20:13.26 | p3nguin | How do I edit this paste? |
| 20:13.34 | fubada | changed dtmfmode to rfc2833 |
| 20:13.44 | fubada | theres an Edit link on the right |
| 20:13.58 | p3nguin | not for me. |
| 20:13.59 | fubada | and youre right, every phone does use the same passwd |
| 20:14.32 | fubada | http://pastebin.ca/2151829 |
| 20:14.36 | fubada | same thing in pastebin |
| 20:17.18 | p3nguin | http://pastebin.com/LLkvXnSv |
| 20:17.35 | fubada | thank you |
| 20:17.57 | fubada | setvar=externalCID="Foo Bar" <2039873397> |
| 20:17.59 | fubada | is this also valid? |
| 20:18.07 | p3nguin | Don't do it. |
| 20:18.11 | fubada | ok |
| 20:18.19 | p3nguin | Quotes are not valid like that. |
| 20:18.48 | p3nguin | Correct format of CALLERID(all) is CALLERID(all)=Some Name <2039873397> |
| 20:18.55 | p3nguin | But we're not dealing with the name. |
| 20:19.08 | p3nguin | For that, we need to add something else into the peers... |
| 20:19.22 | fubada | I think the name came be dropped |
| 20:20.35 | p3nguin | http://pastebin.com/LLkvXnSv |
| 20:20.58 | fubada | cool, thats useful |
| 20:21.11 | p3nguin | Wait, I messed that up for your implementation. |
| 20:21.34 | p3nguin | refresh |
| 20:22.08 | fubada | dont see anything change |
| 20:22.14 | fubada | after refresh |
| 20:22.17 | fubada | what did you correct? |
| 20:22.44 | p3nguin | Doing it this way, when Joe calls Jane, Jane's phone will show "Joe 1111" is calling, but when Joe calls outside, his callerid number will set to 3145551111. |
| 20:22.54 | p3nguin | I set numbers like 1133 instead of 3333. |
| 20:23.02 | p3nguin | changed it, though. |
| 20:23.22 | fubada | ok, thank you |
| 20:25.12 | p3nguin | There may be other ways to accomplish this, but my method seems to work for every system I have used it on. |
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| 20:26.46 | fubada | p3nguin: callerid name is only set on local extention calls in this configuration right? |
| 20:26.55 | fubada | external calls just get the number set? |
| 20:27.15 | *** join/#asterisk skirmisha (~vk@46.47.82.189) |
| 20:27.16 | p3nguin | I thought I already explained that part. |
| 20:27.19 | skirmisha | guys |
| 20:27.35 | skirmisha | can someone tell me how to configure quintum caller id translation |
| 20:28.10 | skirmisha | i am doing something wrong and number is not replaced at all |
| 20:28.17 | skirmisha | how should i match |
| 20:28.43 | p3nguin | fubada: When you have a callerid set for a device, that is the callerid info used for all calls... until and unless you change it in dial plan. |
| 20:29.18 | p3nguin | The Set() I gave you earlier is how it gets changed when making a call outside. |
| 20:29.47 | fubada | right, but i think the name will be dropped on outgoing calls according to: http://pastebin.com/ZyHjZJGf |
| 20:30.05 | fubada | i think the first set needs to be changed like: exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=${CID_${CALLERID(all)} |
| 20:30.13 | fubada | er |
| 20:30.28 | fubada | thats not right |
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| 20:32.28 | p3nguin | fubada: http://pastebin.com/iPtvgNVe |
| 20:34.01 | p3nguin | To see it, add a Verbose() before the Dial(): http://pastebin.com/hiHdmfwG |
| 20:35.25 | p3nguin | You can't control the caller id name that a callee will receive on a per-call basis, so there isn't a lot of need to adjust it for a call going out. |
| 20:37.24 | fubada | thanks, looking |
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| 20:48.15 | TechSmurf | Anyone have a preferred managed PDU? |
| 20:48.16 | fubada | p3nguin: im having issues with one way audio on outgoing calls, phones behind asterisk have no incoming audio |
| 20:48.46 | p3nguin | behind? |
| 20:48.49 | fubada | in the past with the gui, i was able to tweak nat/externalip settings and eeventually the problem was solved |
| 20:48.54 | fubada | i mean phones managed by asterisk |
| 20:49.02 | p3nguin | So all the phones, then. |
| 20:49.05 | fubada | yes |
| 20:49.13 | fubada | im guessing its RTP issue |
| 20:49.13 | p3nguin | That's a NAT configuration problem. |
| 20:49.24 | p3nguin | ~sipnat |
| 20:49.24 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 20:49.32 | fubada | my pbx is not behind nat, it has an external IP interface, my phones ARE behind nat |
| 20:49.52 | p3nguin | Are they on the same LAN? |
| 20:49.57 | fubada | yes |
| 20:50.07 | p3nguin | The asterisk box acts as a gateway, then? |
| 20:50.12 | fubada | but the pbx has a secondary interface directly exposed to the net |
| 20:50.20 | fubada | no, the asterisk does not act as a gateway |
| 20:50.47 | p3nguin | But it is on the LAN as well as world-facing? |
| 20:50.55 | fubada | yes |
| 20:51.09 | TechSmurf | comforts p3nguin |
| 20:51.28 | p3nguin | Configure the nat and localnet values, at least. |
| 20:51.37 | fubada | already in place |
| 20:51.49 | fubada | in general nat=no, externip and localnet are set |
| 20:51.55 | p3nguin | A sip debug will be required. |
| 20:51.55 | fubada | in phone template, nat=yes |
| 20:52.25 | p3nguin | If asterisk has a public address, it already knows its own external address, so that setting isn't needed. |
| 20:52.49 | p3nguin | And if they phones are on the same LAN as asterisk, you could safely set nat=no for phones. |
| 20:52.56 | p3nguin | s/they/the/ |
| 20:53.09 | fubada | p3nguin: problem fixed, i needed canreinvite=no |
| 20:53.10 | fubada | in genera |
| 20:53.11 | fubada | l |
| 20:53.17 | p3nguin | What version of asterisk are you using? |
| 20:53.23 | fubada | 1.8.11 |
| 20:53.30 | p3nguin | It's directmedia. |
| 20:53.37 | p3nguin | ~sipnat |
| 20:53.37 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 20:53.40 | p3nguin | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:54.05 | fubada | oh, the first link suggests canreinvite |
| 20:54.09 | fubada | looking for directmedia :) |
| 20:54.24 | p3nguin | canreinvite is the only setting. |
| 20:54.26 | p3nguin | It's directmedia now. |
| 20:54.42 | p3nguin | Just like externip is the old setting and it is externaddr now. |
| 20:54.55 | p3nguin | but that isn't needed when asterisk has a public address. |
| 21:03.54 | SeRi | p3nguin: did you move to asterisk 10/ |
| 21:03.58 | SeRi | ? |
| 21:04.04 | p3nguin | Nope, and I never will. |
| 21:04.16 | SeRi | ah! nice to know.... Is it that bad? |
| 21:04.18 | p3nguin | I don't do non-LTS branches. |
| 21:04.23 | SeRi | ah ic |
| 21:04.34 | p3nguin | The next branch I see will be 11. |
| 21:04.39 | SeRi | cool |
| 21:05.15 | SeRi | I am still on 1.8.8.1 |
| 21:05.32 | p3nguin | I think I use 1.8.8.0 on my own box. |
| 21:05.40 | p3nguin | Newer for other people. |
| 21:05.48 | SeRi | cool did you ever get the atom box you wanted? |
| 21:06.08 | p3nguin | I don't remember having any intention of getting one. |
| 21:06.25 | SeRi | Mhhhhh That was probably me. |
| 21:06.26 | SeRi | lol |
| 21:08.55 | SeRi | I am still using my alix and dont see a reason to upgrade. |
| 21:10.27 | p3nguin | I'm still using my Neoware. |
| 21:10.51 | SeRi | cool |
| 21:11.04 | SeRi | brb wife calls |
| 21:17.24 | *** join/#asterisk af_ (~getsmart@78.134.18.236) |
| 21:21.43 | fubada | any point in setting callerid for international calls? |
| 21:22.30 | p3nguin | Do you want people to know you are calling and how to reach you? |
| 21:23.03 | fubada | :) yes |
| 21:23.24 | p3nguin | Then you will probably want to set the callerid. |
| 21:23.55 | p3nguin | You may have to check with your ITSP for the format of callerid they accept for international calls. |
| 21:25.17 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:28.30 | SeRi | p3nguin: I sold enough accounts on voip.ms that I no longer have to pay for my bill :) |
| 21:28.34 | SeRi | is nice. |
| 21:28.43 | p3nguin | Yeah. |
| 21:30.49 | p3nguin | I usually only have to add funds when I need to do ports. |
| 21:31.43 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 21:31.59 | fubada | p3nguin: can I show you my polycom digit map i arrived at |
| 21:32.13 | fubada | tell me if it makes any sense |
| 21:32.15 | p3nguin | sure |
| 21:32.20 | fubada | [2-9]11|0T|011xxx.T|[1-9]xxx.T|1[2-9]xxxxxxxxx|[2-9]xxxxxxxxx.T|[2-9]xxxxxx.T|*xx |
| 21:32.26 | fubada | and for every T, its 2 seconds wait |
| 21:33.01 | p3nguin | [1-9]xxx.T is going to be for local phones? |
| 21:33.07 | fubada | yes |
| 21:33.12 | p3nguin | You don't want the dot, then. |
| 21:33.25 | p3nguin | At least I don't think you want it. |
| 21:34.01 | p3nguin | I'm not a Polycom guy, so maybe another opinion would help. |
| 21:34.35 | p3nguin | [2-9]xxxxxxxxx probably doesn't need the .T on it, too. |
| 21:34.53 | p3nguin | Dialing a 10-digit NANP number should go immediately instead of waiting. |
| 21:35.32 | fubada | ok |
| 21:35.34 | fubada | thank you |
| 21:35.41 | p3nguin | Do you know what the dot does? |
| 21:35.51 | fubada | i thought it was wild |
| 21:35.56 | p3nguin | I don't, but I assumed it means any char. |
| 21:36.16 | p3nguin | x would be digits, but . would be anything. |
| 21:36.22 | d_preston215 | Can you create a meetme conference from the command line? |
| 21:36.25 | p3nguin | I assumed |
| 21:36.57 | p3nguin | d_preston215: Sure. You can originate it. |
| 21:38.44 | d_preston215 | Like creating a call? |
| 21:38.50 | p3nguin | Exactly. |
| 21:39.37 | p3nguin | For example: channel originate SIP/your-phone application MeetMe 1234,d |
| 21:39.54 | p3nguin | dynamically create MeetMe conference number 1234. |
| 21:40.05 | p3nguin | and your phone will be in it. |
| 21:40.19 | d_preston215 | Ok |
| 21:40.47 | d_preston215 | And I can just easily join any calls or channels I want into it. |
| 21:40.56 | p3nguin | Yes. |
| 21:41.03 | d_preston215 | And then after, I can just end the conference? |
| 21:41.29 | p3nguin | Depending on what you are trying to add to it, you may have to create a small piece of dial plan for the conf. |
| 21:42.03 | d_preston215 | Dumping multiple chanspy's into it. |
| 21:42.32 | p3nguin | Then you could do something like: channel originate Local/1234@meetme application ChanSpy SIP/whatever-00001111 |
| 21:42.51 | p3nguin | But you would have to create the dialplan for 1234@meetme ahead of time. |
| 21:43.05 | d_preston215 | Yeah. |
| 21:47.18 | pabelanger | Seem like an overcomplicated way to use dynamic conferences |
| 21:48.14 | p3nguin | I can't think of a better way to dump multiple ChanSpy()s into a single conference. |
| 21:48.49 | d_preston215 | I would use spygroup for what I need to do, but its not flexible enough to specify two extensions per spygroup. |
| 21:49.09 | d_preston215 | Then again, I feel completely noobish at asterisk right now. |
| 21:50.50 | p3nguin | Both ExtenSpy() and ChanSpy() only listen to one channel at a time, regardless of grouping channels or extensions together. |
| 21:51.13 | d_preston215 | I get that. |
| 21:51.48 | d_preston215 | I only need it to monitor the active channel or extension. |
| 21:52.36 | p3nguin | That sounds like you only need to listen to one channel at any given time. |
| 21:53.48 | d_preston215 | I need it to have both available. |
| 21:53.50 | p3nguin | If you group some channels together, you can spy on the group. Then you'll hear a channel when it goes in use. |
| 21:54.01 | d_preston215 | Thats what I want. |
| 21:54.06 | p3nguin | If a second goes in use, you can cycle the channels with the star key. |
| 21:54.25 | p3nguin | You won't hear two channels at once, though. |
| 21:54.45 | p3nguin | I also don't know if you can spy an extension like that. I think you can only spy channels in that manner. |
| 21:55.04 | p3nguin | I think in order for extenspy to work, a channel has to be using the extension already. |
| 21:55.16 | fubada | how is parking supposed to work? I have it set up for *724, with the park range of 700-720 |
| 21:55.25 | fubada | but im not sure how to use the feature |
| 21:55.45 | p3nguin | Perform an attended transfer to the park extension. |
| 21:56.12 | p3nguin | You have real phones with normal buttons, so you don't need to use DTMF codes in most cases. |
| 21:57.00 | p3nguin | If the park extension is 700, be on a call, press transfer, dial 700, wait to hear the parking space number, press transfer key again. |
| 21:57.25 | p3nguin | To pick up the call, directly dial the parking space number. |
| 21:57.29 | fubada | exten => *724,1,Page(${MULTICAST},i,120) |
| 21:57.36 | fubada | nevermind |
| 21:57.40 | p3nguin | Don't know what that is. |
| 21:57.45 | p3nguin | It isn't parking, though. |
| 21:57.50 | fubada | right |
| 21:59.50 | ZenMaster | so here is a new one. I have a 8x 6755i Ip Phone connected to *. I can call another machine logged in via xlite, but when I answer it drops the call. |
| 22:00.02 | ZenMaster | However machines running xlite can call and answer eachother without a problem. |
| 22:00.12 | ZenMaster | 8x = 8x8. |
| 22:00.34 | navaismo | ZenMaster, codecs issue? |
| 22:00.35 | ZenMaster | But they also can not call the hard phone. |
| 22:00.44 | p3nguin | sip debug needed |
| 22:00.54 | fubada | p3nguin: do I need to include my parkedcalls context in LocalSets? |
| 22:00.55 | ZenMaster | navaismo: I did see an error at one point and time saying something like that. Could not assign audio or something. |
| 22:00.58 | navaismo | start guessing |
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| 22:01.13 | ZenMaster | p3nguin: Well one thing I see in the log file is it tels me that the line is busy or congested... |
| 22:01.20 | p3nguin | fubada: Probably. |
| 22:01.27 | ZenMaster | This is hard phone to softphone. |
| 22:01.51 | navaismo | yeah show us your sip debug, cli output and make sure you are using same valid codecs |
| 22:02.26 | ZenMaster | One second. |
| 22:03.07 | ZenMaster | Doh eth. This has been the day from hell. |
| 22:03.18 | ZenMaster | This might be a second, my synergy broke.... |
| 22:04.18 | ZenMaster | Like the third time today.... |
| 22:05.57 | fubada | p3nguin: nothing happens when i hit transfer and my park extension |
| 22:06.29 | fubada | http://pastie.org/3947001 |
| 22:06.32 | fubada | my features.conf |
| 22:06.34 | p3nguin | Do you have the feature enabled and loaded? Did you remember to reload the dial plan after including the parkedcalls context? |
| 22:06.46 | fubada | i restarted the whole thing |
| 22:06.50 | p3nguin | ewww |
| 22:07.01 | p3nguin | So sloppy and time consuming. |
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| 22:16.14 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 22:16.44 | talntid | when I use AMI to start a call... and connect it to a context in the dialplan, in the context, can I get the called #? |
| 22:17.20 | talntid | without passing it as a variable via AMI. I can do that, but just curious if there is another way |
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| 22:24.44 | p3nguin | Hmm. You can't just connect a call to a context. You have to use extensions. |
| 22:25.02 | p3nguin | If you know the extension you're entering, isn't that already the info you're looking for? |
| 22:25.17 | talntid | hmm |
| 22:25.18 | p3nguin | Or do you need to know the extension programmatically? |
| 22:25.27 | talntid | well, you know how like... in callfiles... |
| 22:25.45 | talntid | you can have * dial the call out, and connect it to a context/extension? |
| 22:25.58 | talntid | for example in that fanout_call thing we were working on.. |
| 22:26.02 | talntid | we were calling it via originate |
| 22:26.04 | p3nguin | Calls must "connect" to extensions. Extensions are within contexts. |
| 22:26.26 | p3nguin | In the case of that thing, the extension was 's'. |
| 22:26.43 | talntid | right, but I want to know the phone# that it is connected to |
| 22:27.01 | p3nguin | Oh, the number that was called on the other side of that call... |
| 22:27.09 | talntid | yeah |
| 22:27.34 | p3nguin | I would probably write it into a variable and refer to it within the extension. |
| 22:27.39 | talntid | the exten is the easy one... I'm looking for like... ${number-channel-is-connected-to} |
| 22:27.53 | talntid | yeah, I figured I'd have to do it as a variable - not a big deal |
| 22:28.16 | p3nguin | You were using direct SIP channels, so you'd have to alter that to use a Local channel and a small bit of dial plan to set the variable. |
| 22:29.51 | p3nguin | Alternatively, you might be able to show the channels in use, parse which channel is using s@fanout_calls, see what channel that one is bridged to, and then check the extension dialed on that channel. |
| 22:29.59 | p3nguin | That seems like a lot of work. |
| 22:30.25 | p3nguin | I'd rather set a variable and parse it in the other extension. |
| 22:30.26 | talntid | much easier to pass variabkle |
| 22:30.35 | talntid | just didn't want to do it, if there was already another built in way |
| 22:30.49 | citywok | are there any speech to text modules other than lumenvox? |
| 22:32.24 | navaismo | citywok: Google module, not official, not supported, not commercial, but works! |
| 22:32.37 | citywok | navaismo: yea i played with it and it works fairly well actually |
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| 22:32.56 | citywok | navaismo: but i don't want to build commercial functionality based on an unsupported, free product that they could kill at any time. lol |
| 22:33.16 | p3nguin | s/could/will/ |
| 22:33.22 | danfromuk | Hi. I'm just upgrading from 1.4 to 1.8. I'm trying to find the replacement for 'extensions reload' but struggling. can anyone assist? |
| 22:33.24 | citywok | :p |
| 22:33.30 | citywok | dialplan reload |
| 22:33.34 | danfromuk | ah, thanks |
| 22:33.35 | p3nguin | danfromuk: dialplan reload |
| 22:33.50 | p3nguin | Even 1.4 used dialplan reload. |
| 22:33.58 | citywok | module reload extensions maybe? |
| 22:34.10 | danfromuk | p3nguin: i wasnt aware of that. i always did extensions reload. |
| 22:34.25 | p3nguin | I've never even heard of it. |
| 22:35.17 | citywok | p3nguin: The 'extensions reload' command is deprecated and will be removed in a future release. Please use 'dialplan reload' instead. |
| 22:35.26 | citywok | i just tried it on a 1.4.36 test box i have |
| 22:35.41 | p3nguin | Must be left over from versions <=1.2 |
| 22:35.49 | p3nguin | started in 1.4 |
| 22:36.11 | navaismo | citywok, me too, hope someday google release that module or make an affordable license not like google maps license |
| 22:36.59 | citywok | my boss wants to use a true IVR w/ speech, but lumenvox is $$$ |
| 22:38.51 | navaismo | yep but works |
| 22:39.05 | talntid | citywok, there's another one |
| 22:39.14 | talntid | and it wasn't terrible, one second |
| 22:39.34 | talntid | http://zaf.github.com/asterisk-speech-recog/ |
| 22:39.48 | talntid | oh, it's the google speech api. already mentioned. |
| 22:40.10 | talntid | i experimented with it a few days ago. :) |
| 22:40.18 | navaismo | maybe this link helps http://www.voip-info.org/wiki/view/Voice+recognition |
| 22:40.41 | citywok | talntid: yea it works pretty well, and that's exactly what i used heh |
| 22:42.34 | citywok | navaismo: hey, that's helpful |
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| 22:42.43 | citywok | navaismo: lol microsoft tellme, but you can't buy it or anything |
| 22:43.02 | danfromuk | argh. one way audio |
| 22:43.45 | navaismo | lol |
| 22:44.01 | citywok | i went straight to msdn to see if i could download it lol |
| 22:46.49 | danfromuk | does anyone have an idea why i get one way audio. asterisk on public ip, softphone behind a nat, no firewalls enabled. config worked fine in 1.4. just trying the same config on 1.8 on a new server. |
| 22:50.01 | navaismo | danfromuk, do you set up exterip, externhost localnet, open rtp ports and set the remote device to NAt=yes? |
| 22:50.26 | danfromuk | exterip is set correctly. |
| 22:50.41 | danfromuk | what should externhost localnet be set to? |
| 22:50.58 | danfromuk | bear in mind that i have no idea what the local ip range of remote phones will be |
| 22:51.02 | navaismo | externhost only if you have a dynamic IP |
| 22:51.08 | p3nguin | externhost is used for a dynamic DNS host name. |
| 22:51.17 | p3nguin | If you have a static address, use externaddr instead. |
| 22:51.18 | danfromuk | its a static ip so i can ignore that |
| 22:51.26 | p3nguin | externaddr, then. |
| 22:51.30 | navaismo | localnet is the asterisk's localnet |
| 22:51.38 | p3nguin | localnet is whatever network asterisk is on. |
| 22:51.58 | danfromuk | only one ip assigned to the box. its not in a range. |
| 22:52.08 | p3nguin | Asterisk isn't behind a NAT? |
| 22:52.24 | danfromuk | no, asterisk has a public ip. |
| 22:52.26 | p3nguin | Then it has no localnet and externaddr/externhost have no reason to be configured. |
| 22:52.42 | p3nguin | If it has a public address, it already knows its own address. |
| 22:52.58 | danfromuk | ok |
| 22:53.02 | navaismo | pbx firewall? |
| 22:53.07 | danfromuk | nothing |
| 22:53.09 | p3nguin | extern{addr,host} are for asterisk behind a NAT where it does not know its own public address. |
| 22:53.15 | danfromuk | ill enable the firewall later once all is ok |
| 22:53.40 | p3nguin | Be sure you have set nat=yes for your phones. |
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| 22:53.52 | danfromuk | let me double check. im sure it was from the 1.4 config |
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| 22:55.00 | danfromuk | i cant see it under sip show peer name |
| 22:55.29 | navaismo | see it in your peer config |
| 22:55.46 | p3nguin | It's marked as force_rport, I think. |
| 22:55.47 | danfromuk | yep. they are all set to yes |
| 22:56.09 | p3nguin | Did you set directmedia to no? |
| 22:56.10 | danfromuk | ok, force rport is yes |
| 22:56.25 | danfromuk | DirectMedia : No |
| 22:56.36 | p3nguin | I guess it is time for a sip debug. |
| 22:56.43 | danfromuk | ok, 1sec |
| 22:56.48 | danfromuk | i'll pastebin a dialog |
| 22:58.15 | danfromuk | http://pastebin.com/CwEiuhJ1 |
| 22:59.53 | danfromuk | dont know if this is anything. i get this warning during asterisk start up |
| 22:59.53 | danfromuk | <PROTECTED> |
| 23:01.22 | danfromuk | I also get this warning |
| 23:01.23 | danfromuk | http://pastebin.com/cGr2kRR7 |
| 23:03.44 | p3nguin | The warning means they want you to set nat=yes in the general section and don't set it anywhere else in any of the peers. nat=yes will be a default value used for all the devices configured. It doesn't force anything to work incorrectly, just assumes devices are behind NAT and allows them to work behind NAT. |
| 23:04.28 | p3nguin | The transport warning can be rectified by adding transport=udp to the general section, too. |
| 23:07.43 | danfromuk | ok, ive added that just for neatness |
| 23:11.01 | danfromuk | audio is working fine between the asterisk server and our sip trunks. |
| 23:11.06 | danfromuk | must be a nat issue |
| 23:13.04 | danfromuk | argh. i know what the problem is. |
| 23:13.20 | danfromuk | i think the stupid softphone sending the audio to the wrong sound device |
| 23:14.42 | danfromuk | yep |
| 23:14.43 | danfromuk | thats |
| 23:14.44 | danfromuk | it |
| 23:14.50 | danfromuk | thanks for all your help |
| 23:14.55 | danfromuk | sorry to mess you around. |
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