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| 03:34.16 | D30 | hi all, is it possible to set a gateway(DID) per extension for outbound calls?? |
| 03:35.21 | WIMPy | What does that mean? |
| 03:37.34 | D30 | like if your extension number will call for an international number, this specific DID will be used |
| 03:38.01 | WIMPy | First of all, extensions don't make calls. |
| 03:38.30 | WIMPy | And off course you can set callerid in your extensions for outbound calls. |
| 03:40.48 | D30 | WIMPy: i mean if you have an extension number and you are going to call using that extension number, you can set what DID to use |
| 03:41.11 | WIMPy | Your reversing things. |
| 03:41.30 | WIMPy | You call to an extension. |
| 03:41.30 | D30 | why is that so? |
| 03:41.43 | D30 | i think you misunderstood my question hehe |
| 03:41.49 | WIMPy | Like a pattern for international calls. |
| 03:43.13 | WIMPy | And without mre details I can't make any suggestions as to how to do it. |
| 03:43.34 | D30 | okay let me rephrase it.. |
| 03:44.41 | D30 | i had an extension number like 201, and I'm using that extension as my local extension.. its setup on a polycom phone... |
| 03:44.47 | WIMPy | Add some information on how you want to select caller id as well. |
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| 03:45.13 | WIMPy | Now that is a call TO that phone. |
| 03:45.44 | D30 | so i am making a call coming from my local extension to an international number like a US number, so i want that call to be routed to a specific DID.. |
| 03:46.05 | WIMPy | No. |
| 03:46.33 | WIMPy | You make a call from your phone to an extension which is an international number. |
| 03:46.52 | WIMPy | And DIDs are inward as well, as the name already says. |
| 03:47.21 | WIMPy | Unless by DID you mean the called number. |
| 03:47.44 | D30 | agh right..sorry for that :p |
| 03:48.16 | WIMPy | So in that extension you can set the callerID. |
| 03:48.23 | WIMPy | How depends on your needs. |
| 03:48.39 | D30 | yup yup.. like it would appear on the caller id on the other end |
| 03:49.10 | D30 | can that be setup in asterisk? and that can be change by some provider when it reaches the other end? |
| 03:49.45 | WIMPy | Some providers take the ID you send, others whatever they configured for your account. |
| 03:49.55 | D30 | ahhh |
| 03:50.11 | D30 | I'm using flowroute btw... |
| 03:50.25 | WIMPy | doesn't know them |
| 03:54.38 | D30 | hnnn anyways thanks.. will try to search it first :) |
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| 04:21.05 | Penguin | Flowroute will take what you send, as long as it is formed well. |
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| 06:00.51 | tbsky | Hi all! asterisk transcode makes me crazy :( hope someone can give me some hint. I want to use G722 to dial internal phone, and use G711 to dial outbound trunk. is that possible? |
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| 06:02.08 | tbsky | asterisk always transcode when I dial-out, unlees I force all calls to use G711 initially. or even when both legs are using G711, asterisk still trancode in the middle. |
| 06:02.58 | tbsky | like the crzay trancode below: |
| 06:03.04 | tbsky | NativeFormats: (alaw) |
| 06:03.04 | tbsky | <PROTECTED> |
| 06:03.04 | tbsky | <PROTECTED> |
| 06:03.04 | tbsky | <PROTECTED> |
| 06:03.04 | tbsky | <PROTECTED> |
| 06:03.58 | tbsky | although both size use alaw finally, asterisk transcode in the middle.. is there parameter in sip.conf to prevent this? |
| 06:04.06 | tbsky | s/size/site |
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| 06:08.27 | tbsky | I have tried asterisk 1.6.2, 1.8, 11. I remember in old days with asterisk 1.4, this seems not a problem, but maybe my memory is not correct. |
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| 07:01.19 | tbsky | tried asterisk 12.5. I have found a article which may fix this: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite . but the feature seems not here yet. maybe I should try asterisk 13 |
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| 07:26.01 | tbsky | 13.0 beta2 didn't notice SIP_CODEC in dialplan. maybe there is something new or something broken. |
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| 08:09.23 | luvmyh0nda | Good morning! Working on a simple IVR which reads a "system status" script via google TTS, and I want to offer the option to press 1 to go back to the IVR, or press 2 to transfer to a ring group, but I am not familiar enough with the terminology to know what to google :( I've tried the "Dial" command & a few different formats, but nothing seems to work - any advice? |
| 08:11.31 | ChannelZ | What part are you actually asking about? |
| 08:12.12 | luvmyh0nda | how to transfer to the ring group |
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| 08:12.17 | luvmyh0nda | sorry, should have been a bit more specific |
| 08:12.52 | ChannelZ | Well to dial multiple devices, you just Dial(SIP/Joe&SIP/Bob&SIP/Jane) etc |
| 08:13.17 | luvmyh0nda | the issue is that joe & bob and jane, may end up being steve & bill & don next week |
| 08:15.14 | luvmyh0nda | I did see that solution, and it works, but in order to change the dialplan, I would have to manually edit the code, which, for me is okay, but if I put someone else in charge who has more of a marmoset-level experience with asterisk, but works OK using a GUI, it's a bit more simple for them to swap out an extension than edit (and possibly break) the config file |
| 08:15.42 | ChannelZ | Well how you administer that is up to you then. You could make it a config variable, or have an AGI look it up through some external means, or use the AstDB to store it.. |
| 08:16.14 | luvmyh0nda | that's genius! I could use CURL |
| 08:16.37 | luvmyh0nda | to just return the portion that goes within Dial() |
| 08:16.48 | ChannelZ | Or you could use a custom queue and admin it that way by adding agents.. there's a dozen ways, just depends on how you want to integrate it |
| 08:17.36 | luvmyh0nda | ideally, I'd like to use the ring group, and then just switch over to that. Can I use Goto() to get to a ring group? |
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| 08:18.32 | luvmyh0nda | sorry, like i said, somewhat new to the terminology, by "switch over to" i mean jump to, or initialize |
| 08:19.13 | ChannelZ | Well I'm not sure what you mean, Asterisk doesn't have a native 'ring group' function per se |
| 08:19.45 | ChannelZ | Yes you can Goto to jump to any extension you want |
| 08:19.47 | luvmyh0nda | but wouldn't freepbx write it to a config file somewhere? and is there a way to execute the code associated with that? |
| 08:20.11 | luvmyh0nda | i'm fairly proficient with linux command line - about 15 years experience with programming in general & linux use, but only 2-3 of asterisk |
| 08:20.32 | ChannelZ | I have no idea about freepbx, it's a whole separate beast |
| 08:20.49 | luvmyh0nda | gotcha |
| 08:20.52 | ChannelZ | #freepbx is more applicable |
| 08:21.13 | ChannelZ | There might already be a plugin or module or whatever for it for all I know |
| 08:21.20 | luvmyh0nda | so i'll go peck around over there for the ring group question |
| 08:22.01 | luvmyh0nda | perhaps something a bit more suited to this channel: if they don't want to go to the ring group for "support" - how do I send them back to the IVR? I would think I would use Goto(ivr-name) |
| 08:22.15 | luvmyh0nda | well, goto (ivr-name,step,etc) |
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| 08:22.46 | ChannelZ | core show application goto |
| 08:23.04 | ChannelZ | for all the syntax. but more or less yes |
| 08:23.36 | ChannelZ | brb |
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| 08:31.38 | luvmyh0nda | welcome back! |
| 08:32.28 | ChannelZ | ty |
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| 08:35.33 | luvmyh0nda | ok - my IVR is defined as ivr-entry, shouldn't I hypothetically be able to use Goto(ivr-entry,s,1) to transfer back to the IVR? |
| 08:36.30 | ChannelZ | assuming ivr-entry is a context which has an s extension, yes |
| 08:39.32 | luvmyh0nda | i don't quite understand the 's extension' part :\ |
| 08:40.48 | ChannelZ | An extension is an arbitrary string of digits (or sometimes not) in the diaplan that you dial to make something happen.. play a sound, dial a device, enter voicemail, whatever |
| 08:42.41 | ChannelZ | So assuming you have an exten => s,1,whatever in your dialplan in the [ivr-entry] context, your Goto statement would work |
| 08:42.46 | luvmyh0nda | ok |
| 08:43.03 | luvmyh0nda | i see what it did, and probably why it failed. I named it ivr-entry, but it's actually called ivr-1 |
| 08:43.17 | luvmyh0nda | so i am going to the wrong (or non-existant) ivr |
| 08:43.56 | ChannelZ | well like I said, 'assuming it exists...' :) |
| 08:44.39 | luvmyh0nda | eureka! |
| 08:45.27 | luvmyh0nda | is there a way to read input while executing an agi? |
| 08:47.13 | ChannelZ | Yes-ish. AGIs just control the dialplan via external script, so any of the dialplan applications that do such things are still applicable.. Background, Read, or just being in a context that hasn't timed out yet.. |
| 08:48.41 | ChannelZ | (I assumed by "read input" you meant the user pressing digits) |
| 08:49.05 | luvmyh0nda | yes, Read() |
| 08:50.03 | luvmyh0nda | so I can Background(agi(tts.agi,"text to be read")), followed by Read(var,,#ofdigits) |
| 08:50.49 | ChannelZ | Mmm, no. |
| 08:51.06 | luvmyh0nda | would I have to put the background in the tts.agi file? |
| 08:51.45 | ChannelZ | Yes, Background plays files |
| 08:52.57 | luvmyh0nda | oh.. yeah.. it didn't like that whatsoever :\ |
| 08:53.00 | ChannelZ | Presumably your tts.agi does something and generates an audio file somewhere, which you can then use Background to play |
| 08:55.34 | ChannelZ | whether or not you send a command through the AGI to Asterisk to do the Background/Read or your AGI just passes the filename back as a channel variable or something and you do the rest of the work in the dialplan is up to you |
| 08:56.12 | luvmyh0nda | are command names case-sensitive? |
| 08:56.58 | ChannelZ | dialplan applications, no |
| 08:57.27 | ChannelZ | past my bedtime, gotta go |
| 08:57.38 | luvmyh0nda | same here. you have a paypal? |
| 08:58.43 | ChannelZ | no, but not necessary even if I did |
| 08:59.12 | luvmyh0nda | ah, i'd totally buy you a cup of coffee lol |
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| 08:59.58 | ChannelZ | heh no worries. I answer what and where I can |
| 09:00.09 | ChannelZ | s/what/when/ |
| 09:00.16 | luvmyh0nda | cool, thank you. i do the same for php, sql and jquery |
| 09:00.48 | luvmyh0nda | i write cloud based health record software for a living, and tinker with this stuff for "fun" |
| 09:00.52 | luvmyh0nda | what a world LOL |
| 09:01.38 | ChannelZ | Not unusual. Anyway good luck, have fun |
| 09:01.46 | luvmyh0nda | thank you, have a good night! |
| 09:01.48 | luvmyh0nda | well, morning |
| 09:01.53 | ChannelZ | you too |
| 09:01.59 | ChannelZ | ZzzZzzzZzz.. |
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| 09:40.48 | mfauzir | hi all |
| 09:43.26 | mfauzir | need help please |
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| 09:43.49 | mfauzir | i have installed asterisk with TDM410 ( 2 FXS, 2 FXO) |
| 09:45.10 | mfauzir | and i have doing what everything instruction to configure DAHDI |
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| 09:47.52 | mfauzir | what the problem is when i try to outbound calling, |
| 09:50.06 | mfauzir | the receiver has answered the call, but the asterisk still ringing and not hear in the extension user (caller) |
| 09:52.04 | mfauzir | does anyone know what should i do to solve my problem ? |
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| 10:29.19 | youjelly | Hi guys, can anyone give me start on how to develope IVVR solutions on opensource |
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| 12:42.10 | Stefan27 | hi americans |
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| 12:45.32 | [TK]D-Fender | What about the rest of us? |
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| 12:49.12 | marceloamorim | morning guys, do you thing that I should use this for tell me who is hangup? |
| 12:49.24 | marceloamorim | at this moment I`m using hangup handlers |
| 12:49.56 | marceloamorim | but I didn`t find another way to do this, so I tried used the options "g" for app Dial |
| 12:50.46 | marceloamorim | so if the next Set is being set and at hangup handlers show me that variable, it means the callee hangup the call |
| 12:51.11 | seanbright | i'm stuck on the fact that you are using a backtick as an apostrophe... |
| 12:51.16 | marceloamorim | what do you think about it? should I keep it? or should I find another way |
| 12:52.03 | marceloamorim | o.O |
| 12:52.11 | marceloamorim | bad configuration for keyboard |
| 12:54.56 | [TK]D-Fender | marceloamorim: "core show application dial" <- |
| 12:55.31 | marceloamorim | [TK]D-Fender: I`m working on it like a week |
| 12:55.47 | marceloamorim | the other ways that I use this, isn`t work well |
| 12:55.57 | marceloamorim | the better option is using the hangup handlers |
| 12:56.06 | marceloamorim | this new feature for asterisk 11 |
| 12:56.38 | marceloamorim | I did this using another ways, but there was other complications using another methods, I need hangup handlers |
| 12:56.49 | [TK]D-Fender | marceloamorim: I didn't say that you should not be using Hangup Handlers. |
| 12:57.02 | [TK]D-Fender | marceloamorim: If you want to know who hung up.... read Dial's instructions. |
| 12:57.42 | marceloamorim | I read and read again this dial instructions |
| 12:58.33 | marceloamorim | but I`ll check again |
| 12:58.35 | marceloamorim | ty |
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| 13:22.33 | adsc | I have a problem that sometimes incoming sip call gets rerouted to call itself, and I think it might be due to my peer configurations...would be glad if someone could take a look at it: http://pastie.org/9590732 |
| 13:23.15 | adsc | my guess is because the host from the [twilio] context is the same as the one from the [twilioip-2] context |
| 13:24.18 | [TK]D-Fender | adsc: What "incoming sip call"? |
| 13:24.25 | [TK]D-Fender | adsc: You haven't shown us the call or the dialplan |
| 13:24.32 | [TK]D-Fender | adsc: sip.conf is not "call processing" |
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| 13:25.05 | adsc | my understanding was that when something is incoming, it first selects a matching context from sip.conf |
| 13:25.22 | adsc | then uses the context defined there to look up the dialplan |
| 13:25.30 | [TK]D-Fender | adsc: it finds a matching DEVICE ENTRY |
| 13:25.56 | [TK]D-Fender | adsc: don't call things like your [twilio] in there a "context" |
| 13:26.07 | adsc | yeah, thanks, that might be confusing |
| 13:26.50 | [TK]D-Fender | adsc: once a match is found and authed it sends the call into the context specified in the entry |
| 13:27.04 | adsc | yes |
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| 13:28.57 | marceloamorim | [TK]D-Fender: I just find "g" and "F" which is practically the same |
| 13:29.12 | adsc | I have appended the dialplan here and will now try to capture a failing call: http://pastie.org/9590741 |
| 13:29.54 | [TK]D-Fender | adsc: exten => _X!,1,Dial(SIP/phone1) <--- what is the point of this? |
| 13:30.03 | [TK]D-Fender | adsc: where ANY dialed number only dials that ONE device? |
| 13:31.05 | adsc | all incoming calls to phone1 |
| 13:31.57 | [TK]D-Fender | adsc: You are clearly not paying attention to where you are sending your calls. |
| 13:32.16 | [TK]D-Fender | adsc: Go look at that again. |
| 13:32.32 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:34.01 | adsc | can you be more specific? the way I understand it is that incoming calls always use the context [fromtwilio], which means someone who dialed 41315280456 gets connected to phone2 and everyone else to phone1 |
| 13:34.39 | adsc | also, outgoing calls always use context [twilio], which simply always connects with ${EXTEN}@twilio |
| 13:34.57 | [TK]D-Fender | Go get your call |
| 13:34.59 | *** join/#asterisk sgriepentrog (~sgriepent@162.17.68.211) |
| 13:35.00 | adsc | okay |
| 13:36.21 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 13:37.39 | adsc | there is so much spam with sip debug on |
| 13:39.42 | marceloamorim | I already used this F, works pretty well but I had an issue with ${DIALSTATUS} when I used this option |
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| 13:40.39 | *** join/#asterisk sgriepentrog (~sgriepent@162.17.68.211) |
| 13:42.38 | adsc | this is a call that worked: http://pastie.org/9590767 |
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| 13:46.50 | adsc | here is one that didn't work: http://pastie.org/9590776 |
| 13:48.06 | adsc | it says Found peer 'twilio' for '+41799445543' from 107.21.231.147 in the second sip packet |
| 13:48.59 | adsc | whereas in the one that works it finds one of the correct ones |
| 13:49.11 | adsc | it seems my original assumption was correct |
| 13:49.48 | adsc | because one of the possible incoming ips matches the one in the "host" setting of the outgoing context, it chooses this one occasionally |
| 13:50.05 | adsc | how can I solve this? |
| 13:50.23 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 13:50.39 | adsc | i meant of the outgoing device, sorry |
| 13:51.27 | adsc | is it possible to define a device for outgoing use only, so that it couldn't select it on incoming? |
| 13:54.35 | [TK]D-Fender | adsc: There is no such thing as a "context" for outgoing |
| 13:54.48 | [TK]D-Fender | When you call a device the "context" sepcified in it's definition means nothing |
| 13:55.00 | [TK]D-Fender | "context=" is for calls FROM the device, not TO it |
| 13:55.35 | [TK]D-Fender | ALL of your twilio device definitions should be pointing to the same context |
| 13:56.12 | adsc | I see |
| 13:56.43 | adsc | and I would then determine in the dialplan to either dial out or in? |
| 13:58.00 | adsc | maybe you need some background information |
| 13:58.14 | [TK]D-Fender | no |
| 13:58.22 | [TK]D-Fender | there is no such thing as a "dial out" |
| 13:58.33 | [TK]D-Fender | "context=" is used for calls FROM it |
| 13:59.00 | adsc | that is why I have two different twilio devices |
| 13:59.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 14:01.31 | adsc | I really appreciate all your explanations, but my concrete problem is that the device that should be used for outgoing calls also matches some of the incoming calls because of the same IP |
| 14:02.16 | adsc | maybe it is enough if I change the order |
| 14:02.42 | *** join/#asterisk calum_ (~calum_@host81-156-24-144.range81-156.btcentralplus.com) |
| 14:05.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 14:05.42 | [TK]D-Fender | gah |
| 14:06.40 | *** join/#asterisk pigpen (~pigpen@216-177-181-17.block0.gvtc.com) |
| 14:06.52 | adsc | yeah, does the trick |
| 14:07.08 | pigpen | hey all. It has been awhile. |
| 14:07.10 | adsc | it now always matches the "incoming" device |
| 14:07.24 | adsc | and the outgoing is only used from the dialplan |
| 14:07.36 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-34-104.user.veloxzone.com.br) |
| 14:07.47 | pigpen | I have a 11.12.1 box with a pri. I upgraded from 11.10.x if I remember correctly. |
| 14:07.53 | adsc | anyway, thanks for all the explanations, it really helps me in the long run |
| 14:08.44 | pigpen | I get no âringâ indications on inbound dialing. I have tried to force the âr([us])" as part of the dialplan, but only get "Unable to start playtone" |
| 14:09.22 | pigpen | indications.conf is present and looks fine, diff'ed the "us" section against previous versions, no change there. |
| 14:09.24 | [TK]D-Fender | pigpen: Verify that you have a proper indications.conf and your zones set properly |
| 14:10.12 | pigpen | yeah, the [us] section is identical to 11.11.0 |
| 14:10.22 | pigpen | general just sets the default to âusâ |
| 14:10.51 | pigpen | We upgraded just because we could, and adding in the res_fax_digium |
| 14:11.02 | pigpen | didnât have the issue prior. (naturally) |
| 14:11.14 | pigpen | Oh, and thanks TK, it has been awhile. |
| 14:11.21 | [TK]D-Fender | pigpen: yup. |
| 14:11.24 | *** join/#asterisk shadej (~bzegeju@213.55.73.146) |
| 14:11.53 | [TK]D-Fender | pigpen: go grab a call with PRI debug enabled. |
| 14:12.02 | pigpen | very odd issue. typically forcing ârâ takes care of it. I need to test outbound too. |
| 14:12.03 | pigpen | k. |
| 14:12.11 | pigpen | iâll be back in a few. |
| 14:24.21 | pigpen | [TK]D-Fender: any particular level? |
| 14:26.48 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-tkomddtouxsknplq) |
| 14:28.14 | sekil | how can one tell asterisk to force outboundproxy in peer? |
| 14:28.27 | sekil | not to use non-resolvable host |
| 14:29.38 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-maskgcbogigywtmw) |
| 14:29.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:34.52 | pigpen | [TK]D-Fender: http://pastebin.com/0YdT7irD |
| 14:47.16 | [TK]D-Fender | pigpen: [Sep 24 09:26:32] WARNING[24449][C-0000006c]: app_dial.c:2049 dial_handle_playtones: Unable to start playtone 'us' |
| 14:47.34 | [TK]D-Fender | pigpen: If you're going to force ringing, you should probably be answering the channel explicitly |
| 14:52.27 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-qzkxufmymkvbxtcs) |
| 14:57.34 | pigpen | [TK]D-Fender: Yeah, it was that way before I attempted this. |
| 14:57.44 | pigpen | Iâll run one without. |
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| 14:59.41 | [TK]D-Fender | You should alos force some audio on to it to ensure that the channel is brough up. Like 1s silence or something similar. |
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| 15:06.09 | pigpen | [TK]D-Fender: k. |
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| 15:14.03 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 15:14.33 | pigpen | [TK]D-Fender: http://pastebin.com/tykLGBTj |
| 15:15.04 | pigpen | so this is with answer, then forcing ring |
| 15:15.35 | pigpen | no ring tone still. |
| 15:15.43 | pigpen | Iâll do another playing silence |
| 15:15.50 | pigpen | right after the answer |
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| 15:18.30 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:19.37 | pigpen | http://pastebin.com/Ncp4VCmN |
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| 15:20.09 | pigpen | ^^ this one is with playback of silence, still with answer and forcing the ring. Next Iâll try to remove the answer and the force ring, leaving the playback of the silence. |
| 15:22.14 | pigpen | yeah, same ting |
| 15:22.19 | pigpen | s/ting/thing |
| 15:22.39 | [TK]D-Fender | Not sure at this point... |
| 15:22.50 | pigpen | [TK]D-Fender: yeah. me too. |
| 15:23.50 | *** join/#asterisk jwr__ (~quassel@205.196.167.54) |
| 15:24.55 | pigpen | Iâll see about posssibly downgrading to 11.12.0 and see what happens. If it were a simple setting, the problem would still exist. if things are resolved, it is a code issue. |
| 15:25.14 | pigpen | we were on 11.10.something, and all was fine. |
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| 15:29.08 | pigpen | [TK]D-Fender: I donât remember if I unload the dahdi modules or not when we upgraded, Iâll unload and re-modprobe them in a bit. who knows. |
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| 15:47.20 | *** join/#asterisk bananapie (~pumpkinpi@207.134.195.126) |
| 15:48.07 | bananapie | Hello. I can reliably cause asterisk to crash by calling a MEETME with the right parameters after the user hungup. Given that I should not be calling MEETME after a user hangs up, should I open a bug report ? |
| 15:49.11 | [TK]D-Fender | I suppose the fact it actually crashes * is still a bug. |
| 15:49.34 | [TK]D-Fender | However you would also be advised to to configure your system against the rules. |
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| 16:04.55 | bananapie | Yea, I will be auditing all my agi scripts. But it still crashed Asterisk. If my AGI misbehaves, worst case scenario asterisk should not crash. |
| 16:05.26 | mjordan | please do open a bug report. What version of Asterisk are you running against? |
| 16:05.54 | bananapie | 11.12.1 and latest 1.8 cert |
| 16:06.10 | bananapie | One of the messages that appears in the console is '[2014-09-24 16:05:29] WARNING[24917]: file.c:1241 ast_writefile: No such format 'pG' |
| 16:06.10 | bananapie | ' |
| 16:06.30 | bananapie | in the quotes, it's random characters, sometimes non printable characters. It's like if C were to be reading beyond the string it should be reading. |
| 16:06.35 | mjordan | a backtrace will be needed. |
| 16:06.52 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 16:07.40 | bananapie | ok. |
| 16:07.45 | bananapie | I'm going to eat lunch |
| 16:07.47 | bananapie | and file it after lunch |
| 16:09.04 | justdave | ~nat |
| 16:09.05 | infobot | from memory, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 16:09.57 | justdave | heh "see docs" thought that used to have a url in it to said docs, was just looking for a shortcut to click :) off to browse for it then |
| 16:10.41 | *** join/#asterisk apb1963 (~apb@107-146-213-112.res.bhn.net) |
| 16:13.21 | justdave | isn't nat=yes supposed to make it us comedia to set the destination IP and ignore what the client sends it? |
| 16:13.40 | justdave | (for SIP) |
| 16:14.07 | Qwell | justdave: Best to read sip.conf.sample |
| 16:14.13 | Qwell | It's complicated. |
| 16:14.16 | justdave | Qwell: that's what I'm reading |
| 16:16.33 | justdave | I've got two asterisk servers, both in the same network configuration. Both are behind a static NAT with port forwarding. I have a client also behind a different NAT. The client is configured identically for both servers. |
| 16:16.38 | justdave | I get audio on one server and not on the other |
| 16:17.26 | justdave | the sip.conf on both servers is almost identical (the only difference being the externip and localnet settings - which should be different because they have different public IPs) |
| 16:19.51 | justdave | wishes all phones used IAX ;) |
| 16:20.31 | justdave | packet trace eon the non-working server shows asterisk is sending the RTP traffic to the client's internal IP address from behind its NAT |
| 16:20.57 | justdave | which sounds broken because nat=yes I thought was supposed to make it ignore that and send the traffic to the IP it came from instead |
| 16:21.42 | file | if it received it, yes |
| 16:22.29 | justdave | if it received what? |
| 16:23.33 | justdave | packet trace showed outbound with the correct external IP in the SDP going back to the client. |
| 16:23.40 | file | if it received an RTP packet from the client |
| 16:23.46 | justdave | guess I should trace on the client end, too, and make sure it's still that IP when it gets there |
| 16:23.54 | justdave | maybe we have an interfering SIP ALG somewhere |
| 16:24.02 | file | if it has not received an RTP packet then it can not know where to send media |
| 16:24.42 | Qwell | Have any ALGs in the history of ever actually helped in any way? |
| 16:25.32 | justdave | nope, first thing we do in every office we set up is disable the ALG on the firewall :) |
| 16:25.41 | Qwell | I suppose the ones that don't live on cable modems and bring down your Internet connection when they see a well formed packet are a step ahead of the game. |
| 16:25.44 | Qwell | glares at 2wire |
| 16:27.36 | justdave | so it doesn't use the source IP of the SIP to set where to send the RTP to? |
| 16:28.16 | file | the source IP is only part of the equation, you still need the port |
| 16:28.35 | file | won't know that until you get a packet from it |
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| 16:32.08 | justdave | ok, I see outbound RTP from the client going back to the correct IP on the server end... and said traffic is never making it to the server. |
| 16:32.14 | justdave | pokes his netops guys |
| 16:32.48 | justdave | so that'd be the problem then, rtp never makes it there to re-point the other direction |
| 16:33.00 | file | correct |
| 16:33.25 | justdave | obviously the rtp port range didn't get forwarded correctly |
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| 16:51.35 | justdave | yep, that was the issue. They had the RTP rule missing from the firewall config |
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| 17:43.12 | Dovid | http://community.redhat.com/blog/2014/09/critical-bash-security-vulnerability-update-your-systems-today/ |
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| 18:16.14 | WIMPy | And where is the secutity issue with that? |
| 18:21.16 | Qwell | WIMPy: Think something along the lines of a git connection via ssh, where it's restricted to the git command. |
| 18:22.15 | WIMPy | And where would you be able to set environment variables without the ability to just execute whatever you wanted directly? |
| 18:22.16 | Qwell | env x='() { :;}; echo TEST' git pull |
| 18:22.17 | Qwell | TEST |
| 18:22.17 | Qwell | Already up-to-date. |
| 18:23.18 | Qwell | well, no, that's not valid, but w/e |
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| 21:17.42 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.0 (2014/09/24), 1.8.31.0 (2014/09/24); Standard: Asterisk 12.6.0 (2014/09/24); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 21:32.53 | *** join/#asterisk adium (~adium@unaffiliated/adium) |
| 21:55.22 | adium | Hello, I'm doing some tests on DDoS attacks on VoIPs more focused on SIP systems for a paper, on the simulations it's great but I don't know how to use Asterisk (or any server in that matter) to read incoming messages after my defense is set. |
| 21:55.57 | adium | I tried reading wikis and everything just saw a way to make Asterisk read on both ports, in that way the server will still be flooded with the DDoS |
| 21:56.25 | adium | Could any of you guys help me? Really appreciate it |
| 21:57.54 | WIMPy | "both ports"? What are you talking about? |
| 21:58.34 | adium | I need the incoming to go through my defense |
| 21:58.42 | adium | To test it |
| 21:59.03 | adium | So the defense will hear everything from the default port and then will send the "OK" ones to the Asterisk server |
| 21:59.13 | adium | And then I'll measure the effect of my defense |
| 21:59.27 | WIMPy | Ok, and what's the question then? |
| 22:03.17 | adium | I want the Asterisk server to hear on only one port that is not the default one. |
| 22:03.23 | adium | Is there a way to do that? |
| 22:04.28 | WIMPy | Configure the port, just as you would do anywhere. |
| 22:08.15 | adium | Really? I tried a lot to change the port and couldn't find it |
| 22:09.30 | WIMPy | See *bindaddr |
| 22:15.01 | adium | ty a lot, WIMPy |
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| 22:56.18 | *** join/#asterisk MasterChen (~chen@chenb0x.net) |
| 22:56.21 | MasterChen | hello everyone |
| 22:56.39 | MasterChen | I am currently working on a Polycom SoundStation IP 4000 conference phone |
| 22:57.06 | MasterChen | it grabs boot configs, etc, but it won't register to the PBX no matter what setting i toggle |
| 22:57.20 | MasterChen | not on a vlan... provisioning server is the same IP as the PBX |
| 22:58.22 | MasterChen | it's asterisk 1.2.24 with FreePBX 2.7.0 |
| 22:58.40 | MasterChen | I've been working on this for a while and now I am just at a loss |
| 23:05.54 | [TK]D-Fender | pastebin your configs for it |
| 23:05.57 | [TK]D-Fender | ~pb |
| 23:05.58 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:06.46 | MasterChen | which config? for the peer or ? |
| 23:06.50 | [TK]D-Fender | the phone |
| 23:07.31 | MasterChen | k sec |
| 23:11.35 | MasterChen | http://pastebin.com/WN2wjRVW |
| 23:11.46 | MasterChen | thatis just the SIP peer in sip.conf |
| 23:11.49 | MasterChen | extension 219 |
| 23:12.12 | [TK]D-Fender | PHONE configs for provisioning.... |
| 23:12.42 | MasterChen | oh |
| 23:12.59 | MasterChen | um.... is that a cfg file? |
| 23:13.16 | [TK]D-Fender | yes. |
| 23:13.22 | [TK]D-Fender | You said you were provisioning them |
| 23:13.28 | MasterChen | k. I'm still learning the ropes |
| 23:15.33 | MasterChen | http://pastebin.com/uG0bzpPh this is the <mac>.cfg file |
| 23:15.44 | MasterChen | let me also paste the extension xml file |
| 23:18.20 | MasterChen | http://pastebin.com/B6yaBQkC and here is the ext_219.cfg file |
| 23:22.31 | *** join/#asterisk Skten (~Sktenau@monitor.ic.ipera.net.au) |
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| 23:24.40 | [TK]D-Fender | Describe the networking between your phone & server |
| 23:24.48 | [TK]D-Fender | in proper detail |
| 23:26.23 | MasterChen | okay so... |
| 23:27.33 | MasterChen | the server is on-site at the client's location |
| 23:28.25 | MasterChen | it has a public IP, and it appears that the Polycom provisioning server is on the PBX as well |
| 23:28.57 | MasterChen | it is behind the client's firewall, but i have ssh access. All phones are on the intranet of the client |
| 23:29.04 | MasterChen | it's not a very big network |
| 23:29.16 | MasterChen | I'd say about 10 phones max |
| 23:30.41 | MasterChen | now, i am currently trying to register this phone from the outside, I am currently at my office desk with the phone, but the phone is set to pull the configs from the client's server witht he public facing IP |
| 23:30.56 | MasterChen | It pulls the configs.... runs sip.ld....just no registration |
| 23:40.41 | [TK]D-Fender | go look on the phone itself for the server it is trying to register to |
| 23:41.12 | [TK]D-Fender | Then you need to confirm exactly how the server gets to the internet, verify open/forwarded ports, etc |
| 23:41.20 | [TK]D-Fender | then look on CLI directly for SIP traffic |
| 23:41.37 | [TK]D-Fender | you should also be checking on YOUR side to make sure your network is routing them properly |
| 23:42.03 | *** join/#asterisk Cynagen (~cynagen@ip72-208-60-104.ph.ph.cox.net) |
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| 23:43.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:50.26 | MasterChen | Thank you TK. I will look into that asap |
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