| 00:26.24 | *** join/#asterisk themalik (~themalik@c-98-235-212-140.hsd1.pa.comcast.net) |
| 00:40.16 | themalik | im having trouble getting a softphone going in elastix |
| 00:40.26 | themalik | i keep getting registration failures |
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| 01:00.24 | themalik | i keep getting 503 service unavailable |
| 01:01.05 | themalik | ! |
| 01:01.10 | themalik | !context |
| 01:01.14 | themalik | darn...no factoid |
| 01:03.30 | p3nguin | What are you trying to find out? |
| 01:03.49 | p3nguin | The definition of context? |
| 01:04.56 | themalik | im trying to figure out why i cant have my softphone register on elastix |
| 01:05.05 | themalik | i figured context might be an issue... |
| 01:05.20 | themalik | its set to from-internal |
| 01:05.35 | p3nguin | The context setting of a peer indicates where a call enters the dial plan. |
| 01:06.21 | themalik | so it doesnt explian why i cant register |
| 01:06.33 | p3nguin | That's right. |
| 01:07.16 | themalik | what can give a service unavailable error? |
| 01:10.58 | p3nguin | Which soft phone are you using? |
| 01:11.13 | themalik | i tried xlite, and sflphone |
| 01:14.59 | p3nguin | Since I can't support Elastix, all I can recommend is verifying all the settings for the phone. Make sure the registrar, domain, user name, and password are all correct. |
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| 01:15.30 | themalik | my domain might be the issue |
| 01:15.58 | themalik | can i use the external ip of my network if i forward port 5060 to my internal ip with the server |
| 01:16.27 | p3nguin | If you are on the same LAN, why would you want to? |
| 01:16.40 | themalik | im not |
| 01:16.45 | themalik | im trying to use it outside my lan as well |
| 01:16.50 | themalik | i cant get register inside or out |
| 01:16.58 | p3nguin | Oh, then yes, you need to use the outside address. |
| 01:17.16 | themalik | could i use a dyndns |
| 01:17.21 | p3nguin | sure |
| 01:18.16 | themalik | lets hope it works..i know its not elastix thats affecting it, its just the asterisk part im working with |
| 01:22.03 | themalik | dang, still didnt work |
| 01:22.26 | themalik | hmmm...i have host as dynamic, and deny and permit are set to 0.0.0.0/0.0.0.0 |
| 01:23.20 | themalik | whats accountcode? |
| 01:23.46 | p3nguin | It's the account code that your phone uses for CDRs. |
| 01:23.56 | themalik | by default its blank right |
| 01:24.11 | p3nguin | Maybe. It doesn't affect registrations, though. |
| 01:24.20 | themalik | maybe my qualify setting shouldnt be set to yes |
| 01:25.05 | p3nguin | It most likely should be set to yes if your phone or asterisk is behind NAT. |
| 01:25.21 | themalik | yeah its behind my router nat |
| 01:26.14 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 01:26.19 | themalik | i also changed host to dynamic, isnt default internal or something |
| 01:26.43 | p3nguin | host=dynamic is used for phones which send registration. |
| 01:26.54 | themalik | so i need that setting |
| 01:27.00 | themalik | and my nat setting is yes... |
| 01:27.04 | themalik | why wouldnt it work |
| 01:27.08 | themalik | everything seems fine |
| 01:27.19 | p3nguin | If the phone does not send registration, you have to set the host value to the IP address of the phone. |
| 01:27.59 | themalik | it sends registration, just fails every time |
| 01:28.47 | p3nguin | Is there any reason you don't want to paste your sip.conf in full, masking only passwords, so I can see what you've got going on? |
| 01:29.06 | themalik | nope, not at all ill paste it |
| 01:29.11 | themalik | id appreciate that |
| 01:31.15 | themalik | what directory is it under? |
| 01:31.25 | p3nguin | /etc/asterisk/ |
| 01:31.48 | *** join/#asterisk shido6 (~shido6@c-24-130-58-117.hsd1.ca.comcast.net) |
| 01:31.48 | p3nguin | (I'm only offering help with Asterisk, not Elastix) |
| 01:32.00 | themalik | yeah i know, my issue is only with asterisk |
| 01:32.04 | themalik | they said to ask in this channel |
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| 01:34.40 | themalik | p3nguin: pastebin.com is down |
| 01:34.47 | themalik | what alternative do u prefer |
| 01:34.58 | p3nguin | pastebin.com is fine. |
| 01:35.00 | themalik | nvm |
| 01:35.02 | themalik | it works |
| 01:35.10 | themalik | it wouldnt open last time i checked sorry |
| 01:36.16 | themalik | sip.conf? |
| 01:36.39 | SpaceJockey | hey, any hints on what I need to look at if I keep missing the first word of prompts? example: I call vm extension and I hear "...en mail" |
| 01:37.16 | SpaceJockey | is that an early media issue? |
| 01:37.38 | themalik | p3nguin: umm...there doesnt appear to be any passwords i must mask |
| 01:37.44 | themalik | in sip.conf |
| 01:38.26 | shido6 | add a wait period before playing the file or add 3 seconds of silence some other way |
| 01:38.30 | p3nguin | I'd probably add a Wait(1) or Wait(2) in the extension before the VoiceMailMain() command. |
| 01:38.40 | shido6 | nods |
| 01:39.20 | SpaceJockey | thanks |
| 01:40.02 | p3nguin | themalik: Perhaps Elastix doesn't store its peer definitions in sip.conf. Check for included files. |
| 01:41.57 | themalik | http://pastebin.com/witMchHk |
| 01:42.02 | themalik | those are all files in asterisk |
| 01:42.05 | themalik | folder |
| 01:42.11 | p3nguin | directory? |
| 01:42.31 | p3nguin | I don't do Windows, so I hope you mean directory. |
| 01:43.15 | themalik | lol thats what i meant |
| 01:44.07 | p3nguin | No clue. I use Asterisk, and I don't have a quarter of these files. |
| 01:45.40 | themalik | http://pastebin.com/f3GH8upz |
| 01:46.12 | p3nguin | I see that this is a FreePBX box. |
| 01:46.13 | themalik | is my sip.conf, but i doubt its the right one |
| 01:46.23 | p3nguin | ~freepbx |
| 01:46.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 01:46.56 | themalik | is freepbx the asterisk in elastix |
| 01:47.10 | p3nguin | No, FreePBX is a GUI. |
| 01:47.31 | themalik | oh, yeah then i think its freepbx, elastix has a webgui for asterisk |
| 01:47.34 | themalik | my bad |
| 01:47.35 | themalik | sorry |
| 01:47.50 | p3nguin | The GUI controls all aspects of Asterisk's configuration via scripts and stuff that are not part of Asterisk. |
| 01:48.06 | p3nguin | And because of that, we don't support FreePBX here. |
| 01:48.07 | themalik | oh...that makes sense |
| 01:48.18 | themalik | sorry, once again, i had no clue |
| 01:49.02 | p3nguin | If you configure your phone in sip.conf (or if you can figure out what file it does get configured in), I can help you with the parameters. |
| 01:49.15 | themalik | ok |
| 01:49.17 | themalik | thanks |
| 01:49.18 | SpaceJockey | Wait(1) did it thanks ;) |
| 01:49.29 | themalik | ill try to find out where it stores |
| 01:49.32 | themalik | ill ask in freepbx |
| 01:49.58 | p3nguin | If I were in your position, I would look in each of the included files as they are listed in sip.conf. |
| 01:50.16 | themalik | ok, that may help |
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| 01:57.34 | themalik | p3nguin: i found the file with my settings for asterisk |
| 01:57.58 | p3nguin | Wait a minute... how did you know host was set to dynamic and deny/permit were configured to 0.0.0.0 if you didn't see the configuration earlier? |
| 01:59.06 | themalik | no i saw it on the freepbx |
| 01:59.17 | themalik | i just found the file thought with all of the settings stored |
| 01:59.30 | p3nguin | alright |
| 01:59.48 | p3nguin | sip_<something>.conf, I guess |
| 01:59.51 | themalik | http://pastebin.com/gxSna4F0 |
| 01:59.56 | themalik | sip_additional.conf |
| 02:01.16 | p3nguin | Those two definitions look okay to me. |
| 02:01.31 | themalik | so its something else... |
| 02:01.34 | themalik | darn |
| 02:01.35 | themalik | lol |
| 02:01.51 | drmessano | Yes, there's at least one other place to look too |
| 02:01.57 | drmessano | But you keep hopping from channel to channel |
| 02:03.40 | themalik | which channel should i use...ill use freepbx..seems more reasonable |
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| 02:35.17 | SpaceJockey | chopp was that you? |
| 02:35.19 | SpaceJockey | hahahah |
| 02:35.32 | SpaceJockey | i got bombarded |
| 02:35.44 | SpaceJockey | I think dive called me too |
| 02:35.48 | SpaceJockey | at the same time |
| 02:36.17 | SpaceJockey | oh man. sorry. wrong window |
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| 03:32.53 | jimi_ | cesar_CR, estas aqui? |
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| 07:49.36 | Sheeplet | lo all |
| 07:52.31 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 07:56.10 | shapr | g'day Sheeplet |
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| 08:25.32 | jayvee | I'm between a rock and a hard place. I previously had allowguest=yes to allow anonymous SIP guests to access my conference number and dial my test tones, and a personal extension. However, I have a PSTN phone hooked up as a SIP user, which means any anonymous user can currently access it until I set allowguest=no, which fixes that. But that now means that the anonymous test tones, conference, etc. is now forbidden. |
| 08:25.56 | jayvee | Is it possible for me to have the best of both worlds? For the record, setting allowguest doesn't make any difference to the users in users.conf. |
| 08:26.38 | jayvee | I'm happy with having allowguest=yes globally and disallowing on the PSTN, or allowguest=no globally and enabling it on one or two extensions. Either way is fine, but neither way works for me. |
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| 08:32.52 | hetii | Hello |
| 08:32.53 | hetii | :) |
| 08:34.33 | WIMPy | Moin |
| 08:34.45 | hetii | i had this issue : http://www.mail-archive.com/asterisk@uc.org/msg09761.html (clear ubuntu 10.10 + Asterisk 1.6.2.7-1ubuntu1 built by buildd @ vernadsky on a i686 running Linux on 2010-06-23 21:08:57 UTC) |
| 08:34.54 | hetii | + latest freepbx |
| 08:35.27 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:35.49 | hetii | there is some bug and to fix them i need to noload this modules: res_odbc.so and app_voicemail_imap.so |
| 08:35.53 | drmessano | jayvee, you need to put the PSTN in a different context and nest the contexts properly |
| 08:36.14 | WIMPy | Moin |
| 08:37.14 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:39.12 | jayvee | drmessano: That's the funny thing. My sip.conf specifies the default context as 'default', which is obviously what context guests go into. Now, the outgoing VoIP and PSTN dial plan is in a context called 'internal', which the users in users.conf are using, which allows them access to external extensions, such as my outbound VoIP dial plan. However, putting my users in a non-default extension does not prevent an anonymous caller from calling that e |
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| 08:39.41 | jayvee | btw, are my messages getting truncated? maybe I should be less sycophantic. ;-) |
| 08:39.49 | shapr | yes, truncated |
| 08:39.58 | jayvee | what were the last few words you got? |
| 08:41.51 | shapr | does not prevent an anonymous caller from calling that e |
| 08:42.18 | jayvee | xtension directly. It does stop them calling my defined dial plans and extensions in extensions.conf, but does not prevent them from calling any user defined in users.conf. |
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| 08:49.23 | jayvee | Ideally I'd like to either have to set something like "allowguest=yes" or create an extension for the user in the default context for them to be able to be contacted anonymously. |
| 08:49.50 | jayvee | So I in effect want allowguest=no, but still have anonymous calls be routed to the default context. |
| 08:53.14 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:54.20 | shapr | How do you define a call that's not anonymous? |
| 08:56.47 | jayvee | A call that's not anonymous is made by somebody from users.conf that has registered. So somebody from the internal context is not anonymous. |
| 08:57.12 | jayvee | Everybody in users.conf is in the internal context. |
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| 09:03.39 | WIMPy | Registering has nothing to do with placing calls. |
| 09:06.36 | jayvee | Registering changes the context that the user is in, and thus can change whether it can place calls depending on what context I put my outbound dial plans in. |
| 09:07.04 | WIMPy | no |
| 09:08.35 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:21b:63ff:feb5:e13b) |
| 09:09.26 | jayvee | Given that what I just said works in practice for me now, would you care to elaborate? |
| 09:10.45 | jayvee | Are you disputing the existence of contexts? |
| 09:11.24 | tuxx- | happy GNU year everyone |
| 09:22.50 | jayvee | shapr: So do you sort of follow what I'm getting at? |
| 09:25.44 | shapr | Yes, I understand. |
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| 09:27.13 | shapr | I don't know how to put authentication between incoming calls and users.conf definitions. |
| 09:29.39 | jayvee | Yeah, it's a funny one. |
| 09:30.10 | asterisk-learner | hi |
| 09:30.18 | asterisk-learner | this is not directly related to asterisk but |
| 09:30.38 | jayvee | I mean, I could sort of work around it by having really obscure usernames. Something like myuser-`pwgen`. |
| 09:30.45 | asterisk-learner | in the VOIP world, is there a standard meaning for a "consult call" or " call consult" |
| 09:30.58 | asterisk-learner | ? |
| 09:31.22 | jayvee | asterisk-learner: Call transfer, perhaps? |
| 09:31.59 | asterisk-learner | ah ok i thought it would mean more inbound call..... |
| 09:32.20 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 09:33.01 | asterisk-learner | so u guys think it might refer to a transfered call ? |
| 09:35.12 | jayvee | I think so. It probably refers more to the call transfer where you dial the destination extension, but it allows you to talk to the person before completing the transfer. |
| 09:36.09 | jayvee | So a consulted transfer would mean you transfer to extension 104 and say "Hi Brian, I have Anna on the line now. Do you have a moment to talk to her?". And then if he accepts, you can complete the transfer and the person will be forwarded through in a single step. |
| 09:36.44 | asterisk-learner | ah ok so it does not refer to blind transfer u mean |
| 09:37.23 | WIMPy | jayvee: No, I'm just saying that registration have nothing to do with it. |
| 09:38.40 | hetii | Q: Hmm i had some strange issue: i set on general settings for sip the bindport to custom one then i set also for exxtenstion this same port. After restarting * i set on my phone also new port. but the phone cannot register itself and the command |
| 09:38.50 | jayvee | WIMPy: Actually yeah, you're right. I have a Sipura phone that I point to in users.conf that doesn't register to Asterisk, but is still defined with a host and port. And it gets assigned to the internal context. |
| 09:39.04 | jayvee | WIMPy: So yes, you're right. Replace "registration" with "being a defined user". |
| 09:39.20 | hetii | sip show peers show me that user is is still on port 5060 but status is unknow |
| 09:40.06 | hetii | then i change on phone again port to 5060 and phone register itself (how its possible when * bind on other port ????) |
| 09:40.13 | hetii | but no call is possible |
| 09:42.12 | hetii | on sip debug i see <--- SIP read from UDP:xxx.xxx.xx.xxx:5060 ---> and then Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:myCustomPort |
| 09:43.26 | hetii | netstat -pantu |grep 5060 show nothing so how its possibe that i see some sip message in asterisk when he don`t bind to 5060 at all |
| 09:43.33 | asterisk-learner | jayvee: ur right ! http://en.wikipedia.org/wiki/Call_transfer |
| 09:43.34 | asterisk-learner | :-) |
| 09:44.49 | jayvee | hetii: The port number of Asterisk doesn't need to match the port number of the SIP UA. Just as long as they have each other's correct port numbers specified in the respective configs. |
| 09:45.13 | jayvee | hetii: From what you've written, it looks like you're trying to change Asterisk's bind port? |
| 09:46.21 | hetii | hmm please explain me that how it can works on different ports in normal way the some network client need to connect to some port on server side. |
| 09:46.52 | hetii | so if i set on server please bind to my custom port then the client need to use this port to connect into server |
| 09:47.18 | jayvee | hetii: That's right. If you tell Asterisk to bind to a custom port, you need to tell the client to use that port. |
| 09:47.39 | jayvee | Or you could make it easier for yourself and just use DNS SRV records to do the work for you. ;) |
| 09:48.20 | hetii | yep so now imagine that * don`t bind to 5060 and my client is able to register itself on it but only on port 5060 not my custom one. |
| 09:48.43 | jayvee | So you're saying your phone doesn't have an option to change the port? |
| 09:49.13 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 09:49.15 | hetii | he had and i set them to my custom one. But then he is not able to register itself. |
| 09:49.47 | hetii | i try also with softphone and had this same issue |
| 09:50.24 | jayvee | Odd. I have used a custom port from the beginning and not had an issue. |
| 09:50.47 | hetii | other interesting things is that i had also my custom port set on my extension context but on sip show peers i still see the 5060 |
| 09:51.04 | kaldemar | hetii: sounds like your client is misconfigured or buggy. |
| 09:51.09 | hetii | i ofc. restart whole * |
| 09:51.37 | kaldemar | sip show peers shows the port that the client is listening on, not asterisk. |
| 09:51.41 | hetii | i check it with grandstrem phone and some softphone so its not possible imhop that they don`t use my custom ports |
| 09:53.30 | hetii | ok so let said that my phone still use 5060 for everything and * bind only on my custom port, how its possible that my phone register itself |
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| 09:54.26 | hetii | ofc. i cannot made call but the registration process work 0_o |
| 09:54.52 | kaldemar | hetii: they don't need to use the same port to LISTEN on packets. |
| 09:56.22 | hetii | ok but if * had my custom port set for LISTENING (so not 5060) how its possible that my phone (whatever port he use) is able to register itself |
| 09:59.45 | hetii | IMHO bind port for * means the LISTENING port where client connect to for registering itself and do the rest job(like tell * on with * he listen or send all sip stuff). |
| 10:00.17 | shapr | that would make sense |
| 10:00.49 | shapr | hetii: Where did you configure the sip bind port? |
| 10:00.51 | shapr | which file? |
| 10:02.21 | hetii | in sip_general_additional.conf (i use freepbx) and netstat -pantu confirm that my * listen on my custom port instead 5060 |
| 10:03.57 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 10:04.24 | shapr | g'day ChrisInSydney |
| 10:04.32 | ChrisInSydney | howdy |
| 10:04.57 | ChrisInSydney | shapr 9pm here |
| 10:05.05 | shapr | It's 4am here. |
| 10:05.15 | ChrisInSydney | thats hard core |
| 10:05.45 | ChrisInSydney | done a few of those. Thats when the #vuc conference call stacrts my time |
| 10:06.19 | ChrisInSydney | (how did that c get in there ?) |
| 10:06.44 | shapr | hetii: Which variable did you set? udpbindaddr? |
| 10:11.49 | hetii | bindport=3060 |
| 10:12.06 | *** join/#asterisk dr_ (~duckz@78.96.111.117) |
| 10:12.32 | hetii | netstat show that * bind ofc. on my custom udp port |
| 10:12.33 | *** join/#asterisk af_ (~getsmart@78.134.21.0) |
| 10:14.40 | *** join/#asterisk _zoom_ (~zero@196.1.219.211) |
| 10:15.16 | _zoom_ | hello, how does asterisk keeps registration and calls sessions? |
| 10:16.04 | kaldemar | _zoom_: yes |
| 10:16.36 | kaldemar | _zoom_: be a little more elaborate. :) |
| 10:17.42 | _zoom_ | kaldemar: am trying to print the current status of asterisk, online users, current calls, ... using php, where to find such info? |
| 10:18.01 | kaldemar | _zoom_: use the manager API. |
| 10:19.27 | shapr | hetii: Have you tried udpbindaddr=0.0.0.0:3060 and tcpbindaddr=0.0.0.0:3060 instead? bindport is under the "NAT support" section, and appears to do something different. |
| 10:19.50 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:19.57 | _zoom_ | kaldemar: can i call it directly from php |
| 10:20.01 | _zoom_ | ? |
| 10:20.54 | _zoom_ | kaldemar: wiki? site? anything plz? |
| 10:22.00 | kaldemar | _zoom_: http://www.voip-info.org/wiki/view/Asterisk+manager+API https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+(AMI) |
| 10:22.17 | kaldemar | there's php frameworks to use it if you don't want to do it yourself. |
| 10:24.36 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 10:25.19 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 10:26.21 | sawgood | ~ask |
| 10:26.21 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:30.01 | ChrisInSydney | _zoom_: Hi, have a look at the voip-info site that kaldemar sent you. Plenty of good stuff there. The other thing you can do is turn on verbose logging and tail the log to a socket, this can give your script some real time processing of what is going on as not all of that info is available via the AMI |
| 10:31.59 | ChrisInSydney | _zoom_: There is a good AMI API which shouldn't take too long to get working. |
| 10:32.28 | WIMPy | What events are you missingn on AMI? |
| 10:33.54 | ChrisInSydney | WIMPy: I was using that method to get qualify time outs for remote handsets |
| 10:34.26 | ChrisInSydney | found it was easier to listen to a stream then query ami |
| 10:35.52 | WIMPy | AMI sends that as events as well. No need to query anything. |
| 10:36.50 | *** join/#asterisk salimb (~chatzilla@81.5.139.17) |
| 10:39.23 | ChrisInSydney | WIMPy Fount it :-/ |
| 10:39.27 | ChrisInSydney | found |
| 10:42.11 | ChrisInSydney | A question re 1.8.1.x. Anybody. Does this version of asterisk use codec negotiation, or does it still transcode ?? |
| 10:43.54 | ChrisInSydney | for example if you allow=g722, allow=alaw on your extensions and your trunk is only allow=alaw, does Asterisk transcode 722 top 711 ?? |
| 10:54.28 | drmessano | ChrisInSydney, of course it's gouing to transcode.. Asterisk is a B2BUA and you're telling one of it's peers to ONLY use a specific codec |
| 10:54.38 | drmessano | app_mindreader hasn't been added yet.. maybe in 2.0 |
| 10:55.37 | hetii | shapr, yes i try and its still the same issue :) |
| 10:55.57 | shapr | hetii: That's weird, what version of asterisk? Do you see the same problem with 1.8? |
| 10:56.08 | ChrisInSydney | drmessano: Could do with app_getmeanotherbeer(now) :-) |
| 10:56.38 | hetii | i don`t check it :> so maybe its fixed on 1.8 :> i use Asterisk 1.6.2.7-1ubuntu1 |
| 10:57.51 | shapr | hetii: It's possible, there's 1.6.2.15 and 1.6.2.16-rc1 |
| 10:58.00 | ChrisInSydney | drmessano: There was this I came across for 1.4.xx http://www.rtpproxy.org/wiki/AsteriskCodecNegotiationPatch |
| 10:59.04 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 10:59.33 | drmessano | ChrisInSydney, this has to do with codec negotiation behavior.. you are forcing ONE codec on a peer. You are creating an entirely different issue. |
| 10:59.37 | ChrisInSydney | drmessano: Not that its a major CPU overhead to translate between 722 & 711, but in some ways it would be "tidier" if is negotiated the codecs rather than translating |
| 10:59.51 | drmessano | ChrisInSydney, so allow the other codec.. |
| 11:00.26 | ChrisInSydney | drmessano: The ITSP doesn't allow 722 only 711a/u |
| 11:00.28 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:00.43 | ChrisInSydney | ..and 729... which I am not interested in |
| 11:00.47 | ChrisInSydney | at this point |
| 11:00.52 | drmessano | ChrisInSydney, ok, so there's nothing to allow then.. If they don't support it, you HAVE TO transcode |
| 11:01.39 | drmessano | ChrisInSydney, you can't make them negotiate a codec they don't even support.. |
| 11:02.52 | ChrisInSydney | drmessano: Scenerio is. Handset supports 722 & 711. ITSP only supports 711. Handset to handset is at 722. Handset to ITSP is 711. |
| 11:04.36 | ChrisInSydney | drmessano: So, what you are saying is that even though the handset has allow=g722 allow=alaw it will always use g722 and translate :-/ |
| 11:05.01 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:07.50 | *** join/#asterisk johani (~johani@h-201-129.A176.priv.bahnhof.se) |
| 11:13.12 | _zoom_ | ChrisInSydney: thnx |
| 11:24.44 | *** join/#asterisk joobie (~joobie@CPE-121-214-124-46.lnse4.lon.bigpond.net.au) |
| 11:32.48 | ChrisInSydney | joobie: hey |
| 11:32.53 | *** part/#asterisk johani (~johani@h-201-129.A176.priv.bahnhof.se) |
| 11:34.08 | _zoom_ | kaldemar: i found a good link http://code.google.com/p/asterisk-php-api/ |
| 11:35.07 | *** join/#asterisk sivang (~sivang@unaffiliated/sivang) |
| 11:35.09 | sivang | hi all |
| 11:35.09 | DND | guys can you help me? i wanted to restrict international dialing to only numbers that start with 00965, 009655, 009656, 009657 00965 |
| 11:35.22 | sivang | I am getting an error when trying to call my snorm from twinkle |
| 11:35.35 | DND | can i just put it 00965[5-7] and 00965 |
| 11:35.36 | DND | ? |
| 11:35.46 | sivang | http://paste.pocoo.org/show/313637/ |
| 11:35.59 | sivang | these are my conffiles: users.conf: |
| 11:36.16 | sivang | oops |
| 11:36.27 | sivang | I meant, sip.conf: |
| 11:37.19 | sivang | http://paste.pocoo.org/show/313639/ |
| 11:38.08 | sivang | extensions.conf: |
| 11:38.09 | sivang | http://paste.pocoo.org/show/313640/ |
| 11:38.24 | sivang | what am I doing wrong? |
| 11:38.51 | kaldemar | DND: yes |
| 11:39.47 | kaldemar | sivang: did you reload your dialplan? |
| 11:42.39 | sivang | kaldemar: yes, I did sip reload configuration |
| 11:44.01 | kaldemar | the actual command for realoading sip configuration is just "sip reload". dialplan is reloaded with "dialplan reload". |
| 11:44.08 | sivang | kaldemar: oh |
| 11:44.11 | sivang | tries |
| 11:44.13 | *** part/#asterisk _zoom_ (~zero@196.1.219.211) |
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| 11:51.56 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 11:59.17 | sivang | kaldemar: ah thanks, that was it. Now how do I enable the G729 codec ? |
| 11:59.29 | sivang | kaldemar: I am getting "no compatible codecs, not accepting this offer!" |
| 11:59.47 | sivang | kaldemar: I mean, I followed the instructions to do build it into asterisk and enable it in the conf file, AFAIR |
| 11:59.51 | kaldemar | sivang: by buying licenses from digium. |
| 12:00.04 | sivang | kaldemar: hmm, there was some free codec on the web? :) |
| 12:00.40 | sivang | kaldemar: that even wanted me to use IPP |
| 12:00.43 | sivang | which I did |
| 12:00.58 | kaldemar | gray stuff, not supported here. |
| 12:06.48 | *** join/#asterisk m_tadeu (~quassel@89.180.202.170) |
| 12:07.39 | ChrisInSydney | eject_ck: how did you go with the SPA400 and the DTMF stuff ?? |
| 12:08.43 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
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| 12:15.16 | sivang | kaldemar: k, thanks |
| 12:15.18 | tzafrir | sivang, no such extension. But on which context? |
| 12:15.39 | sivang | tzafrir: hmm, nice to see you here :) |
| 12:15.59 | sivang | tzafrir: that's what I am trying to find out, I think I fixed it. Let me change my snorm back to codec: whatever and retry |
| 12:16.20 | tzafrir | context=phones |
| 12:17.16 | tzafrir | sivang, dialplan show 1001@phones |
| 12:17.59 | tzafrir | (And why would you want to use g729?) |
| 12:29.45 | sivang | tzafrir: what does it mean? |
| 12:29.58 | sivang | tzafrir: when it shows that? twinke is configued on 1000, and seems to "work" |
| 12:30.14 | tzafrir | what is the output of that command in the asterisk CLI (rasterisk) |
| 12:30.20 | sivang | so I am able to call 1000 from the snorm, but not the snorm from twinke; |
| 12:30.26 | sivang | tzafrir: sec |
| 12:31.15 | tzafrir | I suspect it's because the Snom phone get insulted by you calling it snorm |
| 12:32.02 | tzafrir | If you called it 'snort' it wouldn't have even allowed you outgoing calls |
| 12:33.10 | sivang | haha |
| 12:33.13 | sivang | RTOFls |
| 12:33.31 | sivang | 488 Not Acceptable Here |
| 12:33.37 | sivang | that's what I'm getting from twinkle |
| 12:34.32 | sivang | tzafrir: http://paste.pocoo.org/show/313657/ |
| 12:37.55 | sivang | tzafrir: http://paste.pocoo.org/show/313658/ |
| 12:38.02 | sivang | tzafrir: (/messages) |
| 12:45.45 | tzafrir | sivang, try increasing verbosity level |
| 12:46.06 | tzafrir | with a simple dialplan such as yours, you can use 3 or 4 |
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| 12:59.25 | sivang | tzafrir: so debug level 3? |
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| 13:00.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:03.33 | tzafrir | sivang, yes |
| 13:07.35 | *** join/#asterisk kotique (~kotique@crius.pantheon.fused.net) |
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| 13:08.10 | kotique | Hi guys. What's the state of music on hold in 1.8? Is it possible to stream it so new call doesn't get the first song again? |
| 13:16.30 | leifmadsen | kotique: there is an option for that I believe |
| 13:16.33 | leifmadsen | sort= |
| 13:16.40 | leifmadsen | look at musiconhold.conf.sample |
| 13:18.06 | kotique | but asterisk doesn't spawn internal music streamer yet? |
| 13:18.57 | leifmadsen | I don't understand your question |
| 13:20.02 | leifmadsen | sounds like you want something like streamplayer or ICE |
| 13:31.33 | salimb | I'm getting "Exceptionally long queue length queuing to Local" when attempting to place calls. My SIP clients register without issues but when I place a call I can't hear anything. Any ideas? |
| 13:32.44 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 13:33.26 | WindBack | Is there any way to retrieve Channel Event Logging values from the dialplan like CDR values??? |
| 13:37.36 | sivang | tzafrir: enabled debug 3 |
| 13:37.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:37.38 | sivang | tzafrir: still no go |
| 13:39.15 | sivang | tzafrir: am able to call from sno(r)m (:-)) to twinkle but not the other way around |
| 13:40.10 | sivang | tzafrir: |
| 13:40.14 | sivang | http://paste.pocoo.org/show/313658/ |
| 13:40.15 | sivang | err |
| 13:40.24 | sivang | tzafrir: this is what I am getting: Really destroying SIP dialog '6f2dc00921be2de866391dbc76bb189c@10.200.10.194' Method: INVITE |
| 13:40.32 | sivang | tzafrir: together with the 488 "not allowed" in twinkle |
| 13:51.41 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
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| 14:00.40 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 14:01.16 | Katty | morning |
| 14:05.16 | n3hxs | Morning Katty |
| 14:10.09 | tzanger | morning |
| 14:12.07 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 14:12.58 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 14:13.53 | Katty | hugs n3hxs |
| 14:13.55 | Katty | hugs tzanger |
| 14:17.20 | sivang | morning Katty |
| 14:18.37 | Katty | sivang: allo (= |
| 14:28.05 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
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| 14:37.03 | sivang | so people, how do I enable proper logging to see why twinkle is gettgin 488? |
| 14:37.16 | sivang | when calling my snom which is able to call the twinkle on the other hand |
| 14:37.33 | sivang | debug level 3 does not work |
| 14:37.37 | sivang | neither do higher levels |
| 14:37.53 | sivang | it is like asterisk got its tongue eaten by the cat :) |
| 14:44.07 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:44.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:44.59 | sivang | okay |
| 14:45.04 | sivang | I turned on sip debugging |
| 14:45.19 | sivang | that is what I got, "no supported media type" on the snom side, odd |
| 14:45.37 | *** part/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 14:45.45 | sivang | http://paste.pocoo.org/show/313716/ |
| 14:45.48 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 14:45.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:45.49 | sivang | anybody has an idea? |
| 14:46.03 | sivang | this is on the twinkle -> asterisk -> snom |
| 14:46.06 | sivang | path |
| 14:47.33 | *** join/#asterisk Tim_Toady (~moi@77.49.0.182) |
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| 15:02.35 | pabelanger | sivang: pb your sip.conf |
| 15:02.49 | pabelanger | sivang: looks like a codec issue |
| 15:03.20 | sivang | pabelanger: http://paste.pocoo.org/show/313728/ |
| 15:05.21 | pabelanger | sivang: Add a setting for the codec you want to use, see if that helps. |
| 15:05.28 | pabelanger | disallow=all |
| 15:05.31 | pabelanger | allow=gsm |
| 15:05.37 | pabelanger | for example |
| 15:13.36 | tzanger | see that's teh thing that's awesome about katty |
| 15:13.40 | tzanger | she's very huggable |
| 15:20.59 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
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| 15:25.12 | *** part/#asterisk salimb (~chatzilla@81.5.139.17) |
| 15:25.51 | sivang | pabelanger: but I want to allow whatever's is supported |
| 15:25.56 | sivang | pabelanger: can't I do that? |
| 15:26.15 | sivang | funny how snom has no problem to start a call though |
| 15:26.18 | pabelanger | allow=all |
| 15:26.20 | sivang | and it is only twinkle |
| 15:27.04 | sivang | pabelanger: same |
| 15:27.08 | sivang | pabelanger: same error |
| 15:27.31 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
| 15:27.53 | sivang | Warning: 304 x-snom-adr "No supported media type found" |
| 15:28.11 | pabelanger | PB a complete SIP trace |
| 15:28.38 | sivang | pabelanger: out of snom or a*'s ? |
| 15:28.51 | pabelanger | Asterisk |
| 15:30.27 | sivang | pabelanger: http://paste.pocoo.org/show/313750/ |
| 15:31.31 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 15:33.53 | pabelanger | Looks to be a codec issue on your snom. Which codecs are enabled? |
| 15:34.26 | *** join/#asterisk Tim_Toady (~moi@77.49.0.182.dsl.dyn.forthnet.gr) |
| 15:36.32 | sivang | pabelanger: 723.1 |
| 15:37.11 | sivang | pabelanger: but I thought this should be a no issue, since if the snom is able to invite twinkle, through asterisk, then the reverse should work the same? |
| 15:37.35 | *** join/#asterisk DoDaT69 (~DoDaT69@173.160.86.155) |
| 15:38.22 | sivang | pabelanger: thanks that solved it |
| 15:38.25 | sivang | pabelanger: how *odd* |
| 15:38.44 | sivang | pabelanger: I didn't know that the reciever needs to support the INVITors codec |
| 15:38.53 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:38.58 | sivang | pabelanger: as, I was sure audio is going through asterisk or the PBX |
| 15:39.18 | sivang | pabelanger: and this is suboptimal, since I want to enable two clients with different codecs to communicate |
| 15:39.25 | sivang | pabelanger: what's happenign here? :-) |
| 15:39.27 | sivang | is confused |
| 15:46.46 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.85.77) |
| 15:46.53 | sivang | pabelanger: how do I set it to transcode then? (I figure it just pass-thrus the audio) |
| 15:50.23 | kotique | asterisk sending options packets to dead peer like crazy |
| 15:50.25 | kotique | what's going on? |
| 15:50.46 | sivang | how can I find out the codec being used for my chanels? |
| 15:50.50 | kotique | it supposed to send max 7 packets with 60 sec interval |
| 15:51.19 | anonymouz666 | maybe asterisk is trying to bring back the dead peer again |
| 15:52.13 | anonymouz666 | maybe the peer could be too young to die |
| 15:53.05 | anonymouz666 | sivang: sip show channels (if using SIP) |
| 15:53.54 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 15:54.42 | *** join/#asterisk MindTheGap (~MindTheGa@201.48.62.132) |
| 15:59.10 | kotique | it's dead long ago. why is it insisting on that? |
| 15:59.17 | kotique | 15:58:54.987229 IP 174.37.247.106.5060 > 109.185.140.24.5078: SIP, length: 555 |
| 15:59.42 | kotique | 15:58:55.986228, 15:58:56.986263, 15:59:06.986004, 15:59:10.985079 |
| 15:59.45 | kotique | that is CRAZY |
| 16:00.22 | kotique | here's what i see in peer config: Qualify Freq : 120000 ms |
| 16:00.40 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:03.06 | sivang | anonymouz666: it does not show the codec being used |
| 16:03.12 | sivang | anonymouz666: already tried it |
| 16:04.10 | sivang | anonymouz666: oh it does! sorry I confused ti with show peers |
| 16:04.28 | sivang | anonymouz666: thank you. |
| 16:05.15 | *** join/#asterisk sourcode (~code@ppp-61-90-16-123.revip.asianet.co.th) |
| 16:05.50 | *** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net) |
| 16:07.24 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 16:07.47 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 16:09.13 | *** join/#asterisk mrbnet (~mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
| 16:11.59 | *** join/#asterisk Lord_Rahl (~Lord_Rahl@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 16:12.26 | Lord_Rahl | ? if you were installing a new server would go with 1.4 or 1.6 |
| 16:13.00 | Freeaqingme | how about 1.8? |
| 16:13.09 | Freeaqingme | 1.8 is LTS as well |
| 16:13.39 | Lord_Rahl | scared of new things lol |
| 16:14.03 | TimeRider | 1.8 easier, no need to compile addons :) |
| 16:14.18 | Freeaqingme | Lord_Rahl, it would be really scary if you used trunk |
| 16:14.19 | Lord_Rahl | does it still support mysql cdr |
| 16:14.28 | Freeaqingme | yes (afaik) |
| 16:20.01 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 16:20.57 | Lord_Rahl | one more ? I need a gui for management. just for the simple function like time rules and simple stuff. I am not a big lover of freepbx & I know asterisk GUI is a dead project or there any you know that is light weight and ez for management level user |
| 16:23.00 | Naikrovek | Lord_Rahl: freepbx can do what you want. you can set up restrictions based on user so mgmt can't go all dingleberry through yoru system |
| 16:23.16 | Naikrovek | i know you said you're not a huge fan but it'll do that for you even if you don't like it |
| 16:23.39 | p3nguin | If you insist on having a GUI, why not install AsteriskNOW? |
| 16:23.41 | *** join/#asterisk sourcode__ (~code@ppp-61-90-16-123.revip.asianet.co.th) |
| 16:23.45 | Naikrovek | ^^^ |
| 16:23.47 | Naikrovek | what he said |
| 16:24.08 | *** join/#asterisk neurosys (~neurosys@69.199.226.33) |
| 16:24.09 | Freeaqingme | or just use the config files? it's not that hard |
| 16:24.17 | Naikrovek | well he wants a gui for a manager |
| 16:24.46 | p3nguin | If you need/want a GUI, AsteriskNOW is perfect. From CD boot-up to operational phone system in 15 minutes. |
| 16:24.58 | Naikrovek | ^^^ |
| 16:25.00 | Naikrovek | what he said |
| 16:25.32 | Qwell | someone should buy the author of that a beer or something |
| 16:25.44 | Naikrovek | Qwell: good lord are you out of beer again |
| 16:25.58 | Qwell | I'm always out of beer. |
| 16:26.11 | tuxx- | what, free beer? |
| 16:26.21 | tuxx- | raises his hand |
| 16:26.22 | tuxx- | beer plx. |
| 16:26.39 | p3nguin | And with AsteriskNOW, you get the choice of FreePBX, Asterisk GUI, or no GUI. |
| 16:26.39 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:26.44 | Naikrovek | ^^^ |
| 16:26.48 | Naikrovek | what he ... aah you get it |
| 16:32.19 | Lord_Rahl | oooh, are they still updating asterisk gui ? I use and like that light wieght and when you want to touch the config it get out of your unlike freepbx (will the late time I try it) |
| 16:32.47 | p3nguin | If it works, why does it need updated? |
| 16:33.03 | Lord_Rahl | p3nguin, true that :) |
| 16:33.35 | *** join/#asterisk tasca (~tasca@189.4.108.113) |
| 16:33.59 | p3nguin | Unless, of course, it has bugs... |
| 16:34.05 | Naikrovek | i don't think it's maintained anymore, but I'm not sure. It'll likely break with asterisk 1.8 |
| 16:34.06 | p3nguin | Then updates could be useful. |
| 16:34.35 | Lord_Rahl | ? does asterisk now still use rpath or did it move to cent |
| 16:34.46 | Naikrovek | asteriskNOW uses centos |
| 16:35.14 | Lord_Rahl | sweet, love me some cli and vim |
| 16:35.21 | Naikrovek | yeah |
| 16:35.26 | Naikrovek | AsteriskNOW is very nicely done IMHO |
| 16:35.45 | Naikrovek | you can update it all and when it reboot it works. |
| 16:36.37 | drudge` | i do like asterisk now |
| 16:36.59 | p3nguin | I have no use for a GUI, and I still like AsteriskNOW. It makes deployment of CentOS and vanilla Asterisk a breeze. |
| 16:37.07 | drudge` | tho, i never had any issues with trixbox 2.2 or something |
| 16:37.21 | Naikrovek | yeah you can install asterisknow without a gui at all and still take advantage of all the packaging |
| 16:37.30 | Naikrovek | it's slick, yo |
| 16:37.43 | p3nguin | Which is what I would do if I didn't use AsterisNOW, anyway. |
| 16:38.03 | Naikrovek | how DARE you |
| 16:38.04 | Lord_Rahl | Thanks, I will give it a go.. |
| 16:38.09 | Naikrovek | oh wait i misread |
| 16:38.15 | Naikrovek | :P |
| 16:38.19 | Naikrovek | finds lunch... |
| 16:38.58 | Lord_Rahl | use to compiling my own but hey it will save me time and possible compile errors |
| 16:39.08 | p3nguin | If you've already got CentOS, enable the appropriates repos and away you go. |
| 16:39.43 | p3nguin | It doesn't make it any better to compile it on your machine as opposed to having qwell compile it for you. |
| 16:40.14 | *** join/#asterisk dajhorn (~dajhorn@transmisor.vanadac.com) |
| 16:40.45 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:55.33 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 17:03.44 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 17:11.30 | devmod | Is there a way to add an extra sip header to an outgoing call? |
| 17:11.40 | thehar | SIPAddHeader() |
| 17:11.53 | devmod | Add a header to the invite tho |
| 17:11.53 | p3nguin | GENIUS! |
| 17:12.04 | Qwell | if I'm not mistaken, there's a SIP_HEADER() function that can be used too |
| 17:12.21 | devmod | I should add I am using Originate through AMI |
| 17:12.33 | devmod | not sure how asterisk would know what channel to add the header to |
| 17:12.44 | thehar | it guesses |
| 17:12.53 | thehar | i joke |
| 17:14.39 | JunK-Y | originate to a local chan, which will add the header, then will make the call out? |
| 17:15.31 | devmod | Yeah I could do that but I am tracking the uniqueid, that could create a new leg right? |
| 17:15.57 | devmod | s/could/would |
| 17:17.22 | p3nguin | You were close! |
| 17:17.58 | p3nguin | s/You were close/Better luck next time/ |
| 17:21.27 | *** part/#asterisk sivang (~sivang@unaffiliated/sivang) |
| 17:23.21 | *** join/#asterisk megalomano (~kvirc@nggw-of.alocomm.com) |
| 17:23.36 | megalomano | hi people |
| 17:35.12 | Qwell | megalomano: Do not message people without permission. |
| 17:35.17 | Qwell | ~help |
| 17:35.25 | Qwell | glares at infobot |
| 17:35.29 | Qwell | ~ask |
| 17:35.29 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:41.25 | WindBack | Is there any way to retrieve Channel Event Logging values from the dialplan like CDR values??? |
| 17:41.32 | russellb | WindBack: CHANNEL() |
| 17:46.18 | WindBack | russellb: I see the function CHANNEL() in asterisk 1.8.x and also in 1.6.x, But I know that CEL was introduced in 1.8.x. Is the same function? |
| 17:46.28 | russellb | yes |
| 17:46.48 | WindBack | russellb: so, what is the difference? |
| 17:46.53 | russellb | There is no CEL record stored on channel like a CDR is stored on a channel |
| 17:47.10 | russellb | CEL is just a series of events, and the events are comprised of channel data. You can read channel data using the CHANNEL() function. |
| 17:47.32 | russellb | that's why there is no special CEL function |
| 17:52.53 | WindBack | russellb: so basically CEL is the mechanism of logging the channel data events? |
| 17:53.43 | Naikrovek | CEL = Channel Event Logging |
| 17:53.45 | Naikrovek | ... i think |
| 17:53.53 | WindBack | russellb: There is no extra info since CEL was added |
| 17:54.07 | russellb | Naikrovek: correct |
| 17:54.13 | russellb | WindBack: ok |
| 17:54.29 | WindBack | russellb: thanks |
| 17:57.13 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 17:57.48 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 18:00.22 | *** join/#asterisk sizzers (~jerrywilt@c-68-41-174-94.hsd1.mi.comcast.net) |
| 18:01.59 | sizzers | Wow, ok guys, having serious issues with voicemail on 1.6. I'm modifying voicemail.conf to change the fromstring and body for VMtoEMail... then i reload asterisk, leave a VM, and the email I get is perfect. Then, at some point later, somebody else leaves a voicemail, and the fromstring and message body I get are completely different. |
| 18:02.52 | sizzers | I found that the body and fromstring are coming from minivm.conf, but I don't understand why some emails take settings from voicemail.conf, and others are taking params from minivm.conf |
| 18:02.58 | sizzers | it doesn't make any sense to me at all |
| 18:03.30 | sizzers | Can anyone explain how the new minivm.conf and voicemail.conf are working together? |
| 18:03.46 | sizzers | i was not able to find much documentation at all |
| 18:05.18 | Qwell | Are you using minivm in some cases, but voicemail in others? |
| 18:06.18 | sizzers | thats the question |
| 18:06.35 | sizzers | I haven't seen the documentation that talks about invoking one versus the other |
| 18:07.29 | sizzers | my diaplan simply sends people to Voicemail(mailboxnumber,su) |
| 18:07.35 | Qwell | then you aren't using minivm |
| 18:08.15 | sizzers | oh... is using minivm simple by invoking the "minivm()" command instead of "voicemail()" |
| 18:08.23 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 18:08.39 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:09.17 | sizzers | Qwell, This is what doesn't make sense I modified the body and fromstring in voicemail.conf, left a VM, and all was well |
| 18:09.32 | sizzers | Then, some hours later, I got another voicemail with the body and fromstring defined in minivm.conf |
| 18:09.39 | Qwell | Where did you change it? |
| 18:09.50 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 18:09.50 | sizzers | here's my voicemail.conf |
| 18:09.52 | Qwell | NO |
| 18:09.53 | Qwell | ~pastebin |
| 18:09.53 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:09.56 | sizzers | yah pastebin |
| 18:11.14 | *** join/#asterisk engrxyz (~cbvvvvv@host81-150-217-173.in-addr.btopenworld.com) |
| 18:11.51 | sizzers | http://asterisk.pastey.net/144508 |
| 18:12.59 | *** join/#asterisk WonTu (~WonTu@p57B53338.dip.t-dialin.net) |
| 18:13.13 | *** part/#asterisk WonTu (~WonTu@p57B53338.dip.t-dialin.net) |
| 18:14.52 | *** join/#asterisk sivang (~sivang@unaffiliated/sivang) |
| 18:14.55 | sivang | hi all, again |
| 18:15.01 | sivang | I've setup my outgoing conf as in the book |
| 18:15.04 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 18:15.08 | sivang | but I get service unavailable |
| 18:15.34 | sivang | NOTICE[26225]: chan_sip.c:13410 handle_response_invite: Failed to authenticate on INVITE to '"1001" <sip:942@10.200.10.194>;tag=as37568495' |
| 18:15.40 | sivang | This is what I get when trying an outgoing number |
| 18:15.51 | sivang | how does asterisk supposed to know this is an outgoing number btw? |
| 18:16.03 | lirakis | sweet jesus |
| 18:16.07 | sizzers | lol lirakis |
| 18:16.56 | sizzers | sivang, does the nubmer you're trying to call start with area code 942? |
| 18:18.57 | sivang | sizzers: hmm, no :) |
| 18:19.20 | WindBack | russellb: if I do: exten => h,1,Noop(${CHANNEL(linkedid)}) the CLI says that linkedid is not valid. Also I have the same problem using ${CHANNEL(peer)} |
| 18:19.21 | p3nguin | sivang: All calls from your phone are going to be inbound to asterisk. What asterisk does next depends on your dialplan. |
| 18:19.23 | sivang | sizzers: how can I signal asterisk that I want to go outgoing? |
| 18:19.44 | sivang | p3nguin: I did what the book told me to do |
| 18:19.59 | sizzers | sivang, he's trying to answer your quesiton |
| 18:20.02 | Qwell | what did the book tell you to do? |
| 18:20.05 | russellb | WindBack: "core show function CHANNEL". If you don't see it there, then it's not supported. |
| 18:20.14 | p3nguin | sivang: If your dial plan instructs asterisk to Dial() a number on an external peer which is configured correctly, that will make the call an outgoing call. |
| 18:20.22 | WindBack | russellb: thanks, good idea |
| 18:20.32 | lirakis | the book told me to kill everyone .... |
| 18:20.42 | lirakis | i didnt listen this time |
| 18:20.42 | Qwell | lirakis: You weren't supposed to read it backwards.. |
| 18:20.43 | lirakis | ;) |
| 18:21.03 | sizzers | sivang, when you dial a number from your phone, asterisk looks in sip.cfg for your phone's context... then, it goes to that context in the diaplan and tries to find a match for the number you dialed.... then, it follows the steps you've defined for that number |
| 18:21.15 | p3nguin | sip.conf |
| 18:21.16 | sizzers | does that make sense to you? |
| 18:21.55 | sivang | sizzers: it does, but following the book it did not say nothing about how to let asterisk know this should be an outgoing call, in freeswitch that was impossible to miss |
| 18:22.18 | sivang | sizzers: it felt weird, but I went on with the book as I thought maybe it is just different |
| 18:22.21 | p3nguin | sivang: I've already explained that part to you. Please pay attention. |
| 18:22.22 | sizzers | i can't comment on the effectiveness of the book in answering your question |
| 18:23.18 | Katty | hai |
| 18:23.23 | sivang | p3nguin: I did, but you realize my dialplan was instructed by a book for beginners and I followed it, that's all, expecting its promise :) |
| 18:23.30 | Katty | p3nguin: you still up in STL? |
| 18:24.08 | p3nguin | There is no such thing as an outgoing call until the dialplan processes it and sends it out a configured peer. |
| 18:24.13 | sizzers | sivang, from astersik's perspective, the difference between an "outbound" call and a "local" call is simply how you route it in extensions.conf.... "outbound" is really only a relevant term for people |
| 18:24.30 | p3nguin | katty: In STL, no. I'm in the region, though. |
| 18:24.33 | sivang | Qwell: I followed pages 97 to 101 |
| 18:24.49 | Katty | p3nguin: how far from stl you figure |
| 18:25.08 | p3nguin | katty: 70 miles to the river. |
| 18:25.16 | Katty | mmmk |
| 18:25.18 | Katty | IL then? |
| 18:25.23 | sivang | sizzers: I understand that completely. I did not see anytyhing in my lack of understanding the dialplan to let asterisk know it should route it to the provider's account |
| 18:25.30 | p3nguin | katty: correct. |
| 18:25.31 | sivang | sizzers: hence my wondering |
| 18:25.41 | Katty | p3nguin: we may have to meet for lunch or dinner sometime |
| 18:25.51 | Katty | p3nguin: i'm moving to stl probably in 6 months or so |
| 18:26.02 | sivang | also, maybe this is related to the problem? |
| 18:26.07 | sivang | [Jan 3 13:22:03] NOTICE[26225]: chan_sip.c:13719 handle_response_register: Outbound Registration: Expiry for 212.179.142.75 is 120 sec (Scheduling reregistration in 105 s) |
| 18:26.16 | p3nguin | sivang: You have to create the account for any provider in sip.conf. Any provider is configured as yet another peer. |
| 18:26.31 | sivang | p3nguin: correct, which I did, let me PB it |
| 18:26.38 | sivang | oh oops |
| 18:26.43 | sivang | better not, it contains password |
| 18:26.47 | sizzers | sivang, that error is not uncommon and can often be disregarded if "sip show registry" shows a status of registered |
| 18:26.48 | Katty | anyone else in STL? |
| 18:26.59 | carrar | I'm in SeaTtLe |
| 18:27.05 | Katty | :< |
| 18:27.07 | p3nguin | katty: Soup, salad, and bread sticks at Olive Garden? |
| 18:27.13 | Katty | p3nguin: perfect |
| 18:27.23 | sivang | sizzers: it does, it also shows it in sip show peers |
| 18:27.32 | sivang | sizzers: can I dial it from within the CLI? |
| 18:27.45 | p3nguin | sivang: hide the passwords... ONLY the passwords. |
| 18:27.49 | sizzers | you can use the originate command, but that is a bit tricky |
| 18:28.05 | sivang | the timeout expriary warninig is okay other that? |
| 18:28.09 | sizzers | you're better off at continuing to try to fix the diaplan |
| 18:28.12 | sizzers | yes |
| 18:28.21 | sivang | sizzers: may I ask why so? |
| 18:28.34 | sivang | (why is the timeout okay, that is :) |
| 18:29.09 | sivang | I'm running with sip debug and apparently the registeration is re-done due to it every t seconds |
| 18:30.04 | sizzers | sivang, i don't know the answer to that question, i only know that we see it but our registration still works. Instead, tell me the context that your phone goes to |
| 18:31.48 | p3nguin | sivang: If you run "sip show registry", that shows who you have registered TO. Registering TO a peer is what the notice is regarding. If sip show registry says it is registered, turn down the debug level and go on about your business. |
| 18:32.07 | sivang | http://paste.pocoo.org/show/313877/ |
| 18:32.19 | sivang | p3nguin: thanks, will do |
| 18:32.26 | sivang | that's sip.conf btw |
| 18:32.29 | sivang | now the dialplan |
| 18:34.55 | sivang | http://paste.pocoo.org/show/313880/ |
| 18:35.22 | Qwell | sivang: and what are you dialing? |
| 18:35.31 | sivang | 0543077894 |
| 18:35.57 | sizzers | is that a valid phone number? |
| 18:35.58 | p3nguin | exten => _X.,n,Dial(SIP/012Voip/${EXTEN}) should match that. |
| 18:36.08 | sizzers | yep |
| 18:36.16 | sivang | p3nguin: that's what causing it to send it to the 012voip gateway? |
| 18:36.25 | sizzers | yep |
| 18:36.33 | sivang | okay |
| 18:36.43 | sivang | how does it know to differntiate from the internal extension? |
| 18:36.45 | p3nguin | And 0543077894 will be sent to the configured 012Voip peer using the SIP channel tech. |
| 18:37.00 | p3nguin | Your configured dial plan make the determination. |
| 18:37.18 | p3nguin | "internal" and "outgoing" are human concepts. |
| 18:37.24 | sizzers | it deifferentiates based on the number you dial.... becauase every number you dial will match a different rule |
| 18:37.25 | sivang | p3nguin: so if it exactly matches 1001 it will call that extension? |
| 18:37.31 | sivang | sizzers: right, thanks |
| 18:37.36 | sivang | sizzers: that's what I now assumed |
| 18:37.37 | sizzers | i think u got it |
| 18:37.43 | p3nguin | If it matches 1001, it will Dial SIP/1001. |
| 18:38.15 | p3nguin | extension 1001 says to dial peer 1001 as configred, using the SIP channel tech. |
| 18:38.16 | sizzers | and _X. is a very generic match |
| 18:38.42 | sivang | [Jan 3 13:35:22] NOTICE[26225]: chan_sip.c:13410 handle_response_invite: Failed to authenticate on INVITE to '"1001" <sip:942@10.200.10.194>;tag=as10b9bf3b' |
| 18:38.58 | Qwell | So it's failing to authenticate to your provider |
| 18:39.03 | p3nguin | So there's an authentication issue with the ITSP. |
| 18:39.18 | p3nguin | username, secret, IP address... |
| 18:39.36 | sivang | it shows as if I try to auth to 10.200.10.194 which is my local asterik I'm trying to configure |
| 18:39.48 | p3nguin | Are you behind NAT? |
| 18:39.48 | sivang | does that say it fails auth against the 212... soemthing 012voip server? |
| 18:40.01 | sivang | bah, /me checks |
| 18:40.04 | p3nguin | You didn't configure your Asterisk system to operate behind NAT. |
| 18:40.19 | sizzers | there's clearly some NAT here |
| 18:40.34 | sizzers | Qwell... I'm still confused about my issue |
| 18:40.52 | Qwell | sizzers: Did you configure minivm? |
| 18:40.59 | sizzers | never touched it |
| 18:41.03 | sivang | so, there is some yes, but that did not stop other PBX from working without doing anything. What do I need to do to make it work with NAT? |
| 18:41.08 | sizzers | didn't even know it existed until i grep'd for the from string and found it in that file |
| 18:41.18 | sizzers | however I did just figure out one very clear issue. the mailbox that these emails are coming doesn't exist in voicemail.conf |
| 18:41.22 | sizzers | i had a typeo |
| 18:41.27 | Qwell | sizzers: it's possibly just using the default, which would be the same in minivm |
| 18:41.41 | sizzers | but i'm never ever invoking minivm() |
| 18:42.05 | sizzers | and again... some emails use minivm.conf, and others use voicemail.conf |
| 18:42.22 | sizzers | i can't find any distinguishing factors between the voicemail i leave as a test.... and ones people leave later |
| 18:42.24 | Qwell | s/minivm.conf/the default message/ |
| 18:42.30 | sizzers | yah it's there |
| 18:42.33 | sizzers | and it matches |
| 18:42.39 | sizzers | the question is... why is asterisk using that |
| 18:43.47 | p3nguin | sivang: Here is a working example of a sip.conf configured for NAT: http://pastebin.com/m59d17875 |
| 18:43.58 | p3nguin | ~sipnat |
| 18:43.58 | 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 |
| 18:44.09 | p3nguin | sivang: You can also read these guides ^^^ . |
| 18:44.40 | sizzers | wait.. nevermind... the mailboxes do exist in voicemail.conf |
| 18:44.42 | sizzers | scratch that |
| 18:46.25 | Qwell | sizzers: is it forwarded messages? |
| 18:46.57 | sivang | p3nguin: thanks |
| 18:47.07 | sizzers | Qwell, i don't understand what you mean |
| 18:49.10 | sizzers | furthermore, minivm show stats shows 0 voicemail accounts, and <none> received messages since last reset (which was about 48 hours ago) |
| 18:49.38 | sizzers | but i got an email which had the body and fromstring taken from minivm.conf about 6 hours ago |
| 18:51.01 | sizzers | The main voicemail app should only read settings from voicemail.conf (as i currently understand it) . And when I do "voicemail show users" it shows 18 new messages on the mailbox in question |
| 18:51.06 | sizzers | i'm taking this to asterisk-dev |
| 18:51.24 | sizzers | i honestly think this is a bug where voicemail() is using settings from minivm.conf |
| 18:52.05 | sizzers | what the hell happened to asterisk-dev lol |
| 18:53.54 | *** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net) |
| 18:55.33 | sizzers | wait.... when the hell do these pager parameters come into play? |
| 18:56.17 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 18:57.21 | sizzers | OMG |
| 18:57.25 | sizzers | problem solved |
| 18:57.51 | sizzers | you're a genius Qwell |
| 18:57.53 | Qwell | oh, I know |
| 18:57.56 | russellb | ha |
| 18:58.11 | Qwell | what did I do this time though? |
| 18:58.29 | Katty | hugs on Qwell |
| 18:58.30 | sizzers | oh the solution has nothing to do with you actually |
| 18:58.39 | Qwell | Katty: eep! |
| 18:58.41 | sizzers | i just wanted to give u a compliment for all the help you give people around here |
| 18:59.01 | sizzers | Turns out i didn't understand something about this line..... |
| 18:59.46 | sizzers | (voicemail.conf) 10000 => 10000,Main Mailbox,email1@email1.com,email2@email2.com,attach=yes,delete=yes |
| 19:00.32 | sizzers | email1 and email2 are not a list of emails in a comma-separated notation... they're 2 distinct fields... one is email address, and the other is pager address... and they are governed by different rules |
| 19:00.52 | sizzers | it just so happens that the default fromstring and body format for pager address is the same as the default for minivm |
| 19:01.44 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 19:01.44 | *** mode/#asterisk [+o angler] by ChanServ |
| 19:01.53 | sizzers | and since i deelted the pager parameters from vociemail.conf , the grep made it seem like they were coming from minivm.conf, when in fact asterisk was using the built in defaults for pager (without it existing in any text file)... which is strange |
| 19:05.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 19:08.41 | *** join/#asterisk z4nD4R (~zandar@bband-dyn170.178-41-107.t-com.sk) |
| 19:09.34 | z4nD4R | hi, asterisk place record call in /var/spool/asterisk/monitor... can i change this dir? |
| 19:09.55 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 19:10.16 | Qwell | z4nD4R: specify the dir, and it'll use it |
| 19:10.28 | Qwell | Record(/path/to/something/filename.ulaw) |
| 19:10.29 | sizzers | lol Qwell, obviously he would need to know where to change it |
| 19:11.00 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:11.15 | z4nD4R | Qwell: i use Monitor function... is the same? |
| 19:11.26 | Qwell | should be |
| 19:11.50 | z4nD4R | Qwell: ok .. w8 :D |
| 19:16.28 | citywok | argh, my prod 1.8.1.1 has now locked up twice. it stops responding, ican *-r and if i do sip show peers it just shows nothing. looking at verbose logs the last thing it did was queue(queuename,tnr,30) |
| 19:16.43 | Katty | grooves |
| 19:16.47 | Katty | dances with Qwell |
| 19:19.36 | russellb | citywok: build with DEBUG_THREADS and grab "core show locks" output |
| 19:20.39 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 19:20.52 | z4nD4R | Qwell: it works... thx |
| 19:22.27 | citywok | argh, i kinda figured. will it still show locks even when it wont do anything else? |
| 19:22.44 | russellb | citywok: it should, unless there is a deadlock in the lock tracking code itself |
| 19:22.51 | russellb | which would be rather unfortunate. |
| 19:22.58 | citywok | last time when i did core stop now it wouldn't shut dowwn, this time it actually worked when i did core stop now |
| 19:23.21 | citywok | this tiem i tried to kill-9 it right away b/c thats what i had to do last time, and it wouldn't even unload on that. |
| 19:23.34 | citywok | although without any contacts and on a few hours of sleep, maybe i missed the PID |
| 19:24.07 | citywok | i'm tempted to install 1.6.2.x on it, i don't want to piss these guys off with constant crashes. |
| 19:24.17 | p3nguin | pkill or killall helps alleviate such problems. |
| 19:26.00 | citywok | good to know |
| 19:27.24 | citywok | rus should i dont_opt as well? |
| 19:27.54 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 19:28.04 | russellb | that would be best, yes |
| 19:28.45 | citywok | kk. will compile this crap and fire it up when all the calls end. |
| 19:29.03 | citywok | it may not be until later, we're beach bound shortly. stupid calls on vacation |
| 19:31.20 | citywok | ah this one was 1.8.1, was there anything major in 1.1? |
| 19:31.44 | russellb | not for a deadlock ... fixed a crash and fixed google voice calls |
| 19:33.01 | citywok | kk ty |
| 19:35.43 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 19:36.04 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
| 19:36.26 | *** join/#asterisk ketas-l (~ketas@227.82.191.90.dyn.estpak.ee) |
| 19:41.37 | *** join/#asterisk twanny796 (~twanny@a171.201.adsl.nextweb.net.mt) |
| 19:42.24 | twanny796 | trying to compile asterisk-gui, how do I download the code, trying according to the book, but there's nothing in the directory!! |
| 19:42.35 | *** join/#asterisk ketas-l (~ketas@227.82.191.90.dyn.estpak.ee) |
| 19:47.28 | z4nD4R | somebody know wich variable i can use in extensions.conf? .. some list of possible variavbles? |
| 19:48.31 | Kobaz | z4nD4R: huh? |
| 19:48.49 | z4nD4R | http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List :D |
| 19:49.03 | p3nguin | You can use any variable you want to use. |
| 19:49.48 | Kobaz | z4nD4R: those are variables that are for specific purpeses... anything else is up for grabs |
| 19:50.25 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 19:50.34 | p3nguin | You can even create arbitrary variables. |
| 19:50.59 | Kobaz | well that's what i meant |
| 19:51.09 | Kobaz | any other name other than those in the list, you can pretty much use |
| 19:51.38 | z4nD4R | p3nguin: Kobaz i need set to callfilename to -> from-to-datetime.wav |
| 19:51.43 | tzafrir | http://code.google.com/p/lynks-ajax-panel/source/browse/trunk - interesting |
| 19:51.51 | tzafrir | but comments are all in russian |
| 19:52.10 | tzafrir | http://code.google.com/p/lynks-ajax-panel/ |
| 19:52.45 | tzafrir | it's indeed a "fop replacement". currently they include the mysql and asterisk manager passwords in their source code |
| 19:52.57 | Kobaz | z4nD4R: i don't know if there's a setting for that, you may have to modify the c code |
| 19:54.12 | z4nD4R | Kobaz: hmm i see .. EXTEN ( to ) , callerid ( may be from ) and same TIMESTAMP? |
| 19:56.33 | Kobaz | I don't know what you're asking |
| 19:57.29 | Kobaz | Perhaps someone here can help you in your native language? |
| 20:01.34 | *** join/#asterisk tasca (~tasca@201.47.74.147) |
| 20:01.46 | z4nD4R | Kobaz: i fix it i need this exten => 600,n,Set(CALLFILENAME=${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) |
| 20:02.34 | *** join/#asterisk circut (~circut@c-71-57-110-244.hsd1.il.comcast.net) |
| 20:03.47 | Kobaz | call recordings? |
| 20:04.01 | z4nD4R | Kobaz: yes |
| 20:04.14 | Kobaz | what asterisk version? |
| 20:04.21 | z4nD4R | Kobaz: 1.8.0 |
| 20:04.44 | Kobaz | using Monitor ? or MixMonitor? |
| 20:05.10 | circut | Kobaz: make sure to talk with your legal team before implementing this though.. |
| 20:05.13 | z4nD4R | Kobaz: monitor... is MixMonitor better? |
| 20:05.35 | Kobaz | don't use Monitor it has problems if you have high io... you will get audio quality issues |
| 20:05.47 | Kobaz | use MixMonitor... and the proper variable to set is MIXMONITOR_FILENAME |
| 20:06.45 | Kobaz | i'm not sure where you got CALLFILENAME from.. it has nothing to do with recording |
| 20:06.45 | z4nD4R | thx.. i try it.. w8 |
| 20:11.37 | z4nD4R | Kobaz: but MixMonitor dont save my call |
| 20:11.56 | z4nD4R | Kobaz: i have set as |
| 20:11.57 | z4nD4R | exten => 600,n,Set(MIXMONITOR_FILENAME=${CALLERID}:${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) |
| 20:11.57 | z4nD4R | exten => 600,n,MixMonitor(wav49,/var/www/${EXTEN}/${MIXMONITOR_FILENAME},m) ;records the call to /var/www/$USER |
| 20:12.35 | *** join/#asterisk oktay (~oktay@92.45.152.39) |
| 20:12.55 | oktay | anybody have a dlink ata? |
| 20:13.36 | Kobaz | you're using the wrong sytax |
| 20:13.37 | Kobaz | syntax |
| 20:13.50 | Kobaz | z4nD4R: read the documentation for MixMonitor... it works differently than Monitor |
| 20:14.08 | z4nD4R | Kobaz: ok... |
| 20:14.18 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:22.50 | z4nD4R | Kobaz: it work fine, my file named 600-20110103-202708.wav but, i want to from who get to call... variable CALLERID? |
| 20:23.10 | jaytee | there are days when I want to change all my passwords to alzheimers.....but I'd probably forget that too. |
| 20:24.01 | Kobaz | ${CALLERID(name)} ${CALLERID(num)} |
| 20:24.19 | oktay | anybody know why a client (dlink ata) in the DMZ might keep trying to REGISTER ? And why would Asterisk send it a SIP/401 ? |
| 20:24.22 | Kobaz | core show function CALLERID |
| 20:24.50 | Kobaz | oktay: if the client is registering, i would assume it's configured to register |
| 20:25.33 | z4nD4R | Kobaz: big thx |
| 20:25.33 | oktay | but normally that would succeed yes? |
| 20:25.33 | Kobaz | if you take 5 seconds to google you'll find that 401 means unauthorized |
| 20:25.49 | Kobaz | oktay: why would you assume that anything would normally succeed |
| 20:25.58 | Kobaz | it depends on your specific configuration entirely |
| 20:26.24 | oktay | I know what 401 means. |
| 20:26.41 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 20:26.42 | Kobaz | the device is trying to register with something that's either not found, or doesn't match |
| 20:27.23 | oktay | something? |
| 20:27.27 | oktay | an extension? |
| 20:27.35 | Kobaz | a user |
| 20:27.43 | Kobaz | i think you need the book |
| 20:27.45 | Kobaz | ~book |
| 20:27.45 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 20:27.50 | jaytee | never used a Dlink ata but if the secret=password for the sip account in sip.conf doesn't equal the password for that account/line on the ata then you'll get a 401 |
| 20:27.59 | russellb | ~newbook |
| 20:27.59 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ |
| 20:28.18 | jaytee | yay, infobot has returned |
| 20:28.26 | jaytee | ~botsnack |
| 20:28.26 | infobot | :), jaytee |
| 20:28.26 | oktay | jaytee: i double checked that. I think I would see a FAIL message on the console in that scenario |
| 20:28.37 | Kobaz | oktay: depends on your asterisk debug/verbose level |
| 20:28.40 | jaytee | oktay, yes |
| 20:28.50 | jaytee | and what kobaz said |
| 20:30.04 | z4nD4R | Kobaz: you have idea how this file save in to places? |
| 20:32.31 | p3nguin | z4nd4r: The standard default location for recordings by MixMonitor() is in /var/spool/asterisk/monitor/ |
| 20:33.07 | p3nguin | If you do not like that path, provide a different full path as app data, or configure asterisk.conf according to your new needs. |
| 20:33.11 | z4nD4R | p3nguin: i know, i change it... but.. i want the recorded files stor in the 2 location .. |
| 20:33.43 | iscario | hi, i wanted to set up asterisk on a OpenBSD server, but i can't find asterisk-addon in the ports. Does it mean that i need to compile addons by myself to benefit from confbridge for exemple ? |
| 20:33.46 | Kobaz | z4nD4R: use the <command> part of MixMonitor |
| 20:33.49 | p3nguin | You're not going to record to two locations. You can record a file in one location and then copy it to a second location afterward if you want. |
| 20:34.10 | Kobaz | z4nD4R: cp or ln the file when the recording is finished, using the last option in MixMonitor |
| 20:34.18 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 20:34.40 | z4nD4R | Kobaz: this command start when recording is done? |
| 20:34.43 | Kobaz | yes |
| 20:34.51 | Kobaz | core show application MixMonitor |
| 20:35.24 | z4nD4R | Kobaz: ok .. w8 |
| 20:35.29 | p3nguin | weight? |
| 20:35.41 | p3nguin | weight is the wrong word. You probably mean wait. |
| 20:36.57 | z4nD4R | p3nguin: :D |
| 20:37.33 | p3nguin | How about something like this? MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.WAV,a,/bin/cp ${MIXMONITOR_FILENAME} /some/other/path/); |
| 20:39.28 | Kobaz | you would want ^{MIXMONITOR_FILENAME} |
| 20:39.48 | z4nD4R | p3nguin: that what i set.. now go i test it |
| 20:40.10 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-220.mobile.uci.edu) |
| 20:40.21 | Kobaz | i think the varset is delayed until it's finished, so you don't want immediate interpolation I think |
| 20:42.06 | z4nD4R | Kobaz: asterisk say: |
| 20:42.07 | z4nD4R | <PROTECTED> |
| 20:42.26 | Kobaz | heh |
| 20:42.43 | Kobaz | having callerid in the filename probably isn't the best of ideas |
| 20:43.15 | z4nD4R | Kobaz: but /var/www/1000 is clean |
| 20:44.12 | Kobaz | meaning? |
| 20:44.55 | z4nD4R | meaning this https://asterisk.4safety.cz/1000/ :D |
| 20:45.55 | oktay | would Asterisk show anywhere WHY it tried to send a 401 Unauthorized ? |
| 20:46.12 | Qwell | oktay: yes, in the logs. turn on debug and verbosity |
| 20:47.34 | oktay | what can I look for? |
| 20:47.40 | oktay | It shows a bunch of those messages but no reason |
| 20:49.11 | Qwell | post them somewhere for us to see |
| 20:50.38 | carrar | PICS!! |
| 20:50.57 | *** join/#asterisk sshock (~sshock@2002:3ff8:8553:1:20e:35ff:fed7:132e) |
| 20:52.21 | *** join/#asterisk VenomX (~venomx@201-0-183-3.dial-up.telesp.net.br) |
| 20:54.54 | VenomX | Hi. I'm new to asterisk and I just wonder: Should I use 1.6 or 1.8 right now? I'm confused about new feature releases for both versions ( I've read the wiki about LTS and Standard ) |
| 20:55.27 | Qwell | VenomX: if it's a new install, go with the latest release |
| 20:55.31 | oktay | is there a pastebin you can upload a file to? |
| 20:55.37 | Qwell | ~pb |
| 20:55.37 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 20:55.51 | Qwell | oktay: oh, a file. not that I know of. nobody here would download it. |
| 20:56.40 | oktay | Qwell: i meant upload to a pastebin like service |
| 20:56.42 | oktay | or email |
| 20:58.00 | oktay | http://paste.ubuntu.com/549984/ |
| 20:58.04 | oktay | ok. i think this shows enough |
| 21:00.07 | Qwell | your ATA is broken. it's not responding to the 401 correctly. |
| 21:00.29 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:01.19 | oktay | Could it be that it's not getting the 401? |
| 21:01.22 | VenomX | Geez... my CentOS box simply refuses to retrieve the digium/asterisk repo data O_o ( yum clean all/makecache not working ) |
| 21:01.38 | Qwell | oktay: sure, but then it's broken in a different way |
| 21:01.51 | Qwell | VenomX: got logs? |
| 21:02.20 | oktay | Qwell: What does Unauthorized mean in this context? I mean , what would the ATA do, if it did get the 401 ? |
| 21:02.38 | Qwell | it would respond with the credentials it's required to |
| 21:02.55 | oktay | oh. so the unauth is normal procedure ? |
| 21:03.02 | Qwell | yes |
| 21:03.21 | oktay | Thanks. It's probably a configuration thing. This did connect to another Asterisk box before. |
| 21:03.28 | oktay | I'll try to figure it out. |
| 21:03.57 | z4nD4R | Kobaz: it works fine.. BIG THANKS man |
| 21:07.06 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 21:07.15 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 21:07.54 | VenomX | Qwell: I'm putting some data together to pastebin it |
| 21:08.27 | *** join/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 21:09.48 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 21:12.23 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:13.47 | VenomX | Qwell: http://asterisk.pastey.net/144512 |
| 21:13.54 | *** part/#asterisk z4nD4R (~zandar@bband-dyn170.178-41-107.t-com.sk) |
| 21:15.46 | tzanger | wow, AMBE codec can get a MOS of 3.5 at 2kbps |
| 21:17.30 | iscario | trying one more time :) hello! i wanted to set up asterisk on a OpenBSD server, but i can't find asterisk-addon in the ports. Does it mean that i need to compile addons by myself to benefit from confbridge for exemple ? |
| 21:17.55 | pabelanger | iscario: most likely |
| 21:18.26 | VenomX | Qwell: If there is any bug/error, it's either CentOS' in general or my box only. My fedora desktop retrieved the repodata OK . I just wonder why the other repos are OK: centos-base, epel, rpmfusion... |
| 21:18.28 | iscario | that was just to be sure. thanks pabelanger |
| 21:19.16 | pabelanger | iscario: your best bet would be to contact the maintainer for OpenBSD, but if you cannot find it within the ports, my guess is it does not exist |
| 21:20.07 | iscario | pabelanger: that was what i thought... i'll compile it then^^ |
| 21:20.54 | *** part/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 21:21.23 | VenomX | Qwell: Sorry to bother you wit this. It is my box's fault. Thanks. ( Tried another centos box ) |
| 21:24.21 | VenomX | Truth be told, It was my fault. Shame on me :) |
| 21:30.23 | *** part/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 21:32.47 | oktay | Qwell: I changed a few things. Now instead of 401 messages, asterisk is trying to send 200 messages but still doesn't seem to make it to the ATA. Any ideas? |
| 21:33.46 | oktay | Via: SIP/2.0/UDP 92.45.152.39:5060;branch=z9hG4bK72edfb94fe1a20d5;received=92.45.152.39;rport=5060 |
| 21:33.56 | oktay | this should be the Private IP I guess. |
| 21:39.26 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 21:41.03 | *** join/#asterisk antiwire_ (~antiwire@unaffiliated/antiwire) |
| 21:42.05 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 21:43.25 | *** join/#asterisk ducky (~dnewell@pool-72-90-76-142.syrcny.fios.verizon.net) |
| 21:46.10 | p3nguin | NAT problems? |
| 21:46.57 | oktay | probably |
| 21:47.37 | p3nguin | Are the ATA and Asterisk on different networks? |
| 21:48.11 | oktay | Yes |
| 21:48.34 | p3nguin | Is Asterisk behind NAT? Is the ATA behind a different NAT? |
| 21:50.05 | oktay | Asterisk is NOT behind a NAT. The ATA is, but it makes no difference if I DMZ it. |
| 21:51.53 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 21:54.44 | p3nguin | That's ridiculous. |
| 21:55.05 | p3nguin | I wish people would shove DMZ where the sun doesn't shine. |
| 21:55.51 | dmz | uhh thanks :( |
| 21:55.59 | carrar | haha |
| 21:56.14 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 21:56.18 | oktay | it's a good way to rule out NAT and/or Firewall issues no? |
| 21:56.25 | oktay | I don't mean to keep it there. |
| 21:56.29 | oktay | Transmitting (no NAT) to 92.45.152.39:5060 |
| 21:56.36 | oktay | I have nat=yes for this extension |
| 21:56.48 | oktay | Why does it say 'no Nat' in the log? |
| 21:57.02 | dmz | i'd avoid NAT & * if at all possible; it kinda works but i've only had luck w/the phones being behind nat and the * box on direct ip |
| 21:57.26 | oktay | dmz: that is what I have now. |
| 21:58.00 | oktay | I would expect issues with no audio etc on one side, but this is failing to register |
| 21:58.31 | p3nguin | There is no nat setting for extensions, only sip devices/peers. |
| 21:59.16 | dmz | what kind of phone are you using? |
| 21:59.23 | dmz | and what codecs are enabled |
| 21:59.35 | dmz | i disable all and only enable *law |
| 21:59.39 | oktay | it's a DLINK SVG-2102 |
| 22:00.01 | oktay | i don't really know about the codecs. Let me see. |
| 22:00.04 | dmz | never tried that; i tend to use polycom :) |
| 22:00.15 | dmz | "you get what you pay for" especially in the voice quality |
| 22:00.22 | *** join/#asterisk CVance (46a781e5@gateway/web/freenode/ip.70.167.129.229) |
| 22:00.27 | oktay | I tend to buy what I can find. That means linksys spa or this |
| 22:00.29 | dmz | :) |
| 22:00.40 | dmz | so i missed it; what exactly is the problem your having? |
| 22:01.26 | oktay | I think asterisk is sending back a SIP/401 which is never making it to the ATA (or it's not behind properly handled). So Asterisk keeps sending 401s forever |
| 22:01.41 | oktay | <PROTECTED> |
| 22:01.50 | CVance | Hello, I am downloading asterisknow to install on my xen server, does it support paravirtualized or only HVM? Second if I have 10 seats in the office, would provisioning 1 core of my X5650 handle typical workloads? |
| 22:02.10 | oktay | dmz: would codecs even come into the picture during REG ? |
| 22:02.19 | CVance | Finally, sorry forgot to ask, how much ram should I give to the asterisknow server? |
| 22:03.48 | dmz | they come in during a call |
| 22:03.58 | dmz | you can debug the sip & it will show the codec setup |
| 22:04.06 | oktay | <--- Transmitting (NAT) to 92.45.152.39:5060 ---> |
| 22:04.19 | oktay | Via: SIP/2.0/UDP 92.45.152.39:5060;branch=z9hG4bK1fcf2369ab3c65d7;received=92.45.152.39;rport=5060 |
| 22:04.35 | oktay | Shouldn't the second IP be PRIVATE.? It is for my other ATA which works. |
| 22:04.37 | dmz | CVance, not sure about asterisknow but my * boxes have 1G |
| 22:05.14 | dmz | the private IP isn't in the SIP communicatoin; SIP is just tcp so it's connecting via the tcp packets; you can check your NAT gateway and see if it can NAT SIP |
| 22:05.14 | p3nguin | Both of mine have 512M system memory. |
| 22:05.21 | CVance | dmz: thanks, how much cpu horsepower would it need for 7 - 10 concurrent calls? |
| 22:05.41 | dmz | depends, are you changing codecs on the box? |
| 22:05.52 | p3nguin | Arch Linux on one, FreeBSD on the other, vanilla Asterisk (no GUI) on both |
| 22:05.53 | dmz | if you have different type phones & different voip provider; then the cpu has to convert it |
| 22:05.59 | dmz | if you force everything to ulaw it should help |
| 22:06.02 | oktay | dmz: thanks for your help. |
| 22:06.05 | p3nguin | SIP is UDP |
| 22:06.08 | dmz | we see no cpu on most of our usage |
| 22:06.08 | dmz | doh |
| 22:06.09 | dmz | sorry |
| 22:06.15 | dmz | :) |
| 22:06.28 | dmz | wonders why my fingers like to type "TCP" |
| 22:06.44 | CVance | dmz: thanks, i'll give it one and add another if it needs it, anyone running asterisknow as a PV guest as opposed to HVM? |
| 22:06.50 | p3nguin | My Asterisk uses about 40M memory and 10% or less CPU at any given moment. |
| 22:06.53 | dmz | CVance shout back if you want a quote on a digium box ;) |
| 22:07.11 | CVance | dmz: will do if I run back with my tail between my legs :P |
| 22:11.00 | CVance | How much space for a AsteriskNow distro? |
| 22:13.58 | p3nguin | I'd assume 4G is a safe bet for a fresh install. |
| 22:16.05 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 22:16.27 | p3nguin | There's a pretty good chance that the more-appropriate #AsteriskNOW channel could provide much better information pertaining to your inquiry. |
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| 22:30.17 | CVance | p3nguin: thanks, installing now |
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| 22:58.00 | syncer | hello |
| 22:58.02 | syncer | <PROTECTED> |
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