| 00:03.55 | paulc | wizard171: thanks.. 1.8 might be worth a look then.. although there's really nothing wrong with the way it works in 1.6.x for what I want to do |
| 00:15.59 | infratel | tron was just filmed about 5 kms from here |
| 00:16.05 | infratel | if anyone is into movies |
| 00:22.28 | JunK-Y | paulc: ! |
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| 00:40.22 | wizard171 | shmaltz: sorry ... I am back ... wading thru now ... |
| 00:40.39 | shmaltz | thanks wizard171 |
| 00:43.31 | wizard171 | ah, you are still getting the same "dynamic load" error? |
| 00:45.15 | wizard171 | ah, well, you know what I meant, is the "undefined symbol: cap_set_proc" still coming no loading of res_musiconhold ? |
| 00:51.29 | shmaltz | wizard171, yes |
| 00:53.06 | wizard171 | okay, then, I think ... its not linking against your "/lib/libcap.so??" ... we can verify by doing a "ldd /usr/sbin/asterisk" at a command line ... a line like that should be there ... if not, let me know ... |
| 00:57.47 | shmaltz | wizard171, here is the pb of that command: http://pastebin.com/m6JTWpX8 |
| 00:59.45 | wizard171 | well, now, can you look for "libcap.so*" in your "/usr/lib" directory? if its there, is the same one in "/lib/libcap.so*"? |
| 01:00.15 | shmaltz | its there, checking now if its the same |
| 01:00.25 | wizard171 | the one in "/lib/libcap.so*" is the one we want ! |
| 01:01.17 | shmaltz | wizard171, so how do I make the one from /usr/lib be the same as the one from /lib? |
| 01:02.42 | shmaltz | there is none in /usr/lib |
| 01:02.42 | wizard171 | first, copy the existing one "cp /usr/lib/libcap.so /usr/lib/libcap-so-bkup" or the like (you may have "libcap.so.2" etc ...) |
| 01:02.51 | shmaltz | only a /usr/lib/libcap.a |
| 01:03.01 | wizard171 | ah, better .. |
| 01:03.50 | wizard171 | then, use "mv /usr/lib/libcap.a /usr/lib/libcap-a.hold" without the quotes ... |
| 01:04.18 | wizard171 | then, you guessed it, "recompile" ... |
| 01:04.48 | shmaltz | lets do that :) |
| 01:06.47 | wizard171 | then, just to make sure ... after the compile ... rerun the "ldd /usr/sbin/asterisk" ... and look for the "libcap.so.2 => /lib/libcap.so.2" line, if its there, it worked! (and so should it) |
| 01:06.51 | shmaltz | is recompiling asterisk |
| 01:08.04 | wizard171 | then, use "mv /usr/lib/libcap-a.hold /usr/lib/libcap.a" after you are done (just in case something else needs it there) |
| 01:08.34 | wizard171 | it won't, but I like to ... put things back where I got them from ... :) |
| 01:17.24 | shmaltz | wizard171,do I have to do make install as well? |
| 01:17.39 | wizard171 | yes, before we check with "ldd" |
| 01:18.35 | shmaltz | kisses wizard171 |
| 01:18.36 | shmaltz | \:D/ |
| 01:18.46 | shmaltz | Its working, ya da man |
| 01:19.17 | wizard171 | ah, you are welcome ... I like kisses ... and all sorts of ... other things ... :) |
| 01:20.11 | wizard171 | my apologies for the delay ... I may be slow, but I get there eventually ... ;) |
| 01:21.18 | shmaltz | nah, we all have things that we have to do ;) |
| 01:21.26 | shmaltz | what other things do you like ;) |
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| 01:25.28 | shmaltz | i was kicked, did I miss anything? |
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| 01:32.26 | wizard171a | hmmm ... that was strange ... my "nick" on "freenode" is being flaky for some reason ? |
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| 01:33.53 | shmaltz | wizard171a, I had the same problem |
| 01:34.12 | shmaltz | wizard, msg nickserv like this: /msg nickserv username pass |
| 01:34.22 | shmaltz | then use /msg nickser ghost username |
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| 01:38.51 | wizard171 | okay, thanks ... but, I had to do a "windows" version on it ... :) |
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| 01:40.14 | wizard171 | shmaltz: well, I guess, when all else fails, just "restart" the dang thing ... eh? |
| 01:40.27 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 01:40.43 | shmaltz | :) |
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| 01:57.42 | jShaf | <-- noob |
| 01:57.49 | jShaf | installed and started asterisk, got this: |
| 01:57.59 | jShaf | error loading res_musiconhold.so |
| 01:58.07 | jShaf | and three other modules |
| 01:58.12 | jShaf | the rest of the modules loaded |
| 01:58.30 | jShaf | any idea why off head? |
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| 02:00.30 | wizard171 | perhaps a "pastebin" of the relevant messages? (and the two or three lines just before "error loading" for each?) |
| 02:02.48 | jShaf | wizard171: how about this? http://pastebin.com/qj1VFPaY |
| 02:06.34 | wizard171 | jShaf: ah, yet another "linking" problem ... let me see what else ... |
| 02:07.02 | jShaf | basically, i did this |
| 02:07.24 | jShaf | wizard171: make && make install && make samples |
| 02:07.34 | jShaf | then at prompt > asterisk |
| 02:12.58 | wizard171 | jShaf: what version are you building? and what version of xtools? |
| 02:26.47 | jShaf | 1.6.1.1 |
| 02:26.49 | jShaf | xtools? |
| 02:30.22 | wizard171 | jShaf: to me, this looks like a "linking" problem ... the module you are loading (res_musiconhold.so for example), at the "make" step did not have one of the paths set correctly ... probably during "configure" ... did you provide "configure" with anything? |
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| 03:22.07 | sawgood | I have a * 1.4 box with a very plain vanilla sip.conf and extensions.conf (in fact I only have one extension in extensions.conf) 1001 |
| 03:22.25 | sawgood | When I try to call OUTBOUND I keep getting an invite error from 'x1001' |
| 03:22.40 | sawgood | I cannot find this x1001 in any config file (what could be sending out the x1001 |
| 03:22.44 | sawgood | the extension is 1001 |
| 03:23.11 | sawgood | The phone GUI! |
| 03:23.15 | sawgood | let me look at that |
| 03:23.35 | WIMPy | What is the meaning of OUTBOUND if you have only one extension? |
| 03:23.58 | sawgood | outbound dialing to a SIP trunk |
| 03:24.03 | sawgood | from the one extension |
| 03:24.17 | WIMPy | So there is at least another extension for that? |
| 03:25.06 | sawgood | no other extension in extensions.conf (as far as a physical phone goes) I have an outbound route in extensions.conf |
| 03:25.24 | sawgood | Failed to authenticate on INVITE to '"1001" |
| 03:25.45 | sawgood | This is the error message I get at the console when I try to dial outbound to a cell phone using the SIP trunk |
| 03:26.42 | WIMPy | Then show us what you dial and what your sip peer looks like. |
| 03:37.47 | sawgood | I see/found the concern (when a call leaves my IP PBX) using the SIP trunk, the invite arrives at the ITSP with the extension 1001 sending the call (not the username of peer) ... |
| 03:37.47 | sawgood | is there a way to fix this? |
| 03:38.23 | WIMPy | Did you configure a peer in your sip.conf at all? |
| 03:38.36 | sawgood | yes of course |
| 03:39.00 | WIMPy | Goo. Then try fromuser= |
| 03:39.23 | sawgood | in the peer details, I definitely have the line fromuser= |
| 03:39.54 | WIMPy | Hmm. There's also defaultuser, but I'n not sure where exactely that kicks in. |
| 03:40.07 | sawgood | cool ... I'll look that up |
| 03:40.37 | WIMPy | And you are definitely using that peer in your dial? |
| 03:41.14 | sawgood | definitely ... (SIP/peername/${EXTEN}) |
| 03:41.29 | WIMPy | ok |
| 03:41.51 | sawgood | I'm sorry ... I am using the IP address off the peer in the dial plan |
| 03:41.52 | sawgood | sorry! |
| 03:43.00 | WIMPy | At least you found it. |
| 03:43.16 | sawgood | I'm getting a bit better :> |
| 03:44.09 | sawgood | What about using the [authentication] contex? |
| 03:44.15 | sawgood | could that be important? |
| 03:44.45 | WIMPy | Where does that come from? |
| 03:45.03 | sawgood | Well, once not that long ago, I had to have BOTH a register= and a auth= |
| 03:45.11 | sawgood | before I could make calls with one ITSP |
| 03:45.36 | WIMPy | Ah, the proxy stuff. Never used that. |
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| 04:23.43 | sawgood | WIMPy: check this out, when I 'changed' the (SIP/ip_address) to (SIP/peername) outbound calls flowed fine |
| 04:23.52 | sawgood | if the IP address was used instead of the peername it did not work? |
| 04:24.37 | sawgood | I can understand now (because all the authentication is listed in the peername details) |
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| 04:24.38 | sawgood | amazing! |
| 04:29.24 | sawgood | the fromuser= statement in sip.conf will OVERRIDE any callerID num entry you have in extensions.conf |
| 04:49.36 | CaneToad | I am registering asterisk with an external provider and am using /[extension] as part of the registration so that incoming calls are forwarded to that extension...and incoming calls work, but there's something different from when registering the ATA directly with the provider... |
| 04:49.38 | CaneToad | the phone displays caller id but alternates between showing the extension number and the caller id, and the phone's distinctive ring for certain numbers doesn't work...any ideas how I might get it to work? |
| 05:06.38 | sawgood | CaneToad: do you have one single account with the provider which you a 'switching' between Asterisk and an ATA? |
| 05:07.03 | sawgood | I meant switching 'between' the Asterisk box and an ATA |
| 05:11.08 | sawgood | Any tips on how-to: get two totally seperate and independent of each other (SIP trunk accounts) registered on Asterisk and working at the same time ... WHEN ... both trunks are going to the same provider (using the same IP address but different authentication information) |
| 05:12.47 | sawgood | Right now, when one of the two accounts registeres, it will 'kick off' the registration of the 2nd account |
| 05:13.06 | sawgood | Then, the 2nd account will re-register and 'kick off' the 1st account and this repeats over time |
| 05:13.11 | drmessano | sawgood, sounds like your ITSP's problem |
| 05:13.20 | drmessano | sawgood, been doing that for years with no issues |
| 05:13.30 | sawgood | drmessano: excellent then! |
| 05:13.32 | sawgood | thanks ... |
| 05:13.42 | sawgood | anything special in your sip.conf file process? |
| 05:14.25 | drmessano | Nope |
| 05:14.27 | sawgood | When I ask the ITSP for 'help' ... they say it is very doable, but they never seem to have the details to do i t |
| 05:14.56 | drmessano | There's nothing to "do" here. Setup is the same for any provider, same or different |
| 05:15.05 | sawgood | its probably ok because I'll probably switch one of the two providers with another one to avoid this |
| 05:15.07 | drmessano | There is no shared information between |
| 05:15.25 | sawgood | except the IP address for the proxy/sip registration is the same |
| 05:15.35 | drmessano | Which is irrelevent |
| 05:15.41 | sawgood | excellent then! |
| 05:16.12 | sawgood | Their side must not like two registrations from the same IP then |
| 05:17.43 | sawgood | changing gears ... can you confirm something for me .... |
| 05:17.50 | CaneToad | sawgood...my scenario is if I config the ATA to register directly with the provider, then everything works including caller id and distinctive ring on the phone, but if I config the ATA to register with my asterisk which in turn registers with the provider, then the distinctive ring on the phone doesn't work and for some reason the phone alternates between displaying the caller id and the extension number |
| 05:18.49 | sawgood | Well, the ATA (when registered to the * box) will have no idea at all about the ring tone and/or caller ID ... unless your * box gives it that information |
| 05:19.18 | CaneToad | the phone handset itself has settings to ring differently when different numbers call (caller id) |
| 05:19.18 | sawgood | You should make an inbound call to the DID (why watching the call from the * CLI) to see what comes into |
| 05:20.21 | sawgood | drmessano: If have a 'single' one line exten => statement in a context ... then it seems to me that none of my includes are ever processed |
| 05:20.45 | sawgood | Is it possible to have includes 'work' if you have exten => entires in a context? |
| 05:21.23 | drmessano | Of course... you're supposed to have everything in contexts |
| 05:23.06 | CaneToad | sawgood, yup, this is what happens for an incoming call: |
| 05:23.06 | sawgood | Well, if I have a context, and the context has include = statements ... they are never 'used' IF inside the context I have even as few as one exten => lines |
| 05:23.06 | CaneToad | 1. when registering directly with the ata, when the phone is ringing, the handset displays the caller id consistently, and its distinctive ring feature based on the caller id of the caller works |
| 05:23.06 | CaneToad | 2. when registering with asterisk, when the phone is ringing, the handset displays the caller id for a second and then displays the extension number for a second and its distinctive ring doesn't work |
| 05:23.47 | CaneToad | sawgood I might snoop the traffic to see what is different |
| 05:23.55 | sawgood | So, why have the ATA go through ASterisk if you have a solution which works? |
| 05:24.26 | CaneToad | because I have multiple extensions that need to call out via the same registration |
| 05:25.01 | CaneToad | calling out that way is working fine |
| 05:25.01 | sawgood | What comes in from your ITSP ... probably will not be any different ... its what Asterisk does with the SIP messaging here which is handed off differently IMHO |
| 05:25.33 | sawgood | Is the ATA supporting a FAX by chance (full time) |
| 05:25.38 | CaneToad | no |
| 05:26.04 | sawgood | Are you using RAW Asterisk 1.4.x or 1.6.x |
| 05:26.21 | CaneToad | 1.6.x |
| 05:26.50 | sawgood | So in extensions.conf ... do you have entries for each DID from your provider? |
| 05:27.41 | CaneToad | yes |
| 05:27.53 | drmessano | It sounds like you're setting the callerID in the dialplan incorrectly, and you need to ring the extension with the correct distinctive ring |
| 05:28.00 | sawgood | This is a SIP trunk coming in, right? |
| 05:28.01 | drmessano | It doesn't just pass through like a switch |
| 05:28.36 | sawgood | This URL might help http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice+distinctive+ring+support |
| 05:30.20 | CaneToad | when I say "distinctive ring", the ring functionality I have here isn't in sip protocol, it is based purely on the phone's ability to be configured to ring differently based on the caller id |
| 05:30.37 | sawgood | neat feature! |
| 05:30.47 | sawgood | no wonder they want to keep an analog phone |
| 05:30.53 | sawgood | who makes that phone? |
| 05:31.11 | sawgood | I can get my Grandstream phones to do that (but they are SIP units) |
| 05:31.54 | sawgood | Buzz ... Buzz ... Buzz (vs) Ring ... Ring ... Ring |
| 05:31.59 | CaneToad | it is an analog phone attached to a linksys ATA |
| 05:32.01 | sawgood | girlfriend (or) wife! |
| 05:32.59 | CaneToad | the analog phone is http://www.productreview.com.au/p/telstra-5200-1.html |
| 05:32.59 | sawgood | Once I had a stalking girlfriend whom I had to set a silent ring tone to ... so I was not bothered by her calls |
| 05:33.16 | kaldemar | sawgood: the includes do work, but extensions are matched before included ones. if you have a matching extension in a context, included ones don't matter. |
| 05:33.56 | sawgood | but 'what if' ... you are using exten => only to SET a variable, but want the includes to continue to match for call routing? |
| 05:34.25 | sawgood | I can give you a direct example if you would like? |
| 05:35.19 | sawgood | pretty nice cordless phone |
| 05:36.50 | kaldemar | sawgood: if an exten with priority 1 matches, others wont match unless you use some goto. |
| 05:37.23 | sawgood | do you have to use goto with exten => ... |
| 05:37.32 | sawgood | oh wait ... what about in the general section |
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| 05:37.42 | kaldemar | no |
| 05:37.44 | sawgood | can I set a global variable there to match all incoming calls? |
| 05:37.54 | kaldemar | no. |
| 05:38.23 | sawgood | See, I have one incoming context with say 30 include statements (and growing) |
| 05:38.41 | sawgood | I do not want to have to put the exten => in 30 includes ... |
| 05:38.55 | sawgood | I have it there now, but I would like to trim down the file if possible |
| 05:38.55 | kaldemar | set the var in some context and then use goto to jump into another. |
| 05:39.13 | sawgood | ok ... let me picture this ... |
| 05:39.39 | sawgood | oh ... from the 'jump' ... I would have the 30 includes |
| 05:39.43 | sawgood | got it! |
| 05:40.06 | sawgood | 1st context = set variable and go to 2nd context = just the includes |
| 05:40.16 | kaldemar | yes. |
| 05:40.28 | sawgood | So, is the goto a function or an application? |
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| 05:40.43 | kaldemar | application. |
| 05:40.57 | sawgood | I'm going to look it ip now ... thank you! |
| 05:42.09 | WIMPy | Goto is good if you're on SIP or the like, but it breaks overlap dialing. |
| 05:42.26 | WIMPy | I'm still searching for a solution to that. |
| 05:42.33 | sawgood | since I do not know what overlap dialing is ... I guess I do not have to worry about that! |
| 05:43.07 | WIMPy | It's what you do with classic phones, I.e. POTS or ISDN. |
| 05:43.19 | WIMPy | But some SIP phones ca also do it. |
| 05:43.26 | WIMPy | can |
| 05:43.50 | kaldemar | and you can disable it in sip.conf. |
| 05:44.21 | WIMPy | Yes, I'm not sure wat that does in SIP however. |
| 05:45.19 | sawgood | Goto(context-in-include-file) ... do I have to actually USE the word (context- |
| 05:45.32 | WIMPy | It seems to be emulated entirely in the phone. |
| 05:46.00 | kaldemar | sawgood: where did you dig that up? |
| 05:46.09 | sawgood | VoIP Info? |
| 05:46.47 | kaldemar | core show application Goto will give you correct syntax. |
| 05:46.53 | sawgood | perfect! |
| 05:47.02 | kaldemar | that one is incorrect. |
| 05:48.05 | sawgood | Goto([[context,]extensions,]priority) .... does this mean to put the context inside of [] brackets litterly? |
| 05:48.27 | kaldemar | no, it means it's optional. |
| 05:48.59 | sawgood | So ... Goto(sawgood),2 |
| 05:49.02 | kaldemar | if there's no given context, the current one is used. same with extension. |
| 05:49.24 | sawgood | neat .. just use the goto application then |
| 05:49.25 | kaldemar | still wrong. |
| 05:49.46 | sawgood | Goto(sawgood,2) |
| 05:50.09 | kaldemar | that goest to exten sawgood, priority 2. |
| 05:50.25 | sawgood | thank you! |
| 05:50.33 | sawgood | oh extension sawgood |
| 05:50.36 | sawgood | wait then |
| 05:51.22 | kaldemar | you need 3 args to go to another context. |
| 05:51.35 | sawgood | Goto([sawgood],2) |
| 05:52.06 | sawgood | Technially, following your lead and the syntax it looks like |
| 05:52.24 | sawgood | Goto([sawgood,]2) |
| 05:53.04 | kaldemar | yes, but you ignore the common way to implicate optional input with []. |
| 05:53.44 | kaldemar | goto(context,extension,priority) |
| 05:53.45 | sawgood | Goto([[sawgood,]2) |
| 05:54.32 | sawgood | or Goto([[sawgood],2) |
| 05:55.23 | sawgood | I'll put them in extensions.conf to see which one works! |
| 05:55.37 | kaldemar | mone of them will. |
| 05:55.41 | kaldemar | none |
| 05:55.42 | sawgood | what! |
| 05:55.49 | sawgood | Goto(sawgood,2) |
| 05:56.23 | WIMPy | That will work, but probably not do what you want. |
| 05:56.25 | kaldemar | as you expect, that is. |
| 05:58.37 | sawgood | I see all types of 'wrong examples' for goto on the net |
| 05:59.07 | WIMPy | The net is evil. |
| 05:59.35 | kaldemar | you better use proper documentation. |
| 06:00.49 | sawgood | So, if you must include both a priority and extension in the goto (what if your include) has to exten to reference in the goto statement? |
| 06:01.36 | WIMPy | cannot parse that question. |
| 06:01.48 | kaldemar | you lost me there, too |
| 06:01.54 | sawgood | If no specific extension, or extension and context, are specified, then this application will just set the specified priority of the current extension. |
| 06:02.12 | WIMPy | correct |
| 06:02.29 | sawgood | So, if I use goto (I need to supply both a context and an extension to goto) |
| 06:02.57 | sawgood | What if the goto is to another context which has NO exten statements? |
| 06:03.24 | WIMPy | And a priority as well. |
| 06:03.33 | kaldemar | if it has includes, it doesn't need exten lines. |
| 06:03.42 | WIMPy | What does it have, if no extens? |
| 06:03.57 | sawgood | the context only has includes inside of it |
| 06:04.23 | WIMPy | That is as if the included extens were in that context. |
| 06:04.47 | WIMPy | That's the point of includes. |
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| 06:05.22 | sawgood | ok ... so inside the various include statement are many different exten => statements ... |
| 06:05.29 | sawgood | I'm not sure which one to pick ... |
| 06:06.11 | WIMPy | The one where you want to continue. |
| 06:06.25 | WIMPy | And that's probably ${EXTEN}. |
| 06:06.51 | sawgood | The exten => statement are to match various incoming DID numbers, so I never know what DID is being called? |
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| 06:07.18 | WIMPy | s.a. |
| 06:07.27 | sawgood | I'm probably reading too much into it |
| 06:07.35 | sawgood | I probably should just practice until it works |
| 06:07.41 | kaldemar | or not enough |
| 06:08.25 | kaldemar | at this point you better try it and show what you have if it doesn't work. |
| 06:08.31 | sawgood | ok |
| 06:12.04 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 06:20.38 | sawgood | exten => _X.,2,Goto(new-context,${EXTEN},3) |
| 06:20.42 | sawgood | what is wrong with this? |
| 06:21.14 | WIMPy | The priority. |
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| 06:21.39 | sawgood | can you tell me why I should not make it 3 (since I have a 1 and a 2 already) |
| 06:22.02 | sawgood | or is this a different type of priority? |
| 06:22.48 | WIMPy | No, but a different extension you Goto. |
| 06:23.02 | sawgood | So, what priority would you suggest? |
| 06:23.07 | WIMPy | So unless you want to skip the first two priorities of your target extension... |
| 06:23.24 | sawgood | oh ... |
| 06:23.26 | sawgood | I get it! |
| 06:23.38 | sawgood | I want it to START at priority 1 when it arrives! |
| 06:23.53 | WIMPy | That's it. |
| 06:25.19 | WIMPy | Actually you could also just have your set() at priority 1 and change all your existing extensions to start at priority 2. Then you don't need a goto. |
| 06:25.22 | sawgood | Got that part, so the process is failing because, "rejected because extension not found" |
| 06:25.48 | WIMPy | But I'll leave you alone with that and try to take a nap. |
| 06:34.20 | sawgood | Ok ... here is what I have ... call arrives to new context ... the NoOp command I want is set at priority 1, the Goto works (sending the call to the next context) at priority 2 .... the call is then failing because there is no extenions in the next context (it only has include statements) |
| 06:36.54 | sawgood | oh ... I see ... got it! |
| 06:37.09 | p3nguin | If you Goto() something that exists, it will work. |
| 06:37.23 | sawgood | problem is I AM going to something which exists! |
| 06:37.42 | sawgood | I am going to a valid context which has only include statements and NO exten => statements |
| 06:39.52 | sawgood | I want the call to be processed by which include = which has the matching extension statement |
| 06:42.42 | p3nguin | Yep, it works just as I predicted. |
| 06:42.51 | p3nguin | [phones] |
| 06:42.51 | p3nguin | exten => 555,1,Goto(testing1,234,1) |
| 06:43.00 | p3nguin | [testing1] |
| 06:43.00 | p3nguin | include => testing2 |
| 06:43.05 | p3nguin | [testing2] |
| 06:43.05 | p3nguin | exten => 234,1,Verbose(234@testing2) |
| 06:43.21 | p3nguin | My phone has a context of "phones". |
| 06:43.42 | sawgood | I see that, but you KNEW specifically what exten was being called (I do not) |
| 06:43.59 | sawgood | I have a list of 30+ DID numbers (I cannot predict) what DID is called |
| 06:44.43 | p3nguin | I'll repeat what I said before: If you Goto() something that exists, it will work. |
| 06:45.14 | sawgood | sure ... you are giving me the right answer (but not helping to fix the problem) |
| 06:45.26 | p3nguin | I don't see any problem. That's the thing. |
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| 06:55.03 | sawgood | http://pastebin.com/SZUkTXhE |
| 06:55.14 | sawgood | This sort of is a 3 line summary of what I am facing |
| 06:56.28 | p3nguin | If there is no matching extension in the context where you decided to send the call, how can you expect anything good to happen? |
| 06:56.55 | sawgood | exactly! ... so how to use the Goto command to look into the includes for the exten match? |
| 06:57.39 | p3nguin | If a matching extension exists, it will run. I've already covered this twice and even given you an example because you insisted I was wrong. |
| 06:58.31 | sawgood | <PROTECTED> |
| 06:58.39 | p3nguin | It doesn't make any difference. |
| 06:58.40 | sawgood | my example needs to be a variable |
| 06:58.44 | sawgood | oh ... ok ... |
| 06:59.09 | sawgood | Do you think the syntax I have is correct? |
| 06:59.36 | p3nguin | You've only shown me ONE of three contexts. How do you expect me to determine if your syntax is correct? |
| 07:00.07 | sawgood | assume I have a exten => statement inside of the next-incoming context |
| 07:00.19 | p3nguin | You said it only has includes. |
| 07:00.35 | sawgood | it does ... inside one of the includes is the exact exten match I need |
| 07:00.44 | p3nguin | If that is the case, it will work. |
| 07:01.13 | sawgood | change your 234 statement to ${EXTEN} and confirm it works! |
| 07:01.17 | p3nguin | Including a context which does have a matching extension means that the context that has the include actually has the matching extension. |
| 07:01.34 | sawgood | I have learned that over the past few days (many wasted hours) |
| 07:02.39 | p3nguin | exten => 555,1,Goto(testing1,${EXTEN},1) |
| 07:02.45 | sawgood | yes ... |
| 07:02.46 | p3nguin | [testing1] |
| 07:02.46 | p3nguin | include => testing2 |
| 07:02.51 | p3nguin | [testing2] |
| 07:02.52 | p3nguin | exten => _X.,1,Verbose(${EXTEN}@testing2) |
| 07:02.59 | sawgood | I think that is what I have |
| 07:03.14 | p3nguin | <PROTECTED> |
| 07:03.19 | p3nguin | 555@testing2 |
| 07:03.28 | p3nguin | It works, just like I said it would. |
| 07:03.34 | sawgood | You did say that |
| 07:03.42 | sawgood | I am testing my end ... brb |
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| 07:04.21 | sawgood | it works! |
| 07:04.45 | sawgood | my priority was OFF |
| 07:04.51 | p3nguin | Dial plan logic isn't really that hard. And for me, unless I'm messed up on drugs, I usually have pretty good confidence in my work. |
| 07:04.55 | sawgood | I had no priority 1 in the include statements |
| 07:05.05 | sawgood | I only have priority 2 and higher |
| 07:05.13 | sawgood | I am very sorry for the bone head mistake |
| 07:05.36 | p3nguin | Extensions have to start with 1, always, and the rest should typically be n unless you have a very specific need for numbering them. |
| 07:05.58 | p3nguin | In most cases, you don't have a need to number them beyond 1. |
| 07:05.59 | sawgood | I know ... but I had to ; out priority 1 to confirm there was not false positive |
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| 07:08.25 | p3nguin | Even when I have a need for numbering my priorities, I usually still don't. I use 1, n, and s priorities with great success. |
| 07:09.02 | sawgood | really ... I was having concern, but I'm re-doing them now |
| 07:13.49 | sawgood | all is well now with 1 and followed by n |
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| 07:16.37 | sawgood | p3nguin: thank you ... I trimed extensions.conf by over 30 lines with Goto application! |
| 07:16.41 | sawgood | Thank you kaldemar |
| 07:16.44 | sawgood | Thanks WIMPy |
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| 07:40.11 | Defraz | Just curious if anyone might have some ideas why my Grandstream HT502 ATAs disconnect the call of the person that is on hold after the caller does a hook flash durring a call waiting. |
| 07:40.48 | Defraz | So the person gets a beep and they switch over and start talking to the new caller then the second call has exactly 30 seconds then they get hug up on. |
| 07:40.53 | Defraz | hung up on. |
| 07:41.31 | Defraz | I try a PAP2T and it works just fine seems to be a problem with the grandstream HT502s |
| 07:42.14 | p3nguin | ~grandstream |
| 07:42.14 | infobot | somebody said grandstream was the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 07:42.31 | sawgood | What about ... SIP invite ...under advanced settings |
| 07:42.43 | sawgood | I think it might be labeled RFC 3261 timer |
| 07:42.51 | sawgood | make sure it is set to 180 seconds |
| 07:43.43 | Defraz | okay |
| 07:43.48 | Defraz | thanks . |
| 07:44.10 | Defraz | I can't seem to find anything. Been looking and trying things for days. |
| 07:44.45 | sawgood | I had a similar concern with the GXW4004 before ... and that was the fix (but that is a 4 port FXS gateway) |
| 07:44.52 | sawgood | Yours is a 2 port version |
| 07:44.58 | Defraz | yea |
| 07:45.01 | sawgood | It does T.38, so that is why I like it |
| 07:45.07 | Defraz | and I seem to have it with a new GPX 2110 |
| 07:45.15 | sawgood | nice phone |
| 07:45.46 | Defraz | but it does the same thing after 2 minutes |
| 07:45.55 | Defraz | the gpx2000 doesn't |
| 07:46.22 | sawgood | Do you have the newer 1.x firmware on the phone? |
| 07:46.29 | Defraz | yea the latest |
| 07:46.29 | sawgood | or the older style firmware |
| 07:46.34 | Defraz | newer |
| 07:47.59 | Defraz | hmmm I don't see those options on my HT502 |
| 07:48.26 | Defraz | just a force Invite |
| 07:48.39 | sawgood | look for something that is a fill in type box for 'timer' |
| 07:51.11 | Defraz | hmmm |
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| 08:11.46 | sawgood | So, is this a fair and correct statement ... When something matches an exten => statement at priority 1 (at least these two things happen): no more priority 1 statements are used in the current context and no include statements are automatically parsed |
| 08:12.42 | sawgood | In fact, I know this to be the case from my scratch box testing ... |
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| 08:15.03 | Polysics | hello |
| 08:15.23 | Polysics | i am still trying to find out how an applet i know works does an IAX2 call over an HTTP proxy :-) |
| 08:15.27 | sawgood | wow ... I'm like a real Asterisk dialplan apprentice now! |
| 08:15.48 | Polysics | afaik the proxy is HTTP, non SOCK5 |
| 08:16.51 | ChannelZ | sawgood: well it's kind of a different way to put it, but once an extension matches it starts executing that exten at priority 1 and continues on. Whether that came of an extension directly in the context or one that was included from that context. |
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| 08:17.37 | sawgood | thank you |
| 08:17.38 | ChannelZ | but you can of course Goto and jump around so "no more priority 1 statements are used" isn't necessarily true |
| 08:18.09 | ChannelZ | and if you were to WaitExten or Background and someone dialed another extension, that could case it to start executing priority 1 of a new exten |
| 08:18.10 | ChannelZ | etc |
| 08:18.14 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:18.22 | ChannelZ | s/case/cause/ |
| 08:18.24 | sawgood | I see ... that is why I am the apprentice |
| 08:18.40 | ChannelZ | farts on infobot |
| 08:20.41 | ChannelZ | Polysics: IAX2 is UDP not TCP first of all so I'm not exactly sure what you're talking about |
| 08:24.35 | Polysics | ChannelZ: i am not sure either, all I know is this Java applet DOES connect and place calls on a network that has ONLY an http/https proxy as its outlet |
| 08:24.44 | Polysics | no open ports, nothing |
| 08:24.50 | *** part/#asterisk X-Raimo (~asmpub@opensuse/member/raimoschmidt) |
| 08:24.53 | Polysics | i am simply trying to figure out how that does it |
| 08:25.33 | drmessano | Sure it's not using UPNP? |
| 08:26.40 | Polysics | UPNP? now that's something i have never heard about |
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| 08:29.10 | Polysics | how are teh two things related? |
| 08:29.19 | sawgood | Verbose([<level>|]<message>) |
| 08:29.43 | sawgood | Technically, in the above application statement ... what do the [] statements do? |
| 08:29.48 | drmessano | Polysics, you should read up on UPnP |
| 08:30.45 | sawgood | Are the <> mandantory for this command to work? |
| 08:30.58 | drmessano | Polysics, some applications use UPnP to open the firewall, such as Hamachi and Skype |
| 08:31.54 | drmessano | Polysics, your assertion that all this magic is going to go over a simple HTTP proxy seems to be getting you nowhere.. Perhaps some broader thinking is involved |
| 08:31.57 | Polysics | drmessano: that looks like something interesting. i will add that info to what i can gather by sniffing the network on the 28th |
| 08:32.40 | Polysics | i have looked at encapsulating UDP over TCP/HTTP, but it looks like a nightmare for voice, so i would say that route isn't getting me anywhere, yes |
| 08:52.54 | Polysics | i am simply not good enough at networking for this, i think :-) |
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| 08:57.51 | joobie | Polysics, just do a tcpdump when making a call wiht that java app |
| 08:58.18 | Polysics | i usually use wireshark, is tcpdump better? |
| 08:58.45 | joobie | yes |
| 08:58.50 | joobie | but wireshark will achieve similar |
| 08:59.12 | joobie | when you trigger the traffic via the java app,what do you see in wireshark? |
| 09:02.22 | Polysics | i will have access to a closed off network where the applet works on the 28th |
| 09:02.32 | Polysics | and am currently documenting myself as much as i can beforehand |
| 09:02.45 | Polysics | doing it without a proxy results in a normal IAX2 call |
| 09:03.08 | drmessano | ok, so then what? |
| 09:03.40 | Polysics | drmessano: in what sense what? :-) |
| 09:03.50 | drmessano | If you do figure out how this one web based app is making calls, how do you then use this information to fix your issue? |
| 09:04.41 | joobie | im not following the problem.. |
| 09:04.56 | joobie | it's pretty straight forward how a java app would tunnel iax2 over http proxy |
| 09:05.03 | Polysics | i am not fixing anything, i would like to implement the same thing |
| 09:05.16 | drmessano | Implement it with Asterisk or using this same app? |
| 09:05.19 | Polysics | joobie: how would it then? that's what i would like to know :-) |
| 09:05.19 | joobie | you want to tunnel iax2 over a http proxy/ |
| 09:05.52 | Polysics | drmessano: asterisk is the server, i would like to implement another applet that does teh same thing but with an API i can use and control |
| 09:06.02 | joobie | why do you want to do this? |
| 09:06.08 | Polysics | i probably don't have the skills, but it could be a great learning project |
| 09:06.52 | Polysics | joobie: government organizations have heavily firewalled and unchangeable environments, the company that wants me to investigate on this intends to provide specialized voice services to them |
| 09:07.20 | Polysics | and apparently it is important to have a zero-install environment such as a browser-based java applet |
| 09:07.25 | joobie | the company that wants you to investigate sounds retarded |
| 09:07.36 | joobie | if they are providing voip services for this gov |
| 09:07.44 | joobie | im sure the gov would plug a hole in their firewall |
| 09:07.52 | Polysics | joobie: not in italy |
| 09:07.58 | joobie | rather than knowinly agree to misusing their http proxy |
| 09:08.08 | joobie | .. and put voip traffic onto their http proxy |
| 09:08.11 | Polysics | the land of useless IT laws and feudal sysadmins |
| 09:08.16 | drmessano | bingo |
| 09:08.52 | drmessano | You don't think IT is going to plug the hole you create when they see all this voice traffic going across their http proxy? |
| 09:08.58 | Polysics | anyway, this is not the problem, i cannot circumvent that. i would not even be investigating if i didn't see this applet an italian voip provider uses that DOES work in that environment |
| 09:09.18 | joobie | Polysics, it's retarded what you are trying to do.. |
| 09:09.19 | joobie | but |
| 09:09.25 | Polysics | from waht i gather, they will not plug hoels in the fw, but the will agree to let the traffic through |
| 09:09.26 | joobie | in java, just create a http session |
| 09:09.32 | drmessano | LOL |
| 09:09.44 | Polysics | i never said i deal with smart people anytime before :-D |
| 09:09.52 | joobie | and take your udp payload and encode it to some relevant http content-type |
| 09:10.09 | joobie | then push each udp packet encoded in the content type over the http proxy |
| 09:10.15 | joobie | on the other end, reverse the process |
| 09:10.18 | Polysics | anyway, since they do pay me for this sort of r&d work ,i am happy even if it gets nowhere, happier if i CAN find out :-) |
| 09:10.25 | joobie | once again, it's very gay what you're trying to do |
| 09:10.49 | Polysics | joobie: it's not certain that the traffic actually goes through the proxy |
| 09:10.52 | joobie | you could also setup a SSL tunnel |
| 09:11.07 | joobie | and tunnel throught this |
| 09:11.09 | joobie | -t |
| 09:11.27 | joobie | traffic would pass |
| 09:11.28 | Polysics | from material i've read, http tunneling voice should all but kill the call quality |
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| 09:11.53 | joobie | well, somewhat |
| 09:11.58 | joobie | people run voip over vpn |
| 09:12.03 | joobie | which is similar |
| 09:12.04 | *** part/#asterisk UQlev (~yuriy@212.50.99.8) |
| 09:12.12 | joobie | it just means you are ordering your packets |
| 09:12.42 | joobie | so you're more lightly to have audible glitches |
| 09:13.18 | joobie | but i hear people often running voip over vpn, which is similar |
| 09:13.44 | Polysics | this should be a fallback mechanism, if people are unhappy with the results they will complain to their own admins |
| 09:14.01 | joobie | i just think if you have to go to the effort of tunneling voip over their http proxy purely because they are not comfortable opening a port - something is seriously wrong .. |
| 09:14.34 | drmessano | Sounds criminal |
| 09:14.35 | Polysics | the real hassle will be that i have to switch to IAX2 from sip anyway to have a single port open |
| 09:14.48 | drmessano | I've alerted interpol of a possible security breach |
| 09:15.04 | Polysics | you are both very right - but the issue at hand is not "CSI sysadmins are idiots", also because i do know that already :-) |
| 09:15.25 | drmessano | Has nothing to do with sysadmins being idiots |
| 09:16.19 | Polysics | in this case it does, trust me |
| 09:16.25 | Polysics | i would not be doing this otherwise |
| 09:16.28 | joobie | someone is an idiot |
| 09:16.33 | Polysics | we DID ask about having the ports opened |
| 09:16.41 | drmessano | Has to do with the process being self-defeating. You're suggesting an entity is requesting services they refuse to support on their network, that they would then effort to allow you to circumvent the support roadblocks? |
| 09:16.57 | Polysics | i suppose it is just a case of "policy" |
| 09:17.28 | joobie | maybe you should address this with them |
| 09:17.35 | joobie | to find out why they are hesitant to open up the ports |
| 09:17.47 | joobie | and then squash their chain of thought.. |
| 09:17.52 | drmessano | "We want 1000 cheeseburgers. No you may not deliver them, but there is this keyhole" |
| 09:17.54 | drmessano | .... |
| 09:18.02 | joobie | heh |
| 09:18.35 | drmessano | Sorry, this just sounds suspicious |
| 09:18.43 | joobie | they are paying a business man to stand by the door and squeeze those 1000 cheeseburgers through the keyhole.. |
| 09:19.27 | joobie | Polysics, is this a http proxy they have btw |
| 09:19.31 | joobie | or is it just "port 80 is open" |
| 09:19.54 | joobie | .. and what about port 443, is this proxied too? |
| 09:19.56 | Polysics | joobie: nope, it is a proxy, telnetting on 80 doesn't get anywhere |
| 09:20.39 | Polysics | didn't test 443 but there is no reason it should be different, but i will write this down |
| 09:21.07 | joobie | there is reason it could be different |
| 09:21.39 | joobie | not everyone has an ssl proxy |
| 09:21.59 | joobie | and they already sound like noobs with policies like that |
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| 09:22.10 | joobie | it takes a little more knowledge to deploy an ssl proxy |
| 09:22.26 | joobie | and they sound short on that.. |
| 09:22.27 | joobie | :) |
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| 12:54.04 | WIMPy | Works for me. |
| 12:54.19 | krion | deadlock deadlock deadlock |
| 12:54.26 | krion | WIMPy: works now for me too thanks |
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| 14:24.09 | b_d | hi all |
| 14:24.50 | b_d | I have a quick question regarding agi/macrtos |
| 14:24.53 | b_d | macros * |
| 14:26.20 | b_d | I currently have an agi/ivr script which uses cmd dial to initiate a call, dial uses the "M" method to fire up a macro which plays a message for the callee to hear, the question is, do I set the absolute call duration/timeout in the agi script or the macro that is called for the callee? |
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| 14:28.16 | b_d | the reason I ask is that $agi->exec_absolutetimeout() doesnt seem to be effective, unless I'm implementing it incorrectly |
| 14:29.24 | b_d | can someone please advise? |
| 14:30.59 | WIMPy | Use Set(TIMEOUT(absolute)=...) |
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| 14:32.34 | _Corey_ | In the context of AGI, you might do ->exec("SET TIMEOUT(absolute)=x") |
| 14:33.19 | WIMPy | There is a setvar or something, isn't there? |
| 14:33.29 | _Corey_ | I think that's deprecated |
| 14:33.38 | _Corey_ | not sure though |
| 14:33.49 | _Corey_ | I do it that way in my AGIs |
| 14:34.19 | WIMPy | I haven't looked into teh AGI stuff for ages. |
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| 14:53.39 | b_d | thanks guys |
| 14:53.41 | b_d | I will test it out |
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| 17:03.09 | atan | I have two clients connected but one does not transmit audio back to the other |
| 17:03.29 | atan | NAT issue? Something I might change on the server? |
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| 17:11.09 | p3nguin | ~sipnat |
| 17:11.09 | infobot | [~sipnat] Quick guide on configuring * + 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 |
| 17:11.22 | p3nguin | atan: This ^^^^ |
| 17:12.07 | atan | reading, thanks |
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| 17:36.18 | atan | Crap. Still no audio. |
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| 18:08.13 | atan | We're good now! My bad. He had no nat. I did. hah! |
| 18:08.19 | atan | Ciao guys. Happy holidays! |
| 18:08.25 | root52 | Something new today. Trying to use the register utility to register the digium g.729 lic I just pick up. However this time I am running on a VM and I get the message "Make sure that you have eth0 enabled." well I am using vmnet0:1 as my nic and I could not find a option to point to something different than eth0. Any thoughts? |
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| 20:56.29 | Mango | What could cause logging not to work? The output of logger show channels is: |
| 20:56.36 | Mango | <PROTECTED> |
| 20:56.40 | Mango | yet, that file is completely empty. |
| 20:56.47 | Mango | Verbose is 3 as usual. |
| 20:56.57 | Mango | and that file is CHMODed to 777. |
| 20:57.22 | Mango | ah. |
| 20:57.26 | Mango | it needed a restart after the chmod |
| 20:57.28 | Mango | ignore me. |
| 21:12.55 | raden | <PROTECTED> |
| 21:13.10 | drmessano | ccccombo breaker |
| 21:15.02 | raden | how it going drmessano ? |
| 21:15.08 | raden | u really a doctor / |
| 21:15.09 | raden | ? |
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| 21:39.47 | drmessano | Its going |
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| 23:43.21 | path | uuuhbvygy6ttt}++¿¿'¿ |
| 23:43.29 | path | ¿¿'''00{ñññppp |