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| 00:17.03 | carrar | mmm agi |
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| 00:46.24 | orioni | i have defined a trunk in asterisk and when i dial an outgoing number it says all line busy |
| 00:46.35 | orioni | if i do the same with a softphone it works ok |
| 00:46.49 | orioni | the registration is OK on asterisk |
| 00:46.58 | orioni | and the dialplan patterns match ok |
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| 00:47.51 | orioni | where might be the problem ? |
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| 00:55.12 | p3nguin | Whers's the sip debug and core verbose output? |
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| 01:08.42 | ManxPower | Sounds like you are using a GUI. |
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| 01:10.40 | cdose1 | hello everyone, I'm new to asterisk, reading the book right now, but i just compiled and installed asterisk 1.6, the book covers 1.4. what's the best resource for learning how to configure 1.6? can I follow the book loosely, or has too much changed? |
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| 01:40.51 | cdose1 | why do i get "No such command 'dialplan reload'" in the CLI? |
| 01:41.41 | WIMPy | * too old? |
| 01:45.02 | cdose1 | it's 1.6.2.7 |
| 01:45.09 | cdose1 | i do believe that's rather new |
| 01:45.36 | WIMPy | So there is a too new as well? |
| 01:46.40 | cdose1 | lol |
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| 01:54.08 | p3nguin | cdose1: I guess pbx_config probably isn't loaded. |
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| 01:56.16 | cdose1 | p3nguin, hey thanks, that worked. i think i'm starting to get the hang of this... |
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| 01:56.45 | p3nguin | If that problem persists, make sure you force it to load in modules.conf. |
| 01:58.46 | puckett_jw | I have a question about some unexpected results when using a / in a family or key in the database. Is there someone that point me in the correct direction? |
| 01:59.57 | puckett_jw | I am looking for documentation on valid characters in the database |
| 02:00.10 | cdose1 | p3nguin, thanks, the problem was that i only had a modules.conf.default. this is a brand new install, i'm trying to learn how to configure it. could you possibly point me to any good resources on configuring 1.6? i'm reading the book on 1.4, but i figure it's slightly outdated now... |
| 02:00.42 | p3nguin | It will be similar. |
| 02:02.40 | p3nguin | You'll need to create a conf for each component that you want to use. .conf.default files are just for your reference and generally are not working examples. |
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| 02:03.00 | cdose1 | p3nguin, yeah that's what I thought. the main thing I'm worried about though is differences between zaptel and dahdi, since the book seems to have been written before zaptel was renamed to dahdi... |
| 02:03.16 | p3nguin | It works mostly the same. |
| 02:04.36 | cdose1 | ok |
| 02:04.59 | p3nguin | Are you going to be hooking up to the PSTN, anyway? |
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| 02:06.41 | cdose1 | yeah i am |
| 02:07.04 | p3nguin | What method of connection will you be using? |
| 02:08.08 | cdose1 | i have an x100p clone fxo card |
| 02:08.25 | p3nguin | alright |
| 02:08.33 | p3nguin | You'll be using dahdi, then. |
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| 02:17.36 | ManxPower | "Great work David, but this only needs one Dial() option. I believe the 't' option is the one you are looking for here. Same goes for the blond transfer test." |
| 02:17.58 | ManxPower | tries to come up with a Blond Transfer Test |
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| 02:19.07 | coppice | is Blond Transfer what happened when I had children? If so, it failed. They have dark hair |
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| 03:59.05 | spenguin[work] | hey after editing fax.conf do I need anything more than a dialplan reload? |
| 04:00.27 | spenguin[work] | eh ignore |
| 04:00.49 | spenguin[work] | its just part of the dialplan - fax.conf |
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| 04:33.00 | Gambit- | evenin' chaps |
| 04:33.41 | Gambit- | Anyone awake particularly familiar with SIP and can tell me some details about it's connection behaviors? |
| 04:34.21 | p3nguin | What, exactly, about it is connection behaviors do you want to know? |
| 04:35.26 | Gambit- | Does the SIP protocol allow for the connection used to do the SIP signaling to also be used for end-to-end throughput? |
| 04:35.40 | Gambit- | or do you have to, in terms of the protocol, use a separate connection? |
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| 04:36.53 | Gambit- | p3nguin, sufficient details? |
| 04:37.32 | p3nguin | Are you talking about media? |
| 04:38.08 | p3nguin | The media stream (carried over RTP) is separate from SIP, if that's what you mean. |
| 04:39.21 | Gambit- | Well you don't technically /have/ to use RTP with SIP, that's just what's commonly used with media. I'm trying to figure out if SIP supports an end-to-end tcp proxy mode, where the SIP proxy you exchange dialing information with opens up a TCP connection on the other side and forwards traffic through until a BYE message. |
| 04:40.24 | spenguin[work] | is there a way I can test fax online? |
| 04:42.23 | spenguin[work] | nvm |
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| 05:08.01 | ManxPower | Gambit-, The SIP RFC says it can be UDP or TCP. It's just that nobody except Microsoft supports SIP/TCP (the require it in fact). |
| 05:08.24 | Gambit- | ManxPower, Does it /have/ to be a separate connection, however? |
| 05:08.29 | Gambit- | wait |
| 05:08.32 | Gambit- | let me rephrase that |
| 05:08.44 | ManxPower | separate from what? |
| 05:09.14 | Gambit- | The SIP exchange can happen over TCP or UDP. I'm trying to figure out if I can then exchange data between the two endpoints within the same TCP or UDP connection without horribly violating protocol. |
| 05:09.54 | ManxPower | Gambit-, you mean transport audio? |
| 05:10.02 | Gambit- | I want to use the SIP signalling to notify a proxy who I'd like to talk to, and the proxy will then connect the tcp streams and forward them across and step out of the way -- that's how I'd /like/ it to work, but I don't know how far off base I am. |
| 05:10.16 | Gambit- | I'm actually not transporting audio, it's a completely different protocol that's not actually related to VOIP at all. |
| 05:10.50 | ManxPower | in theory I believe that can be done. See OpenSER/SER/somesillynametheyforked |
| 05:11.23 | Gambit- | Ah, "Kamailio" :) |
| 05:11.28 | Gambit- | I'll take a look at that. |
| 05:11.34 | ManxPower | you are not going to transport blocks of data that way. At most you could send NOTIFY packets |
| 05:11.41 | Gambit- | Hm. |
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| 05:12.00 | ManxPower | So no you are not going to tunnel thru your corporate firewall using SIP. |
| 05:12.01 | Gambit- | I was really hoping to be able to use existing SIP infrastructure w/o having to open multiple connections. |
| 05:12.34 | ManxPower | Gambit-, read the SIP RFCs |
| 05:12.41 | [TK]D-Fender | Gambit-: Considering how many devices use TCP (few), you should refrain from saying "connection" in reference to a STATELESS protocol |
| 05:12.45 | Gambit- | I can, but they're *huge*. |
| 05:13.12 | Gambit- | [TK]D-Fender, I'm intentionally saying TCP and connection because that's what I want the resulting data channel to use, but understood. |
| 05:13.20 | [TK]D-Fender | Gambit-: And of course you'll have to have the gateway care about this signalling which is more work.. and then make it look like it even came from the device in the first place. |
| 05:13.51 | Gambit- | well, I often wire tcp connections together at my work, so I guess that's not something I think very much about, but that's true, most applications don't do that. |
| 05:14.24 | [TK]D-Fender | Gambit-: Perhaps you shouldn't be turning a wicker basket into an airplane, but rather be starting with something a lot closer... |
| 05:14.34 | Gambit- | chuckles |
| 05:14.55 | [TK]D-Fender | On that note... bed-time. Later all. |
| 05:15.15 | Gambit- | I don't disagree with him, but I would have been curious if he had any suggestions. |
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| 05:19.17 | Naikrovek | Gambit-: if you want to see how SIP works, read Asterisk source code or read the SIP RFC |
| 05:19.41 | Naikrovek | as i understand it it's pretty clear what can and can't be done. |
| 05:19.59 | Naikrovek | you can use SIP over TCP, yes. but virtually no one does this |
| 05:20.17 | p3nguin | What is the technical reason that no one does it? |
| 05:20.25 | Naikrovek | i don't know that there is one |
| 05:20.39 | Naikrovek | but UDP makes more sense based on the time-sensitive nature of the RTP |
| 05:20.51 | spenguin[work] | hopes to be a sipmeister someday |
| 05:21.00 | Naikrovek | (if a packet is lost, it won't help if TCP retransmit can get it back) |
| 05:21.08 | Gambit- | It sounds like it's not possible to overload the SIP channel, though, to send data as well. |
| 05:21.22 | Naikrovek | Gambit-: audio IS data |
| 05:21.59 | Naikrovek | if you're looking for a way to sneak data around on the sly, do it via ICMP |
| 05:22.09 | Naikrovek | there are already implementations of that |
| 05:22.14 | Gambit- | Yes, but ignoring for a moment the UDP vs TCP thing, doesn't the SIP negotiation establish the identify of the endpoints to allow them to then send data between eachother via whatever other means they choose? |
| 05:22.32 | Gambit- | I am quite obviously a SIP neophyte, so my reading of that may be off. |
| 05:22.43 | Naikrovek | Gambit-: i don't know if endpoint negotiation works that way; you'll have to read the RFC |
| 05:22.50 | Naikrovek | it may very well work that way |
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| 05:23.37 | Naikrovek | but I believe that the device common to both endpoints (in this scenario, an asterisk server) must be able to stay in the stream in some cases |
| 05:23.50 | Naikrovek | to record a call, for example |
| 05:23.57 | Naikrovek | or to barge in on a call |
| 05:24.01 | Gambit- | nods |
| 05:24.16 | Gambit- | That would be an interesting thing to hear the details of. |
| 05:24.23 | Naikrovek | SIP RFC |
| 05:24.24 | Naikrovek | :) |
| 05:24.32 | Naikrovek | super dry read, but it's all in there |
| 05:24.59 | Gambit- | It's on my list, I assure you :) |
| 05:25.05 | Naikrovek | and if you do read it, you'll be more of a SIP expert than anyone in here, I'll bet |
| 05:29.28 | Gambit- | Alas :) |
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| 06:33.00 | ChannelZ | JABOOTY |
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| 06:40.21 | spenguin[work] | ? |
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| 06:48.02 | grharry | quick dumb question, while make menuconfig in asterisk-1.6.2.7 ->codec translators -> codec_resample , it says ( SLIN Resampling codec - Depends on resample ), which it shows as not available [XXX], What do I need to get (libraries,packages...) in order to enable this ? |
| 06:48.56 | ChannelZ | it should sort-of say at the bottom |
| 06:50.12 | grharry | That is what I posted ( It, says on the bottom ....) But I am not sure what packages to download ( Debian ??? ) |
| 06:50.51 | ChannelZ | libresample |
| 06:52.00 | grharry | <PROTECTED> |
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| 06:53.07 | p3nguin | Check the dependencies on the asterisk package. That could provide some useful info for building your own. |
| 06:54.11 | grharry | which is of-course in squeeze ( :-( ) .... not in lenny ... |
| 06:54.55 | p3nguin | You're running testing? |
| 06:55.09 | p3nguin | or unstable? |
| 06:57.57 | p3nguin | yeah, squeeze is testing. |
| 07:00.02 | grharry | I am running lenny |
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| 07:00.37 | grharry | building from source now ! |
| 07:00.37 | p3nguin | And you're saying there is no asterisk package in lenny for comparison? |
| 07:01.18 | p3nguin | Of course it isn't going to be 1.6.2.7, but it should still have almost the same dependencies. |
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| 07:01.54 | ChannelZ | <PROTECTED> |
| 07:01.54 | ChannelZ | <PROTECTED> |
| 07:03.29 | grharry | thanx man !! |
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| 07:15.32 | grharry | It's ok ... and compiling now ..... |
| 07:15.48 | grharry | tnx! |
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| 07:20.10 | MAbbas | hi all |
| 07:20.46 | MAbbas | I am facing some problem using Queue() with asterisk 1.6.2.X .. |
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| 07:22.13 | spenguin[work] | cool, is that all MAbbas |
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| 07:22.34 | MAbbas | I have my agents logged in using AgentLogin() in dialplan. My incoming calls goes to Queue(). However after sometime asterisk hangs out .. |
| 07:23.00 | p3nguin | Are you Answer()ing the calls before sending to Queue()? |
| 07:23.00 | MAbbas | All agent log out and I am not able to give commands on CLI ... |
| 07:23.07 | MAbbas | nop .. |
| 07:23.12 | p3nguin | You should. |
| 07:23.31 | MAbbas | well, same thing worked with 1.4.X |
| 07:23.56 | p3nguin | Alright, then, don't Answer(). It's okay with me. |
| 07:24.11 | MAbbas | :) .. |
| 07:24.28 | MAbbas | let me try that .. |
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| 07:39.00 | carrar | moo |
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| 07:46.05 | ChannelZ | cooks carrar well-done |
| 07:48.01 | zamba | is veresion 1.4.24.1 considered safe? |
| 07:48.03 | zamba | version* |
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| 07:53.20 | kaldemar | zamba: it has a bunch of known vulnerabilities, see http://downloads.asterisk.org/pub/security/ and start from AST-2009-001.html |
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| 08:22.19 | zamba | kaldemar: if i use the asterisk package from ubuntu, will i then have the latest safe release? |
| 08:22.31 | zamba | kaldemar: i know about features maybe not being updated, but for security, is it a good way to go? |
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| 08:26.12 | kaldemar | zamba: you won't have latest release. i don't know about their patch policy though. |
| 08:26.42 | zamba | kaldemar: ubuntu's still at 1.4.x |
| 08:27.04 | kaldemar | what ubuntu? 10.04 has 1.6.2.5. |
| 08:27.24 | zamba | this is 9.04 |
| 08:27.32 | zamba | i'm going to upgrade it to 10.04, though |
| 08:27.35 | zamba | since it's LTS |
| 08:29.20 | kaldemar | i'd compile from source. at least you know what you get then and you'll get better support here. |
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| 08:30.34 | Tim_Toady | ~book |
| 08:30.34 | 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 |
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| 08:39.35 | xheliox | good evening |
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| 09:04.47 | hurdman | Tim_Toady: the 3thrd edition is not out ? |
| 09:05.19 | Tim_Toady | no idea |
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| 10:14.34 | lirakis | ~nat |
| 10:14.34 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 10:14.42 | lirakis | lol "see docs" |
| 10:17.41 | kaldemar | ~sipnat |
| 10:17.41 | 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 |
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| 10:33.07 | *** join/#asterisk davido1 (~davido1@p54B0B3E1.dip0.t-ipconnect.de) |
| 10:40.47 | davido1 | hello =) |
| 10:41.32 | davido1 | I have a quick question, maybe someone knows?... I'm having a weird behaviour with my Asterisk box |
| 10:42.41 | tuxx- | ~ask |
| 10:42.41 | 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. |
| 10:43.54 | davido1 | When a call comes from my Patton SmartNode box (SIP) to Asterisk, and Asterisk Answer()'s (i. e. with Playback(), or Answer()), the caller can hear a messge (played by Playback()) and my phone rings (the destination phone, which is correct), but then if the caller hangs up and I didn't answer the call, my phone continues to ring... |
| 10:43.58 | davido1 | Is that normal? |
| 10:44.31 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-grhuamgsjajconms) |
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| 10:45.28 | tuxx- | pretty much, could you maybe provide a cli output log? http://pastebin.org |
| 10:47.33 | davido1 | yep, gimme a min |
| 10:50.41 | davido1 | tuxx: http://pastebin.org/221179 |
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| 11:26.39 | rhot | hai to all, we have to upgrade asterisk-1.6.0-beta to asterisk-1.6.1, any guide url ? |
| 11:37.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 11:37.49 | Wimme | davido1, were you able to fix it? i have a patton here that does the same thing |
| 11:38.25 | davido1 | Wimme: Yes and no |
| 11:39.07 | Wimme | meaning? |
| 11:39.08 | davido1 | Wimme: If you configure your Patton's sip interface to send early disconnect, you avoid a lot of problems. But sometimes --only sometimes--, the problem stays |
| 11:40.00 | davido1 | Wimme: Are you using a smartnode? What firmware are you running? |
| 11:40.54 | Wimme | im using a SN4554 with fw 5.3 |
| 11:42.12 | davido1 | Wimme: Thanks, you just saved me some time... I'm using a SN4634 with fw 5.5.... Now I know it would be the same with an older version... |
| 11:43.23 | Naikrovek | hurdman: third edition of the book is not out yet |
| 11:43.52 | Wimme | ill try with early disconnect |
| 11:43.55 | Naikrovek | hurdman: leifmadsen is an author of the book, but don't bug him about it. done when it's done kind of thing |
| 11:44.15 | leifmadsen | working on it at this very second actually :) |
| 11:44.23 | Naikrovek | there ya go :) |
| 11:44.33 | leifmadsen | every morning from 7:30am -> 9:00am EST |
| 11:44.56 | carrar | is it done yet? |
| 11:45.02 | carrar | How abou tnow |
| 11:45.03 | leifmadsen | define done? |
| 11:45.06 | carrar | heh |
| 11:45.21 | carrar | The tablet in stone |
| 11:45.26 | leifmadsen | then no. |
| 11:46.18 | carrar | HTML versions are better, then you can change it without telling anyone :) |
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| 11:53.44 | Wimme | davido1, can you predict chen your patton gives you problems with disconnecting? or is it at random? |
| 11:54.03 | Wimme | that early disconnect works for me, only tried a few times tho |
| 11:54.15 | *** part/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 11:54.36 | davido1 | Wimme: It only gives me problems if I "Answer()"ed the call in Asterisk |
| 11:54.55 | davido1 | Answer(), Playback(), Background()... |
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| 11:55.51 | rhot | >hai to all,i have to upgrade asterisk-1.6.0-beta to asterisk-1.6.1, any guide url ? |
| 11:56.06 | Wimme | davido1, thats strange, i start incoming calls from that patton with an answer() all the time |
| 11:57.24 | davido1 | Wimme: Really? What happens if the caller hangs up before the callee answers? Don't you get the same behaviour? |
| 11:59.22 | Wimme | well, that was how i noticed it. i have an isdn coming in through a patton and entering an ivr. when no input is detected it goes tru to support anyway, sometimes calls went to support that were allready hung up by the caller |
| 11:59.49 | davido1 | Exactly |
| 12:00.06 | davido1 | And the problem for me is that sometimes it even crashes my asterisk |
| 12:00.10 | Wimme | now that i enabled rearly disconnect its no longer happening |
| 12:00.19 | Wimme | early* |
| 12:00.57 | davido1 | Wimme: It still happened to me once after that though... And I still think it's bizarre, hahaha, that's why I asked here :P |
| 12:02.48 | MAbbas | hi, I have agent logged in(AgentLogin()) and calls going to Answer() -> Queue(). But after sometimes asterisk stop responding. I am using ver 1.6.X |
| 12:03.14 | MAbbas | the above scenario works fine on 1.4.X |
| 12:04.53 | davido1 | MAbbas: Does that happen after the caller hangs up? => Caller calls, enters queue, hangs up, but your agent keeps ringing? |
| 12:06.14 | tuxx- | anyone have an idea how you can do group call pickups with asterisk 1.4.31? THe pickup() application is for directed call pickup only it seems, and cant find another application that supports group call pickup. |
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| 12:08.25 | [TK]D-Fender | MAbbas: " I am using ver 1.6.X" <- this is God knows what version of any of 3 BRANCHES. |
| 12:08.25 | [TK]D-Fender | MAbbas: Please get SPECIFIC |
| 12:08.33 | [TK]D-Fender | rhot: upgrade.txt <- |
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| 12:11.51 | *** join/#asterisk niekvlessert (~niekvless@92.70.112.34) |
| 12:11.53 | niekvlessert | hello! |
| 12:11.59 | tuxx- | haie. |
| 12:12.02 | niekvlessert | how would i do call pickup groups in asterisk 1.4? |
| 12:12.05 | niekvlessert | hehe |
| 12:12.13 | tuxx- | anyone have an idea how you can do group call pickups with asterisk 1.4.31? THe pickup() application is for directed call pickup only it seems, and cant find another application that supports group call pickup. |
| 12:12.17 | tuxx- | ;-) |
| 12:12.37 | Wimme | tuxx-, use the pickup feature |
| 12:12.45 | rhot | <[TK]D-Fender>thanks |
| 12:13.03 | tuxx- | Wimme: pickup feature is only for directed, not for group calls? |
| 12:13.21 | Wimme | what do you mean with group calls? |
| 12:13.43 | kaldemar | callgroup and pickupgroup options will help you with those. |
| 12:13.47 | Wimme | with directed pickup i understand you use *code* + exten you want to pickup |
| 12:13.48 | tuxx- | couple of sipaccounts with the option 'callgroup=1 and pickupgroup=1'. thats what i call a callgroup |
| 12:13.51 | tuxx- | =] |
| 12:13.58 | tuxx- | yep, group pickup should be something like |
| 12:13.59 | Wimme | well, the pickup feature does just that |
| 12:14.02 | tuxx- | pickup(groupnumber) |
| 12:14.07 | niekvlessert | wimme: the same as tuxx :) |
| 12:14.17 | tuxx- | ok, let me debug some more. brb/bbl |
| 12:14.53 | Wimme | do "features show" on the console |
| 12:15.02 | Wimme | that will give something like : Builtin Feature Default Current |
| 12:15.02 | Wimme | --------------- ------- ------- |
| 12:15.02 | Wimme | Pickup *8 *8 |
| 12:15.03 | kaldemar | see also pickupexten in features.conf. no need to write custom extensions for it. |
| 12:15.18 | MAbbas | asterisk 1.6.2.X |
| 12:15.25 | niekvlessert | just to make sure: |
| 12:15.39 | MAbbas | <[TK]D-Fender> I am trying this on 1.6.2.X |
| 12:16.05 | [TK]D-Fender | MAbbas: PRECISE VERSION |
| 12:16.25 | Wimme | tuxx-, niekvlessert also make sure the phones are in the same pickupgroup in your sip.conf |
| 12:16.35 | MAbbas | <[TK]D-Fender> I am trying this on 1.6.2.6 |
| 12:16.53 | MAbbas | <[TK]D-Fender> I have tried this with 1.6.2.4 as well as 1.6.2.4 |
| 12:16.54 | [TK]D-Fender | MAbbas: upgrade |
| 12:17.18 | niekvlessert | if a bunch of phones have the same callgroup&pickupgroup and one of them is ringing, then what should we do to get that phone without involving the actual extension? |
| 12:17.42 | MAbbas | ok let me do that .. |
| 12:17.56 | Wimme | niekie, on the asterisk console, run "features show" |
| 12:18.33 | Wimme | niekvlessert ii mean |
| 12:19.22 | Wimme | the feature "pickup" should have a code assigned to it. what you do to pick up a call is dial whatever code is assigned |
| 12:19.22 | kaldemar | niekvlessert: dial *8 or something else defined in features.conf. |
| 12:19.39 | MAbbas | <[TK]D-Fender> is Queue() a known issue or its just me? |
| 12:19.55 | [TK]D-Fender | MAbbas: Did you check Mantis? |
| 12:20.13 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-rwgpspbjifbdzqra) |
| 12:20.23 | [TK]D-Fender | MAbbas: And your description is pretty weak |
| 12:20.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 12:20.47 | tuxx- | ah ofcourse *8 in features.conf. Thanks guys! |
| 12:20.48 | Wimme | hello tzafrir_laptop :) |
| 12:21.00 | Wimme | we spoke on the phone last week |
| 12:21.08 | tzafrir_laptop | hi :-) |
| 12:21.20 | Wimme | tnx for helping me out btw :) |
| 12:24.06 | MAbbas | <[TK]D-Fender> the situation is pretty weired. It works on 1.4.29 but when I try same scenario on 1.6.2.4. * just stops responding ... |
| 12:24.24 | MAbbas | <[TK]D-Fender> BTW, I am using winsip to generate calls ... |
| 12:24.57 | [TK]D-Fender | MAbbas: Maybe it has nothignt o dow ith Queue itself |
| 12:24.59 | *** join/#asterisk Jumpie (~zz@ip68-230-28-186.ph.ph.cox.net) |
| 12:27.27 | MAbbas | <[TK]D-Fender> I am unable to pin point the problem. Its simple scenario, Agents logged-in AgentLogin() and calls Answer() -> Queue() |
| 12:30.10 | niekvlessert | thanks it's working!!± |
| 12:30.29 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
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| 12:33.52 | rhot | <[TK]D-Fender> i am downloading asterisk-1.6.1.19 and what version should use like asterisk-addons,zaptel and libpri ? |
| 12:34.38 | [TK]D-Fender | rhot: Latest DAHDI, not Zaptel, latest libpri, and latest addons of that same series |
| 12:35.49 | *** join/#asterisk GNU\colossus (anonymous@truschnigg.info) |
| 12:36.04 | GNU\colossus | hi there |
| 12:36.34 | GNU\colossus | after upgrading from ubuntu 8.04 to 10.04, my asterisk process takes 100% CPU all the time. how can I investigate what's the cause of that behaviour? |
| 12:37.31 | GNU\colossus | (the daemon still works though, far more reliably than before ;)) |
| 12:38.05 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 12:38.18 | DennisG | this is a common bug colossus |
| 12:38.28 | DennisG | you have to modify the /etc/init.d/asterisk file |
| 12:38.36 | DennisG | search for the -c startup option and delete the -c |
| 12:38.47 | DennisG | restart asterisk and your problem is solved |
| 12:39.50 | ManxPower | Try "make config" to make sure you have the latest Asterisk init script. |
| 12:39.51 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 12:40.02 | GNU\colossus | DennisG: according to pgrep, asterisk was invoked by issuing "/usr/sbin/asterisk -F -g -vvvv -U asterisk" |
| 12:40.08 | GNU\colossus | so no -c there anywhere |
| 12:41.06 | rhot | Fender, why i am upgrading asterisk to 1.6.2 is bec managebalance takes 99% ?? any idea ? |
| 12:41.12 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 12:41.12 | DennisG | well.. i have that problem a lot and if i remove the -c then the issue is resolved ;) you can try it if you want hehe |
| 12:44.03 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 12:47.29 | GNU\colossus | any other ideas/theories/suggestions? |
| 12:47.41 | GNU\colossus | lowering verbosity didn't yield improvements |
| 12:49.30 | niekvlessert | i do a make with dahdi fulle |
| 12:49.36 | niekvlessert | dahdi_dummy is not being build |
| 12:49.42 | niekvlessert | what's wrong? |
| 12:53.27 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 12:55.03 | ManxPower | IIRC dahdi_dummy was removed in some version of DAHDI. Did you read the docs included with DAHDI? |
| 12:55.08 | niekvlessert | is it because dahdi no supports internal timing |
| 12:55.38 | niekvlessert | ah ManxPower :) : some version? we use latest dahdi from asterisk.org, it's gone i guess |
| 12:55.41 | ManxPower | niekvlessert, no, it is because it supports it internally or something. Did you read the docs included with DAHDI? |
| 12:59.12 | niekvlessert | yeah, nothing in docs |
| 12:59.17 | niekvlessert | in source though is info |
| 13:00.03 | tuxx- | bleurgh, more ranja |
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| 13:13.09 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:14.14 | *** join/#asterisk puckett_jw (~Adium@mail.scientificconservation.com) |
| 13:16.29 | puckett_jw | I have a question about strange results when using a / in the family and/or keys. I have not been able to find any docs (I am sure that I am missing it) about invalid characters. Can anyone point me in the right direction |
| 13:19.01 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 13:19.26 | puckett_jw | pbx2*CLI> database put "test" "key" "one" |
| 13:19.26 | puckett_jw | Updated database successfully |
| 13:19.27 | puckett_jw | pbx2*CLI> database show test |
| 13:19.27 | puckett_jw | /test/key                     : one           |
| 13:19.27 | puckett_jw | 1 results found. |
| 13:19.27 | puckett_jw | pbx2*CLI> database put "test/key" "" "two" |
| 13:19.27 | puckett_jw | Updated database successfully |
| 13:19.28 | puckett_jw | pbx2*CLI> database show test |
| 13:19.28 | puckett_jw | /test/key                     : one           |
| 13:19.29 | puckett_jw | /test/key/                    : two |
| 13:19.45 | *** join/#asterisk centoslinux (~centoslin@1x-193-157-193-2.uio.no) |
| 13:20.24 | MAbbas | <[TK]D-Fender> I have tried with Asterisk 1.6.2.7 but still the problem persists .. |
| 13:21.06 | MAbbas | <[TK]D-Fender> while the calls are getting to queue() all agents logs out and asterisk stucks at CLI |
| 13:24.02 | *** join/#asterisk Gothicmaster86 (~root@ppp-93-104-169-22.dynamic.mnet-online.de) |
| 13:24.34 | *** join/#asterisk Brooklyn (miracle@the.city.that.never-sleeps.us) |
| 13:24.34 | Gothicmaster86 | hello together |
| 13:26.11 | *** join/#asterisk voxter (~voxter@c-66-176-249-104.hsd1.fl.comcast.net) |
| 13:26.26 | GNU\colossus | after upgrading from ubuntu 8.04 to 10.04, my asterisk process takes 100% CPU all the time. how can I investigate what's the cause of that behaviour? |
| 13:27.22 | *** join/#asterisk coppice (~chatzilla@m121-202-62-41.smartone-vodafone.com) |
| 13:27.52 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 13:29.18 | [TK]D-Fender | GNU\colossus: Perhaps you could tell us something else useful.... like your VERSION |
| 13:30.57 | *** join/#asterisk captiancrash (~jonmoore@12.71.218.225) |
| 13:31.19 | tzafrir_laptop | GNU\colossus, look for the asterisk process in 'ps aux' . Is there any '-c' in the command line? |
| 13:32.08 | GNU\colossus | tzafrir_laptop: nope, asterisk was invoked without that switch |
| 13:32.28 | GNU\colossus | [TK]D-Fender: why not tone yourself down a bit to sound a bit more friendly? |
| 13:33.10 | GNU\colossus | my version is Ubuntu Lucid's default: 1.6.2.5-0ubuntu1 |
| 13:33.23 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:33.33 | [TK]D-Fender | GNU\colossus: Tried upgrading? |
| 13:34.12 | puckett_jw | Is this the wrong place to ask /'s in the database? |
| 13:34.14 | GNU\colossus | [TK]D-Fender: it's the latest version, and I won't deviate from my distro. |
| 13:34.21 | tzafrir_laptop | That seems to b ethe latest in ubuntu |
| 13:34.27 | GNU\colossus | `/usr/sbin/asterisk -F -vv -U asterisk` is how asterisk was invoked |
| 13:34.35 | [TK]D-Fender | Oh well |
| 13:35.03 | Gothicmaster86 | when i phone on an extern isdn-line there was an echo in the line. After i edited the misdn.conf with "echocancel=128" the echo was away. But now i have a grumble in the line. Whats wrong? Asterisk=1.4.29.1, Asterisk-Addons=1.4.10, mISDN-1.1.7.2 |
| 13:35.29 | tzafrir_laptop | GNU\colossus, in top, press 'H' (shift-h) to have threads sorted separately |
| 13:35.47 | tzafrir_laptop | This will give you the PID of the rouge thread. |
| 13:35.57 | tzafrir_laptop | Next, run: strace -p THAT_PID |
| 13:36.16 | tzafrir_laptop | This will hopefully give you an idea of what it actually does |
| 13:36.26 | coppice | Gothicmaster86: you do realise that the m in mISDN stands for misery, don't you? |
| 13:36.31 | MAbbas | <[TK]D-Fender>, I have upgraded to 1.6.2.7 but still my scenario, Agents logged-in using Winsip and call coming Answer() -> Queue(), asterisk stucks after 2-3 mins. Both calls and agents are logged in using winsip .. |
| 13:36.38 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 13:37.36 | Gothicmaster86 | coppice: *lol* thanks but that doesn't help ;) have you an alternative? |
| 13:38.00 | tzafrir_laptop | What device is it? |
| 13:38.21 | GNU\colossus | tzafrir_laptop: the process itself is rather silent; if I enable strace to follow forks, there's a crapload of threads running that all seem to call gettimeofday and read from various fds. |
| 13:38.41 | tzafrir_laptop | strace -f follows forks |
| 13:38.53 | GNU\colossus | tzafrir_laptop: I know, that's what I did ;) |
| 13:39.29 | tzafrir_laptop | Does it get an error from the read immediately? If so: what error? |
| 13:40.03 | Gothicmaster86 | tsafrir_laptop: Do you mean myself? |
| 13:40.47 | GNU\colossus | tzafrir_laptop: all the reads seem to return 8. it doesn't resolve to the symbolic error name though. |
| 13:41.21 | tzafrir_laptop | 8? not -8? This means 8 bytes were read |
| 13:41.34 | *** join/#asterisk MindTheGap (~caio@mail.lpj.com.br) |
| 13:41.53 | GNU\colossus | ah yeah my bad |
| 13:42.01 | GNU\colossus | it's been a while with me programming C ;) |
| 13:42.09 | GNU\colossus | yes, it returns 8. so apparently, the read succeeds |
| 13:42.44 | GNU\colossus | [pid 1432] read(14, "\1\0\0\0\0\0\0\0", 8) = 8 |
| 13:42.50 | *** join/#asterisk Mw3 (mw3@2a01:270:90dc::42) |
| 13:43.13 | GNU\colossus | the threads always seem to read that pattern. |
| 13:43.24 | tzafrir_laptop | What is fd 14? ls -l /proc/PID_OF_ASTERISK/fds/8 |
| 13:43.33 | tzafrir_laptop | grr... /14 |
| 13:43.44 | GNU\colossus | lrwx------ 1 root root 64 2010-05-11 15:43 14 -> anon_inode:[timerfd] |
| 13:43.52 | GNU\colossus | I already investigated that :) |
| 13:44.31 | tzafrir_laptop | Try the following workaround: |
| 13:44.51 | Gothicmaster86 | coppice: have you an idea to alleviate the misery? ;) |
| 13:45.13 | tzafrir_laptop | module unload res_timing_timerfd.so |
| 13:45.37 | coppice | Gothicmaster86: if your card can use dahdi, use it |
| 13:45.40 | tzafrir_laptop | module load res_timing_pthread.so |
| 13:46.43 | GNU\colossus | tzafrir_laptop: my asterisk console apparently crashed after that first statement |
| 13:47.25 | tzafrir_laptop | hmm... use /etc/asterisk/modules.conf to disable res_timing_timerfd.so |
| 13:47.50 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 13:47.56 | GNU\colossus | I'll try to |
| 13:48.46 | Gothicmaster86 | coppice: thats the problem. I have a HFC-S PCI card what doesn't work with dahdi -.- it was my first idea. I've tryed dahdi, capi, bristuff, zaptel. But mISDN was the first driver who works with this cards |
| 13:49.39 | *** part/#asterisk davido1 (~davido1@p54B0B3E1.dip0.t-ipconnect.de) |
| 13:49.44 | GNU\colossus | tzafrir_laptop: commited those changes; situation still unchanged. 99.9% CPU. |
| 13:50.51 | GNU\colossus | tzafrir_laptop: since those problems seem to be related to keeping time, I should probably mention that the machine is running as a qemu-kvm guest. |
| 13:51.48 | coppice | Gothicmaster86: well, if you are stuck with mISDN expect nothing predictably good |
| 13:51.55 | GNU\colossus | with the new module loaded, some futex timeout shows up repeadetly in my strace output |
| 13:52.34 | Gothicmaster86 | coppice: hmm i think i need another hardware when it doesn't work. -.- |
| 13:52.59 | tzafrir_laptop | GNU\colossus, is it indeed kvm? Any chance the system has no kvm support and reverts to use the original qemu? |
| 13:53.18 | GNU\colossus | tzafrir_laptop: nope, KVM support is working fine and dandy |
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| 13:56.31 | ManxPower | <-- not crazy enough to run Asterisk in a VM |
| 14:01.47 | rhot | hai all , i have upgraded 1.6.0-beta to 1.6.1.19 , while registering it shows some warning like this "res_config_mysql.c:613 update_mysql: Attempted to update column 'useragent' in table 'sip_buddies', but column does not exist!" |
| 14:02.31 | Kobaz | you are missing columns in your table |
| 14:03.16 | rhot | we have to add those columns |
| 14:03.18 | rhot | ? |
| 14:03.35 | Kobaz | according to your error message, yes |
| 14:04.00 | Kobaz | many times, just by reading the error, you can see what you need to do |
| 14:04.07 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:05.38 | rhot | in change log , they mention "while upgrading it will automatically add extra column for (ODBC, Postgres, MySQL, SQLite) driver !! |
| 14:06.20 | Kobaz | if your user has persission to do so, and your settings are set up for adaptive |
| 14:09.12 | *** join/#asterisk Wildy (~simba@217.197.243.26) |
| 14:09.16 | Kobaz | you can't just upgrade and expect everything to work |
| 14:10.38 | GNU\colossus | tzafrir_laptop: any other ideas? |
| 14:11.16 | ManxPower | Kobaz, if nothing else they should read the UPGRADE*.txt files. |
| 14:11.45 | *** part/#asterisk puckett_jw (~Adium@mail.scientificconservation.com) |
| 14:11.51 | tzafrir_laptop | GNU\colossus, no |
| 14:12.03 | GNU\colossus | tzafrir_laptop: thanks very, very much for your support though! |
| 14:12.24 | rhot | kobaz: yes i knew it |
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| 14:28.29 | ManxPower | "A man without a woman is like a statue without pigeons." |
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| 14:30.33 | gelo | amen |
| 14:34.01 | rhot | hai all |
| 14:35.45 | niekvlessert | is it possible that it's causing problems if we login on the AMI with the same user multiple times from different IP's? |
| 14:35.47 | *** join/#asterisk VEc (~Vector@84.12.253.146) |
| 14:36.21 | rhot | when i am connecting ami for managing events i am getting 99% cpu usage ! |
| 14:36.26 | VEc | Can anyone give any recommendations for an alternative to vicidial / |
| 14:36.27 | VEc | ? |
| 14:36.58 | ManxPower | then report the problem on issues.digium.com |
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| 14:37.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:43.29 | niekvlessert | what module is responsible from the ami? |
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| 14:44.59 | rhot | let me do that thanks |
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| 14:58.19 | leifmadsen | has anyone gotten Asterisk voicemail to connect via IMAP to Gmail? |
| 14:58.44 | leifmadsen | I can't seem to get the right settings but I've seen someone else able to do it on the mailing lists (but he didn't give his configuration) |
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| 15:16.53 | beardy | I am labbing with asterisk and Cisco 7940:s. I run asterisk on my laptop, the phones are connected to eth0 on a 10.x.x.x subnet. The laptop's wifi interface connects to my main gateway, on a 192.168.x.x subnet. Has anyone else experienced problems with asterisk on a multihomed machine, that doesn't route any packets? I thought it would work anyway.. but apparently, it doesn't. |
| 15:18.47 | beardy | The main thing is that the rtp streams are sent, seemingly, to my gateway, even as both phones sit on the same LAN. |
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| 15:19.30 | beardy | Because I don't get any sound, and with a brief sniff.. that is where they are sending BYEs to. |
| 15:20.12 | flapjacks | Our asterisk server is connected to three 4 different subnets and everything works fine. Maybe your routing tables need some tweaks |
| 15:20.26 | beardy | After they also.. get hung up, after some resend problem, but that's separate.. |
| 15:20.45 | *** join/#asterisk thegoat (~thegoat@c-71-224-154-80.hsd1.pa.comcast.net) |
| 15:21.52 | flapjacks | so both the phones and asterisk are on 10.x.x.x? |
| 15:21.58 | beardy | Yes. |
| 15:22.22 | beardy | I have a dumb l2 toyswitch connected to eth0, and the phones connected to it. |
| 15:22.26 | beardy | L2 |
| 15:23.46 | flapjacks | Have you check the subnet mask on both the phone and asterisk for any fat-finger typos. I have had that prob once or twice |
| 15:23.48 | beardy | There is no reason for asterisk, or the kernel (whoever routes) to send things to my outside IP address.. not even my inside gateway address, but the outside one. It is quite strange to me.. |
| 15:24.01 | beardy | Hmm. I will have an extra look. |
| 15:25.40 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:25.58 | [TK]D-Fender | beardy: No audio direct between 2 phones? |
| 15:26.06 | beardy | (You see, I can make calls, until it gets dropped, I get sound to the phone, but nothing from the phones. The soundstream that should come FROM the phones, gets pointed somewhere else.) Guess I should sniff more.. |
| 15:26.35 | beardy | [TK]D-Fender: Nope. |
| 15:26.57 | [TK]D-Fender | beardy: pastebin a comlpete call with SIP DEBUG enabled, along with your sip.conf masking only passwords |
| 15:27.00 | flapjacks | what does a simple traceroute show to the phones IP address should just be 1 hop |
| 15:27.02 | beardy | [TK]D-Fender: If I call the asterisk, I hear my menus and stuff, but nothing comes from the phone. |
| 15:27.47 | beardy | (until the other problem, the resend error and hangup) |
| 15:27.58 | beardy | [TK]D-Fender: Ok. |
| 15:29.03 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:29.31 | beardy | flapjacks: Yep, should be. Hold on. |
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| 16:00.49 | Cain | Hi, is it possible to view date/time in Asterisk CLI when viewing real-time events ? |
| 16:02.02 | *** part/#asterisk gelo (~gelo@mx01.quobis.com) |
| 16:04.14 | Knightfal | Cain: I dont think so |
| 16:04.27 | Knightfal | You can always tail your log file |
| 16:04.39 | Knightfal | messages,full or whatever you use |
| 16:05.10 | Cain | Knightfal : Ok |
| 16:05.14 | Knightfal | it will have timestamps and display anything you want it to per your logger.conf |
| 16:05.59 | Qwell | Cain: see asterisk.conf |
| 16:07.08 | outtolunc | that or mod logger.c as long as it is not a hammering production box the extra gettimeofday shouldn't drag it down |
| 16:09.50 | Knightfal | Qwell: Can you make the timestamp visible in the cli with a setting in asterisk.conf? |
| 16:09.51 | outtolunc | meant cli.c |
| 16:10.40 | Cain | Got it, thx guys ^^ |
| 16:11.03 | Cain | Knightfal : timestamp = yes ; same as -T at startup in [options] |
| 16:11.12 | beefpastry | anybody here on BroadVox? |
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| 16:11.42 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:11.47 | Knightfal | Cain: I see :) |
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| 16:13.56 | ManxPower | I believe 1.6.0 introduced timestamping CLI events |
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| 16:16.23 | beardy | [TK]D-Fender, flapjacks, well do I feel stupid. externip=my.i.p.adr and localnet==192.168.2.0/255.255.255.0 was what caused that problem. Apparently nat=yes is enough, when running it NAT:ed. Calls get through now. |
| 16:17.09 | beardy | That was where my public ip address came from. It was quite interesting seeing it in the packets when it was not physically connected to it.. oh well, live and learn. |
| 16:17.12 | beardy | Thank you. |
| 16:17.53 | ManxPower | beardy, externip and localnet is for when clients are connecting to a server behind NAT |
| 16:18.31 | beardy | ManxPower: Ah.. |
| 16:18.58 | beardy | ManxPower: I wonder if anyone else has screwed up like I did then. Interesting effects anyway. :) |
| 16:21.07 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 16:21.19 | *** join/#asterisk V4mpire (~Gary@82.118.111.253) |
| 16:21.47 | V4mpire | hey guys its not an asterisk support question but im having trouble finding the information online so thought someone here might know... |
| 16:22.21 | ManxPower | V4mpire, so you are going to violate the social contract in this channel for personal gain? |
| 16:22.31 | ManxPower | Go team! |
| 16:22.35 | V4mpire | i have no idea where else to ask |
| 16:22.51 | [TK]D-Fender | [12:16]<beardy>[TK]D-Fender, flapjacks, well do I feel stupid. externip=my.i.p.adr and localnet==192.168.2.0/255.255.255.0 was what caused that problem. Apparently nat=yes is enough, when running it NAT:ed. Calls get through now. <- no |
| 16:22.59 | [TK]D-Fender | beardy: You need this for the OUTSIDE WORLD. |
| 16:23.04 | ManxPower | V4mpire, to answer your question: penicillin should clear it up in a few days |
| 16:23.11 | [TK]D-Fender | beardy: What you needed was to add the OTHER subnet as well |
| 16:23.30 | V4mpire | ManxPower that has noething to do with anything |
| 16:23.44 | V4mpire | -e |
| 16:24.01 | beardy | [TK]D-Fender: If the other subnet had not been directly connected, and NAT:ed, correct? |
| 16:24.42 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
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| 16:25.25 | V4mpire | and ManxPower once i have this part sorted i can then go onto setting up asterisk again but until i can sort this its not worth setting up asterisk until i can get the phone to connect to the internet |
| 16:26.04 | *** join/#asterisk snayder (~douglas@189.8.255.90) |
| 16:27.03 | beardy | [TK]D-Fender: I suppose I thought, when I did that a couple of days ago, that I needed it to register with my SIP provider, since my asterisk server is NAT:ed, in here. I got it now. Thanks again. Been frustrating me all day. :) |
| 16:32.43 | beardy | Or maybe I haven't gotten it perfectly clear.. externip and localnet are used inside the [phonesomething] sections, not in [general], in sip.conf? |
| 16:32.56 | beardy | (When needed.) |
| 16:34.49 | flapjacks | used in the general section |
| 16:37.08 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 16:37.41 | madduck | my asterisk connects to a trunk provider such that i have an incoming phone number. that works, but dtmf never reaches asterisk. |
| 16:38.13 | madduck | the provider says they use "rfc2833 (inband)" (isn't that a contradiction) and suggest alaw |
| 16:38.16 | madduck | oops |
| 16:38.18 | madduck | 509119 |
| 16:38.21 | madduck | http://support.sipcall.ch/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=96 |
| 16:38.42 | madduck | so i configured alaw for this provider and added dtmfmode=rfc2833 |
| 16:38.47 | madduck | but no luck |
| 16:38.53 | madduck | what else can i do? |
| 16:39.14 | tumaix | anybody got asterisk 1.6.2.7 working on a x86_64 with kernel 2.6.33? |
| 16:42.15 | beardy | flapjacks: One pair for each if there's several different NAT:ed subnets then? Does it get read in pair then, or how are they kept separated? |
| 16:44.12 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.138.125.dsl.dyn.forthnet.gr) |
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| 16:45.00 | flapjacks | beardy so you want each subnet to go out as differen public IP? |
| 16:45.32 | flapjacks | you can have as many localnet=X.x.x.x/mask that you want |
| 16:45.51 | flapjacks | but I think you can only have one externip that will be used |
| 16:47.10 | beardy | flapjacks: I'm just curious on how it works, I don't want/need it now. But yes, let's say I wanted to connect one phone from 4 different NAT:ed subnets around the world. |
| 16:49.34 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 16:50.47 | flapjacks | Basicly externip=x.x.x.x will beused when ever a device talks to your asterisk box that is not in your localnet=x.x.x.x/mask. So as long as you can talk to that "public" ip from your each NAT all should be good |
| 16:51.36 | beardy | Ah yes.. |
| 16:52.18 | flapjacks | If they are just differnt subnets on the same LAN then just use localnet no need for externip if they can all route local to what ever LAN IP address your asterisk box is binded to. |
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| 16:57.49 | V4mpire | will anyone help me out with running my hardware phone through my pc so i can setup asterisk on my server ?, if anyone will dhcp is setup on lan for phone(which works) network/internet sharing is enabled on the wifi but the phone cannot connect to the internet |
| 17:00.46 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:01.39 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 17:04.37 | *** join/#asterisk NoobFukaire (~asd@pdpc/supporter/professional/noobfukaire) |
| 17:04.55 | NoobFukaire | do you guys know if allworx is an asterisk derivative? |
| 17:07.14 | jonny330 | V4mpire: what do you need help with? |
| 17:07.28 | Kobaz | NoobFukaire: looks propriatary |
| 17:07.40 | V4mpire | well the phone doesn't appear to be able to connect to the internet through the pc jonny330 |
| 17:07.44 | NoobFukaire | Kobaz: what leads to that conclusion? |
| 17:08.27 | jonny330 | you are trying to do this on a winblowz pc? |
| 17:09.06 | Kobaz | NoobFukaire: their feature list just doesn't really match with something like asterisk |
| 17:09.13 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 17:10.37 | V4mpire | jonny330 unfortunately yes |
| 17:11.43 | jonny330 | and the firewall is turned off? |
| 17:12.05 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 17:12.07 | V4mpire | no but its not blocking any connections |
| 17:12.08 | Kobaz | NoobFukaire: i've seen a lot of these propriatary ip systems popping up, and they are all custom |
| 17:12.40 | Knightfal | What is the best way to maintain a sip header as a call passes across multiple switches. Cisco Call manager ---> asterisk gateway ---->asterisk acd server |
| 17:12.48 | Kobaz | NoobFukaire: and i know for a fact that paetec does it's stuff inhouse |
| 17:12.52 | Knightfal | Id like to preserver the cisco guid |
| 17:13.30 | jonny330 | V4mpire i would at least try turning off the firewall |
| 17:14.57 | NoobFukaire | Kobaz: thanks |
| 17:15.00 | Kobaz | NoobFukaire: and stuff like this screams legacy: http://na1.salesforce.com/_ui/selfservice/pkb/PublicKnowledgeSolution/d?orgId=00D3000000018Wp&id=50130000000ExLn&retURL=/sol/public/solutionbrowser.jsp%3Fcid%3D02n%26orgId%3D00D3000000018Wp&ps=1 |
| 17:15.04 | V4mpire | the phones ip is set to trusted either way so would be able to use any connection jonny330 |
| 17:15.14 | *** join/#asterisk HerdberH (~Kilian@p4FD6479C.dip.t-dialin.net) |
| 17:15.31 | madduck | can i find out what DTMF tones asterisk receives? there seem to be no SIP messages about it, and verbose mode at 100 also doesn't write anything |
| 17:15.55 | Kobaz | NoobFukaire: something like asterisk wouldn't have that problem |
| 17:16.03 | jonny330 | V4mpire are you seeing the phone try to send any traffic? the problem i am wondering about is the traffic from outside being blocked going into the phone |
| 17:16.35 | V4mpire | jonny330 no i am seeing no traffic going in or out on firewall |
| 17:16.40 | *** part/#asterisk hurdman (~ngeek@ys.antredugeek.fr) |
| 17:17.10 | V4mpire | the last thing blocked by firewall was on the 3rd |
| 17:17.47 | jonny330 | V4mpire did this used to work? or are you trying from scratch? |
| 17:18.15 | V4mpire | trying from scratch because dhcp plays up anytime after phone has been connected to it |
| 17:20.07 | V4mpire | on router that is |
| 17:20.27 | V4mpire | jonny330 will you be here in about 20 mins because i have to pop out a minute |
| 17:20.45 | jonny330 | yeah maybe |
| 17:22.20 | madduck | awesome |
| 17:22.21 | madduck | jugband*CLI> rtp set debug ip 62.65.128.62 |
| 17:22.21 | madduck | Lookup failed for 'ip' |
| 17:23.08 | *** join/#asterisk rare1980_ (~rare1980@115.186.10.7) |
| 17:24.32 | Kobaz | madduck: read the help |
| 17:25.23 | *** part/#asterisk NoobFukaire (~asd@pdpc/supporter/professional/noobfukaire) |
| 17:25.30 | madduck | Kobaz: which part. |
| 17:25.32 | madduck | Kobaz: / |
| 17:25.33 | madduck | ? |
| 17:26.10 | madduck | Kobaz: this? http://slexy.org/view/s21MKx1IGc |
| 17:31.57 | Kobaz | madduck: depends what asterisk version you're using |
| 17:32.11 | Kobaz | core show version |
| 17:32.22 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:32.25 | madduck | Kobaz: 1.6.2.6~rc2-0.8157 |
| 17:32.31 | madduck | (debian) |
| 17:32.46 | Kobaz | so you need to find the kernel for the 1.6.2 rtp set debug |
| 17:32.49 | Kobaz | er |
| 17:32.55 | Kobaz | working on too many things |
| 17:32.59 | Kobaz | you need to find the help |
| 17:33.14 | madduck | http://slexy.org/view/s21MKx1IGc |
| 17:33.19 | madduck | i think i found a bug, that's it. ;) |
| 17:33.35 | Kobaz | i don't know what you are showing me |
| 17:33.41 | Kobaz | is that a documentation site? or a pastebin |
| 17:33.44 | madduck | the online help |
| 17:33.46 | madduck | pastebin |
| 17:34.00 | madduck | you could have opened it and you would have seen. ;) |
| 17:34.01 | Kobaz | okay, so that's from your asterisk console |
| 17:34.06 | madduck | exactly |
| 17:34.13 | Kobaz | i did open it, it's a little snippit of text |
| 17:34.27 | Kobaz | so if your console help doesn't match the command processing, then yeah it's a bug |
| 17:34.42 | madduck | anyway, i got what i wanted even without ip filtering |
| 17:35.27 | madduck | my asterisk connects to a trunk provider such that i have an incoming phone number. that works, but dtmf never reaches asterisk. i tried alaw & ulaw for this provider and dtmfmode=rfc2833 and inband but no luck. what else can i do? |
| 17:35.48 | *** join/#asterisk CleanerX (~nix@HSI-KBW-109-192-057-206.hsi6.kabel-badenwuerttemberg.de) |
| 17:36.17 | [TK]D-Fender | madduck: dtmfmode=info |
| 17:37.31 | madduck | [TK]D-Fender: nothing. i call the number and it eventually arrives at my asterisk, but rtp debug doesn't see anything |
| 17:37.39 | madduck | nor does asterisk seem to hear the DTMF |
| 17:39.15 | madduck | i changed it and did a sip reload |
| 17:39.17 | madduck | is that enough? |
| 17:41.12 | [TK]D-Fender | madduck: I don't see your call or configs... |
| 17:45.14 | madduck | [TK]D-Fender: relevant configs: http://slexy.org/view/s21s4D3AOj (this works fine with all-SIP clients) |
| 17:45.28 | madduck | and with call, do you mean verbose output, and/or SIP dialog? |
| 17:46.27 | [TK]D-Fender | madduck: verbose + sip debug |
| 17:46.31 | madduck | coming |
| 17:47.32 | madduck | [TK]D-Fender: http://slexy.org/view/s2BiOyx4t4 -- this is with dtmfmode=info; if I use rfc2833, then the telephone-event negotiation returns 0x1 |
| 17:48.32 | madduck | here: http://slexy.org/view/s2TwlNfP6s |
| 17:50.45 | [TK]D-Fender | madduck: do a PLAYBACK before just calling WaitExten. You also are only waiting 2 s for input. |
| 17:50.56 | [TK]D-Fender | madduck: awefully short. You need to ensure the link is fully up first |
| 17:51.30 | madduck | uh, are you sure? those 2 seconds are just there so that asterisk doesn't pick up right away, which made it easier for me to test |
| 17:51.37 | madduck | the waiting for input happens within MeetMe, no? |
| 17:52.19 | [TK]D-Fender | madduck: not that your call showed. |
| 17:52.44 | [TK]D-Fender | madduck: RFC2833 should do it though... |
| 17:53.19 | madduck | Executing [join@meetme:2] WaitExten("SIP/41445201426-000004bd", "2") in new stack ? |
| 17:53.32 | madduck | that is completely unnecessary. i could just run MeetMe right there |
| 17:53.39 | madduck | Executing [join@meetme:3] MeetMe("SIP/41445201426-000004bd", ",iMPcs") in new stack |
| 17:53.55 | madduck | it then greets and eventually waits: |
| 17:53.56 | madduck | <SIP/41445201426-000004bd> Playing 'conf-getconfno.gsm' (language 'de') |
| 17:54.09 | madduck | after a few seconds, it just plays it again; three times, then MeetMe hangs up |
| 17:54.53 | [TK]D-Fender | madduck: NOot sure what to say here... |
| 17:57.35 | madduck | does it help if I say that it all does exactly the same if I remove WaitExten(2)? |
| 17:58.40 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:00.08 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 18:00.26 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:00.30 | [TK]D-Fender | madduck: if you've gone back to rfc2833 then you've done all you should have to. Is audio flowing both ways? |
| 18:01.16 | madduck | yeah, audio seems working fine |
| 18:02.15 | [TK]D-Fender | madduck: I'm at a loss on this one. |
| 18:02.21 | madduck | thanks for trying! |
| 18:05.21 | *** part/#asterisk HerdberH (~Kilian@p4FD6479C.dip.t-dialin.net) |
| 18:05.39 | rocksfrow | how do you guys do your faxes? |
| 18:05.48 | rocksfrow | currently i have them going through grandstream handytones |
| 18:05.56 | rocksfrow | and i'm having these random issues, to nobodies suprise |
| 18:06.22 | beek | marks this on the calendar: [TK]D-Fender madduck: I'm at a loss on this one. |
| 18:06.30 | beek | looks toward the east for a star. |
| 18:06.39 | Micc_ | I have a customer using a grandstream handytone 502 using T.38 to fax and everything works fine. I setup another customer with the same settings as far as I can tell and t.38 gets negotiated but then the fax machine hanges up. |
| 18:06.51 | Micc_ | The error on the fax machine is no answer. |
| 18:07.15 | rocksfrow | Micc, yeah. |
| 18:07.18 | rocksfrow | simliar here |
| 18:07.29 | rocksfrow | i put debug on |
| 18:07.38 | rocksfrow | all i see is passing the call, then hangup |
| 18:07.44 | Micc_ | I'm glad I'm not the only person having this problem. |
| 18:07.45 | rocksfrow | which to me looks like the fax machine hanging up |
| 18:07.58 | rocksfrow | Micc, are you usign PRI? |
| 18:08.07 | Micc_ | no. |
| 18:08.13 | Micc_ | the machine has a PRI. |
| 18:08.17 | Micc_ | but its not being used for this call. |
| 18:08.31 | rocksfrow | oh okay |
| 18:08.32 | Micc_ | thi sis just a t.38 sip from customer to our server over internet |
| 18:08.39 | rocksfrow | oh gotcha. |
| 18:09.32 | Micc_ | I had this problem on the other customer, but I don't remember what I did to fix it. maybe changed the nat network settings. |
| 18:10.15 | rocksfrow | really? |
| 18:10.21 | rocksfrow | so you don't think its hte handytones fault? |
| 18:10.51 | bmoraca_work | SCSI controllers + Windows 7 = migraine |
| 18:11.53 | rocksfrow | guys, q931.c:3015 q931_disconnect: call 33330 on channel 1 enters state 11 (Disconnect Request) |
| 18:12.08 | rocksfrow | does that resemble the fax client on sending end, hanging up? |
| 18:12.12 | rocksfrow | or is that the PRI hanging up? |
| 18:12.32 | Micc_ | seems like the fax machine is doing the hangup. |
| 18:12.35 | Micc_ | or the handytone. |
| 18:12.38 | rocksfrow | yeaaa |
| 18:12.42 | rocksfrow | okay maybe a config in the handytone |
| 18:12.46 | Micc_ | But I think it may be a network NAT traversal problem. |
| 18:12.51 | rocksfrow | progress....heh, thanks Micc..glad to see i'm not alone |
| 18:13.03 | rocksfrow | bbs |
| 18:13.23 | Micc_ | but I could be all wrong. thats just the thing that changed I think at the time that made it work. |
| 18:13.37 | Micc_ | So I think I want to put the grandstream outside the firewall and see what happens here. |
| 18:15.52 | Micc_ | internet going down for me. brb |
| 18:21.20 | madduck | [TK]D-Fender: it works if I dial in via SIP via another asterisk (sip:conf-0@madduck.net, pw 84) |
| 18:21.40 | madduck | so i am thinking that the SIP trunk provider must somehow kill DTMF on their side |
| 18:21.52 | madduck | let's see if I can reproduce with another trunk |
| 18:23.28 | *** join/#asterisk rare1980_ (~rare1980@119.158.93.14) |
| 18:24.19 | madduck | hmpf, that other trunk uses '#' for transder |
| 18:24.23 | madduck | transfer. grrr |
| 18:24.44 | *** join/#asterisk Micc_ (~quassel@c-98-225-58-58.hsd1.wa.comcast.net) |
| 18:25.01 | Micc_ | Ok, so I got a different error when adapter is hooked directly to the internet. |
| 18:25.07 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 18:25.21 | Micc_ | it didn't say no answer. |
| 18:27.51 | p3nguin | Great. Now that I have my new router in place and everything seemed to be working right, calls from and to an SPA-3102 in a remote location has no audio. |
| 18:28.04 | madduck | [TK]D-Fender: fyi, it works just fine when I go via another trunk. :( |
| 18:28.23 | [TK]D-Fender | madduck: Could be the provider simply has issues |
| 18:29.59 | madduck | [TK]D-Fender: looks like it, yes. |
| 18:30.01 | madduck | thank you again. |
| 18:33.12 | *** join/#asterisk darkdrgn2k (~darkdrgn2@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 18:33.15 | darkdrgn2k | Hye all |
| 18:33.23 | *** join/#asterisk henk (~henk@netwichtig.de) |
| 18:33.24 | darkdrgn2k | is it possible to test for DID using pure dialplan |
| 18:33.26 | henk | hi |
| 18:33.40 | madduck | waves at henk |
| 18:33.55 | darkdrgn2k | IS if Dialplan=1234567890 set variable x |
| 18:33.59 | darkdrgn2k | sorry |
| 18:34.02 | henk | madduck: *wave* :) |
| 18:34.02 | darkdrgn2k | IS if DID=1234567890 set variable x |
| 18:35.31 | rocksfrow | Micc, you there? |
| 18:35.45 | henk | i have quite some problems connecting my sip clients to asterisk. i know it worked a while ago and i can't remember changing anything. twinkle and linphone say 'request timed out' and i have no idea where to start debugging that problem. any suggestions? |
| 18:36.39 | jonny330 | henk is your network connected? |
| 18:37.00 | jonny330 | explain your network a bit more to help troubleshoot |
| 18:37.15 | madduck | henk: firewall |
| 18:38.41 | henk | jonny330: good that you ask, it's possible the network layout is making it difficult... the server is neither firewalled nor natted. the client has a 192.168.x.x ip and a public ip but both are natted. |
| 18:39.11 | jonny330 | what is the server ip? public or internal? |
| 18:39.12 | henk | madduck: i'm not aware of one on the path :-/ |
| 18:39.19 | henk | jonny330: public |
| 18:39.22 | p3nguin | darkdrgn2k: You can use the IF function or the ExecIf application. |
| 18:39.23 | V4mpire | hi jonny330 did you have an idea to my problem ? |
| 18:39.29 | [TK]D-Fender | darkdrgn2k: Clearly |
| 18:40.11 | madduck | henk: does sip debugging on the asterisk console show packets? can the client log the sip stream? |
| 18:40.28 | darkdrgn2k | p3nguin: how do i check for DID? |
| 18:40.49 | jonny330 | what kind of firewall are the clients behind? |
| 18:40.57 | p3nguin | darkdrgn2k: As alreaday stated, you can use the IF function or the ExecIf application. |
| 18:41.08 | jonny330 | V4mpire i forget what your problem was |
| 18:41.14 | darkdrgn2k | p3nguin: sorry i ment what variable is the DID stored in? |
| 18:41.19 | henk | madduck: server: yes. clients: gimme a sec |
| 18:41.29 | henk | jonny330: mean me? no firewall... |
| 18:41.48 | p3nguin | darkdrgn2k: normally, ${EXTEN} |
| 18:41.55 | jonny330 | henk: router? |
| 18:41.55 | darkdrgn2k | than you:) |
| 18:42.33 | p3nguin | Of course there are cases where the exten dialed from your provider to you is not your DID number. |
| 18:43.12 | p3nguin | In any of those cases, you can often parse it from the SIP header. |
| 18:43.32 | henk | jonny330: yup, nat. |
| 18:44.00 | jonny330 | well how many clients do you have behind the nat? |
| 18:44.21 | madduck | henk: nat=yes on the asterisk for each SIP client? |
| 18:44.55 | p3nguin | For each peer who is behind nat, nat=yes should be added to its peer definition. |
| 18:46.27 | *** join/#asterisk Micc__ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 18:48.12 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:49.39 | henk | madduck: no, the client does not receive anything :-/ |
| 18:50.36 | henk | but the server says: 'Retransmitting #4 (NAT)' so i guess it is trying to use nat correctly. |
| 18:51.29 | henk | jonny330: only one atm... |
| 18:51.37 | madduck | henk: your router probably doesn't know how to do connection tracking for SIP |
| 18:52.01 | jonny330 | ata? |
| 18:52.12 | jonny330 | if that is the case i would do a port forward |
| 18:52.35 | jonny330 | to ensure your sip and media packets get passed to the device |
| 18:53.07 | madduck | or use IAX if you can |
| 18:53.59 | p3nguin | I'm experiencing a similar problem after replacing my router. I have no trouble with most SIP calls, but a single device outside the network now has no audio. |
| 18:54.52 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:55.33 | *** part/#asterisk afink (~afink@204.26.87.226) |
| 18:56.24 | jonny330 | p3nguin where is your asterisk server? |
| 18:56.37 | p3nguin | network1, behind NAT |
| 18:57.20 | jonny330 | and you have ports forwarded? |
| 18:57.37 | p3nguin | Considering all other SIP calls work, I would imagine so. |
| 18:58.02 | jonny330 | okay you don't know for sure? |
| 18:58.09 | p3nguin | Oh, I know for sure. |
| 18:59.05 | jonny330 | k well what kind of router are you using now? what was it before? |
| 18:59.49 | henk | madduck: yup, looks like openwrt doesn't support that :-/ what a shame... |
| 18:59.51 | p3nguin | It is now a Cisco appliance, but was a custom Linux-based firewall. |
| 19:00.22 | henk | madduck, jonny330: thanks for your help :) |
| 19:00.22 | jonny330 | henk you are using openwrt? |
| 19:00.23 | madduck | henk: yeah, one of the reasons why i went the soekris route |
| 19:00.27 | henk | jonny330: yes. |
| 19:00.38 | jonny330 | you are trying to have that be your ata? |
| 19:01.23 | henk | madduck: i'm still waiting for the new soekris model to be released... i should probably stop. what do you run on that? |
| 19:01.39 | henk | jonny330: nope |
| 19:01.43 | jonny330 | p3nguin are you allowing voip traffic from that remote phone ip address on the cisco device? |
| 19:02.03 | madduck | henk: i have a net5501 |
| 19:02.08 | madduck | running lenny |
| 19:02.26 | p3nguin | jonny330: Yes, here's the relevant portion of the router config: http://pastebin.com/e0MhPm3J |
| 19:02.35 | madduck | it's my router at home and it works perfectly fine, can even do traffic shaping and openvpn |
| 19:02.56 | madduck | (+ dhcp, dns, radvd, ipv6 tunnel, …) |
| 19:03.17 | p3nguin | jonny330: All other SIP calls work, except now this one remote device has no audio. |
| 19:03.19 | henk | madduck: ah ok. do you use any interfaces or frontends to things or do you do everything manually? (move this to #debian.ch?) |
| 19:03.49 | madduck | henk: everything manually. and if you want to move it, ask concrete questions there (but i am soon going offline) |
| 19:04.32 | p3nguin | I'm going to have to try a softphone on that remote network soon. |
| 19:04.54 | jonny330 | yeah that cisco config looks good to me |
| 19:04.57 | henk | madduck: in that case i don't :) |
| 19:05.44 | p3nguin | jonny330: If all SIP calls had no audio, I would blame it on my router. But this isn't the case. |
| 19:06.35 | jonny330 | well i was thinking you might only allow sip and media from your voip service providor |
| 19:06.41 | Micc__ | how do I escape commas in my dialplan variables? |
| 19:07.34 | p3nguin | jonny330: And since I cannot put the old router back in place to see if audio is restored, then back to the Cisco to make sure audio is gone again, troubleshooting will need to be done from a different angle. |
| 19:07.53 | leifmadsen | Micc__: don't put commas in your dialplan variables... |
| 19:07.54 | beardy | In the sample features.conf there is a function, "automixmon => *3", and a comment next to it says, "Make sure to set the X and/or x option in the Dial() or Queue() app call", I don't see any documented x/X option, only w/W, is there an x/X option for Dial()? |
| 19:08.00 | jonny330 | yeah i would try using a soft phone like you said |
| 19:08.09 | leifmadsen | Micc__: like a backslash to comment them out -- depends on the version of asterisk -- might have to do \\\, |
| 19:08.12 | madduck | beardy: yes. |
| 19:08.15 | Micc__ | they are in callerid variables |
| 19:08.18 | Micc__ | I can't help that. |
| 19:08.23 | beardy | madduck: Is it documented, did I miss it? |
| 19:08.37 | jonny330 | beardy they are talking about wW for call recording |
| 19:09.11 | beardy | jonny330: Yes. But the mixing one says x/X, not w/W.. |
| 19:09.12 | Kobaz | much yays |
| 19:09.57 | madduck | beardy: http://lmgtfy.com/?q=asterisk+dial+x |
| 19:10.07 | jonny330 | oyou are right |
| 19:10.09 | jonny330 | <?php echo _("x: Allow the called user to start mixed recording after pressing *3 (Asterisk v1.6)")?><br> |
| 19:11.55 | *** join/#asterisk V4mpire (~Gary@82.118.111.253) |
| 19:13.10 | darkdrgn2k | ok i got a linksys PAP , but i cant seem to get in call functions workign on the analog line. IE ## for blind transfer. Any ideas? |
| 19:13.26 | beardy | madduck: I checked that, already. Did you? |
| 19:13.51 | [TK]D-Fender | darkdrgn2k: Can you navigate your Voicemail with it? |
| 19:13.53 | madduck | beardy: no, but i remember reading about it on voip-info.org and that usually comes up first |
| 19:14.00 | Micc__ | I'm using asterisk 1.6.2.6, callerid sometimes contains a comma for some reason, I need to filter that out. Can I just remove it? |
| 19:14.20 | beardy | madduck: I don't see an X or x option. Only w or W. |
| 19:14.20 | Micc__ | When I pass a var with a comma in it to func_odbc, it screws things up and crashes asterisk. |
| 19:14.39 | darkdrgn2k | [TK]D-Fender: i can login to the mailbox, yes |
| 19:14.51 | beardy | I'm trying to find out if it's a typo in the features.conf sample file, or something magical. |
| 19:14.55 | darkdrgn2k | ie: *97, 3003, password... |
| 19:15.14 | [TK]D-Fender | darkdrgn2k: And enter your password, pick up messages, etc? |
| 19:16.08 | darkdrgn2k | yes |
| 19:16.09 | jonny330 | beardy i saw the x/X option |
| 19:16.10 | [TK]D-Fender | darkdrgn2k: Then you have a config problem. Your device is fine. Head back to #freepbx <------------------ |
| 19:16.20 | madduck | beardy: I pass X, so you might want to do that too. |
| 19:16.24 | jonny330 | http://www.freepbx.org/v2/ticket/4137 |
| 19:16.28 | darkdrgn2k | [TK]D-Fender: nop, it works find on other devices.. |
| 19:16.43 | beardy | madduck: Just voip-info.org that's incorrect again. |
| 19:16.47 | [TK]D-Fender | darkdrgn2k: DTMF either works, or it doesn't. |
| 19:16.57 | darkdrgn2k | [TK]D-Fender: on the VOIP phone it works fine... |
| 19:17.06 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.08 | darkdrgn2k | [TK]D-Fender: but on the PAP it does not?! |
| 19:17.46 | beardy | madduck: 'asterisk -rx "core show application Dial" | less' there it was apparently, but not on that wiki. |
| 19:19.30 | beardy | jonny330: On the wiki? Good, I didn't. Search term on the page please? |
| 19:22.54 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 19:28.18 | Micc__ | the customer I had t.38 working for last night is now doing that hangup thing. |
| 19:29.00 | *** join/#asterisk orioni (~chatzilla@95.107.226.183) |
| 19:29.32 | orioni | hi, i have a registered trunk on asterisk , but when i make a call to that trunk i get all-lines-are-busy |
| 19:29.32 | *** join/#asterisk MmixX (mixed@61.14.191.133) |
| 19:29.49 | orioni | trying with a softphone it works ok |
| 19:29.55 | *** join/#asterisk megalomano (~klonstein@38.124.169.126) |
| 19:30.14 | orioni | the softphone registered at the other asterisk works ok |
| 19:30.40 | orioni | the softphone when is registered at my asterisk cant make outbound calls |
| 19:31.11 | darkdrgn2k | ok any one know if its possible to pull a current XML settings file from a PAP2? |
| 19:31.51 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 19:32.39 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 19:33.07 | flapjacks | does your sip provider require 10 digit dialing? |
| 19:33.27 | jonny330 | yes |
| 19:33.31 | orioni | tried with and without |
| 19:33.48 | orioni | this is the error SIP/sipTringTrunkName-093c98d0 is circuit-busy |
| 19:34.58 | flapjacks | you sure all your ports are forwarded correctly and your nat settings are correct |
| 19:37.30 | ManxPower | orioni, Trixbox/FreePBX is not supported here. |
| 19:38.29 | ManxPower | orioni, just put Noop(HANGUPCAUSE is ${HANGUPCAUSE}) after your Dial line to see what the real hangup cause is. If that does not help then you'll need to do a SIP debug and pastebin.ca the results |
| 19:38.54 | orioni | manxpower : it says cicuite busy error 21 |
| 19:39.35 | ManxPower | Paste that line and just that line to the channel |
| 19:41.30 | orioni | manxpower : give me a sec |
| 19:41.31 | *** join/#asterisk Alagar (~Administr@122.164.35.243) |
| 19:41.51 | paulc | <vent> When I go to the kitchen room thing, I don't expect the vending machines to be turned off "because a recording session is in progress". Yes, I know the "studio" is next door to the kitchen, but isn't that padded shed you bought from vocalbooth.com meant to screen out all the external noise? Thus letting me fetch a snack or a chilled drink during the scant few minutes I have between ridiculous meetings today? Arrgghhh FML! </vent> |
| 19:42.23 | paulc | goes to Starbucks to cool down before the next pointless meeting... |
| 19:42.35 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 19:43.01 | orioni | manxpower : this is the otuput from asterisk dialplan show http://pastebin.com/wW24tYuZ |
| 19:43.16 | ManxPower | orioni, just put Noop(HANGUPCAUSE is ${HANGUPCAUSE}) after your Dial line to see what the real hangup cause is. If that does not help then you'll need to do a SIP debug and pastebin.ca the results |
| 19:43.34 | ManxPower | ~freepbx |
| 19:43.35 | 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 |
| 19:43.45 | ManxPower | If you don't know how to do that then you should ask on the SUPPORT CHANNEL FOR YOUR INSTALL |
| 19:44.47 | orioni | im gonna modify that in a sec |
| 19:45.40 | *** join/#asterisk edwin_quijada (~macaruchi@200.26.172.50) |
| 19:45.42 | [TK]D-Fender | ManxPower: *'s cause codes = worthless. SIP debug is the only thing of value |
| 19:46.18 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 19:46.24 | edwin_quijada | IS there a way to record the call for a queue getting the agent and extension from that queue?? |
| 19:46.56 | niekvlessert | can i stop asterisk, delete the astdb, start asterisk, will astdb be recreated? |
| 19:47.00 | darkdrgn2k | .. Ok any onw know how to make a PAP2 work with in-call featuers like ##? |
| 19:47.22 | [TK]D-Fender | niekvlessert: Yes |
| 19:47.41 | [TK]D-Fender | darkdrgn2k: It does |
| 19:48.14 | niekvlessert | ok great, will this take more then a few hours? because of sip registers? and will be any data missing? |
| 19:48.47 | *** join/#asterisk maour (~gnu@unaffiliated/maour) |
| 19:49.45 | orioni | manxpower: NoOp("SIP/5001-089a9fd0", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack |
| 19:49.49 | *** join/#asterisk rdircio (~admin@189.137.224.205) |
| 19:49.58 | [TK]D-Fender | niekvlessert: No, it will take about a second. And obviously you'll lose whatever was in there before |
| 19:50.02 | orioni | NoOp("SIP/5001-089a9fd0", "HANGUPCAUSE is 21") |
| 19:50.06 | [TK]D-Fender | orioni: SIP DEBUG <-------------- |
| 19:50.26 | [TK]D-Fender | orioni: Anything else is worthless trash |
| 19:50.31 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:50.43 | edwin_quijada | if I trasfer a call to a queue Can U recording this call using the extension and agent that take the call ? |
| 19:50.59 | darkdrgn2k | [TK]D-Fender: what does? |
| 19:51.05 | [TK]D-Fender | darkdrgn2k: WORK |
| 19:51.25 | [TK]D-Fender | DarkGo into its options. if it doesn't say that it's using that for anything specific then it isn't touching it |
| 19:51.42 | darkdrgn2k | [TK]D-Fender: i dont doubt it because i have a SPA working with hit. but this PAP2 refuses to work... |
| 19:51.42 | orioni | SIP/2.0 407 Proxy Authentication Required |
| 19:52.01 | p3nguin | jonny330: I think I may have figured something out... |
| 19:52.15 | jonny330 | whats that? |
| 19:52.16 | [TK]D-Fender | orioni: PASTEBIN |
| 19:52.18 | orioni | <[TK]D-Fender> : turning on sip debug is says SIP/2.0 407 Proxy Authentication Required |
| 19:52.21 | darkdrgn2k | [TK]D-Fender: it wont even flash :-S |
| 19:52.23 | orioni | just a sec |
| 19:53.10 | p3nguin | jonny330: The ACLs on the Cisco router do not permit RFC 1918 addresses to come inbound on e1. I ran rtp debug, and all the packets from the remote device (which is behind NAT) have the private address from that network on the packets. |
| 19:53.11 | *** join/#asterisk iscario (~div@30.244.71-86.rev.gaoland.net) |
| 19:53.35 | jonny330 | ahh that could cause some problems |
| 19:54.50 | *** join/#asterisk TimeRider (~steve@5ac7b32d.bb.sky.com) |
| 19:55.26 | p3nguin | I think this means I need to deconfigure nat for the peer and configure nat ON the peer device. |
| 19:55.44 | orioni | <[TK]D-Fender> : here the sip debug http://pastebin.com/Ynffp4hm |
| 19:56.05 | p3nguin | Unless someone else has an idea. |
| 19:56.29 | ManxPower | orioni, we need the ENTIRE debug, not just the normal and as expected proxy auth required message |
| 19:57.04 | p3nguin | I guess I could also test my theory by removing that line in the ACL that prevents 172.16/12 from being allowed to come in on the public internet interface. |
| 19:57.06 | V4mpire | anyone else have any ideas why my phone cannot access the internet via my pc and ways around it i am using windows xp |
| 19:57.27 | [TK]D-Fender | orioni: COMPLETE call with verbose 10 as well |
| 19:57.32 | p3nguin | Anyone else ever run into a nat issue like this? |
| 19:57.51 | ManxPower | p3nguin, all nat=yes does is say "ignore the SIP headers and use the packet headers for the source address. |
| 19:58.18 | p3nguin | manxpower: So it is very likely that it still causes rtp to be broken, right? |
| 19:58.18 | ManxPower | maybe not ALL, but that's the basics |
| 19:58.40 | p3nguin | My RTP packets have the remote device's private IP on them. |
| 19:58.48 | ManxPower | Well if your NAT router is somehow sending out packets with a private address you have some really serious problems. |
| 19:59.22 | ManxPower | p3nguin, sounds like the incoming call is not matching the peer you think it is matching. You have allowguest=no in sip.conf [general], right? |
| 19:59.25 | *** join/#asterisk maour (~gnu@unaffiliated/maour) |
| 19:59.38 | p3nguin | Oh it is. It's an audio issue, not a SIP issue. |
| 19:59.55 | ManxPower | um, the source/dest IPs of the RTP is defined by the SIP INVITE |
| 19:59.56 | p3nguin | The call completes end-to-end, no matter which direction the call is made. |
| 20:00.08 | p3nguin | There's simply no audio at all. |
| 20:00.26 | ManxPower | Do you have allowguest=no in sip.conf [general]? |
| 20:00.57 | p3nguin | No, because I accept anonymous SIP calls. |
| 20:00.58 | orioni | here http://pastebin.com/NKbuREux |
| 20:01.43 | ManxPower | p3nguin, I suspect your call is considered "anonymous" and so is not nat'd. Do you have nat=yes in [general] |
| 20:02.01 | p3nguin | No, it's not anonymous. |
| 20:02.20 | ManxPower | orioni, you are not authorized to use your carrier. |
| 20:02.25 | p3nguin | Yes, I have nat=yes in general because asterisk, too, is behind nat. |
| 20:02.31 | ManxPower | p3nguin, I wish you the BEST of luck. |
| 20:02.36 | p3nguin | I'm sure you do. |
| 20:02.37 | [TK]D-Fender | orioni: Contact: <sip:211@192.168.8.21> <--- you also did not set your server up [properly to work from behind NAT |
| 20:02.52 | ManxPower | nat=yes has nothing to do with Asterisk being behind NAT. It is ONLY for REMOTE CLIENTS being behind NAT. |
| 20:03.03 | [TK]D-Fender | ManxPower: Umm.. it does... |
| 20:03.10 | ManxPower | [TK]D-Fender, if you say so. |
| 20:03.17 | orioni | ManxPower : i can use that provider with a softphone on my pc |
| 20:03.19 | ManxPower | I've already wished him the best of luck, so you can help him. |
| 20:03.29 | ManxPower | orioni, Asterisk is NOT your softphone. |
| 20:03.33 | p3nguin | If this were a new deployment or something, I might think you're going down the right path, but it's not new. All I did was swap out a router and lost audio on this remote device. All other SIP calls are fine. |
| 20:03.35 | [TK]D-Fender | ManxPower: in as much as its relation to localnet's filter for externip/host |
| 20:04.02 | ManxPower | p3nguin, sounds like you have SIP ALG enabled on the router. |
| 20:04.02 | orioni | ManxPower : i mean that i can use that provider , no block from them |
| 20:04.10 | orioni | and im looking to find the solution for asterisk too |
| 20:04.18 | ManxPower | orioni, and I mean your provider is rejecting your call. |
| 20:04.19 | p3nguin | Unfortunately I don't have ALG. that would be a quick fix. |
| 20:04.26 | ManxPower | p3nguin, what router? |
| 20:04.32 | p3nguin | Cisco 831 |
| 20:04.38 | orioni | where might be the problem ? |
| 20:05.04 | p3nguin | Here's the relevant part of the config: http://pastebin.com/e0MhPm3J |
| 20:06.08 | p3nguin | This part shows the port forwarding, which is good for all other calls. |
| 20:06.25 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 20:07.13 | p3nguin | <--- SIP read from 24.171.73.146:5060 ---> |
| 20:07.18 | p3nguin | SIP/2.0 200 OK |
| 20:07.18 | p3nguin | To: <sip:203@172.16.255.21:5060>;tag=9e1ac2dd307b88fdi0 |
| 20:07.18 | p3nguin | From: "asterisk" <sip:asterisk@192.168.192.242>;tag=as268d37a2 |
| 20:07.20 | *** join/#asterisk maour (~gnu@unaffiliated/maour) |
| 20:07.58 | p3nguin | I believe this is why it isn't working with this router. The header contains the private address on the remote device, and RTP is using the private address, which I believe is blocking the stream. |
| 20:08.07 | [TK]D-Fender | [16:04]<orioni>and im looking to find the solution for asterisk too |
| 20:08.09 | p3nguin | I'm going to make a couple changes and see if my theory is right. |
| 20:08.14 | [TK]D-Fender | orioni: Contact: <sip:211@192.168.8.21> <--- you also did not set your server up [properly to work from behind NAT |
| 20:08.55 | ManxPower | p3nguin, try no ip nat service sip udp 5060 |
| 20:08.57 | ManxPower | p3nguin, WHICH header? |
| 20:08.58 | p3nguin | You want me to stop natting/forwarding 5060? |
| 20:09.18 | ManxPower | p3nguin, no I want you to stop SIP ALG on udp/5060 |
| 20:09.41 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 20:09.45 | ManxPower | is that the UDP packet header or the SIP header that has "the private IP"? |
| 20:09.55 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 20:10.00 | p3nguin | SIP header pasted above |
| 20:10.12 | p3nguin | Here's the invite: Sending to 24.171.73.146 : 5060 (NAT) |
| 20:10.12 | p3nguin | Using INVITE request as basis request - d9baf483-65781277@172.16.255.21 |
| 20:10.21 | ManxPower | p3nguin, that is TOTALLY expected and normal for a remote client behind nat |
| 20:10.36 | ManxPower | the INVITE obviously isn't. |
| 20:11.22 | *** join/#asterisk jarrod (anon@unixc.org) |
| 20:11.31 | jarrod | how do I get polycom to register using the full user@domain |
| 20:11.36 | ManxPower | Oh! I imagine that could easily be correct. "basis" doesn't mean much as far as I know. |
| 20:12.02 | ManxPower | the ACTUAL invite should be useful to look at. |
| 20:12.36 | ManxPower | p3nguin, http://puck.nether.net/pipermail/cisco-voip/2009-June/002282.html |
| 20:12.57 | ManxPower | IIRC SIP fixup is enabled by default |
| 20:14.02 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:14.33 | orioni | fixed the nat issue , now the packet are sent with the public IP of asterisk |
| 20:14.34 | p3nguin | sh run does not show ip nat service anything. So that tells me that no fixups are enabled. |
| 20:14.39 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 20:14.52 | p3nguin | access-list 101 deny ip 172.16.0.0 0.15.255.255 any |
| 20:14.59 | orioni | but still no luck with outhoging calls |
| 20:15.00 | p3nguin | I believe this is the culprit. |
| 20:15.13 | p3nguin | This ACL is on the wan interface inbound. |
| 20:15.49 | [TK]D-Fender | orioni: Where is the new debug? |
| 20:16.06 | ManxPower | p3nguin, just fscking try it. If it defaults to off then it won't show in the config |
| 20:16.10 | orioni | wait a sec |
| 20:16.13 | ManxPower | and it will not hurt to try it. |
| 20:17.06 | ManxPower | p3nguin, It magically starts working when you remove that ACL? |
| 20:17.42 | p3nguin | I haven't tested that theory just yet. I just got access to try your idea. I'll report back in a moment. |
| 20:17.55 | *** join/#asterisk smooth_penguin (~smoove@59.95.6.242) |
| 20:18.04 | darkdrgn2k | k i got the flash to work but not the in-call tran sfers :-SD |
| 20:18.43 | p3nguin | Okay, added no ip nat service sip udp port 5060 and still no audio. |
| 20:18.58 | ManxPower | does it show up in your sh config |
| 20:18.58 | p3nguin | Now to yank that line from the access-list. |
| 20:19.14 | ManxPower | you may have to do a clear ip nat * I don't know |
| 20:19.21 | p3nguin | Yes, it shows that it turned it off. |
| 20:19.57 | p3nguin | "clear ip nat translation *" coming right up. |
| 20:20.07 | ManxPower | so it must have been enabled by default |
| 20:20.18 | *** join/#asterisk Greek-Boy (~Greek-B0y@41.188.154.137) |
| 20:20.56 | p3nguin | Cleared the trans table and still no audio. |
| 20:21.23 | ManxPower | at least you have eliminated the most common issue with Cisco routers |
| 20:22.04 | jarrod | if I want polycom to register "1000@domain.com" do I just put that in the authId ? |
| 20:23.09 | p3nguin | Fuck. I think I locked myself out by trying to erase that entry in the acl. |
| 20:23.53 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 20:24.15 | [TK]D-Fender | jarrod: Put that in the server ip |
| 20:24.42 | jarrod | put the full user@domain in the server ip? |
| 20:24.53 | [TK]D-Fender | jarrod: No, jsut teh domain. |
| 20:25.07 | jarrod | i have specified the ip address of the multi-domain sip server there |
| 20:25.39 | jarrod | that way i dont need SRV records |
| 20:26.08 | p3nguin | Okay, I have reapplied the ACL without access-list 101 deny ip any any log |
| 20:27.30 | orioni | here the new debug http://pastebin.com/kUsZ5v2i |
| 20:27.53 | p3nguin | That didn't help, either. |
| 20:28.22 | p3nguin | err, wrong paste... |
| 20:28.37 | p3nguin | Okay, I have reapplied the ACL without access-list 101 deny ip 172.16.0.0 0.15.255.255 any |
| 20:29.27 | p3nguin | So now I think I need to configure nat setting on the device instead of on Asterisk. |
| 20:29.33 | ManxPower | p3nguin, It is unlikely the packets are coming in with a source address of a private IP. The SIP header might, but not the UDP header (which is what the Cisco sees) |
| 20:30.56 | p3nguin | I'm starting to feel like that guy last week. I could call him via SIP URI and things were fine, but his devices which registered to his box as peers had no or only one-way audio. |
| 20:31.45 | [TK]D-Fender | orioni: Reliably Transmitting (NAT) to 80.91.120.50:5060: <--- THEY are not behind NAT themselves. |
| 20:31.55 | [TK]D-Fender | orioni: get it right... |
| 20:31.59 | [TK]D-Fender | checkout time.. BBL |
| 20:32.06 | *** part/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 20:32.40 | orioni | 80.91.120.50 this is not with nat |
| 20:32.50 | orioni | it has a public ip directly assigned |
| 20:33.25 | orioni | the .21 has a private ip of 192.168.8.21 and a public of 80.91.125.21 |
| 20:36.52 | orioni | what do i need to change ? |
| 20:37.16 | orioni | alreayd specified external ip and private network on sip_nat.conf |
| 20:40.00 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 20:40.28 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 20:45.21 | *** part/#asterisk orioni (~chatzilla@95.107.226.183) |
| 20:45.27 | *** join/#asterisk orioni (~chatzilla@95.107.226.183) |
| 20:48.41 | orioni | any idea ? |
| 20:54.23 | *** join/#asterisk ido- (~wtf@94.159.237.190) |
| 20:55.02 | ido- | hello. is it possible to make 3g video calls from my gsm cellphone to a sip client/asterisk ? |
| 20:56.48 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 21:01.32 | p3nguin | Problem with private IP address in RTP: confirmed. Loaded zoiper on a computer on the same LAN with this remote phone... rtp debug shows the PUBLIC address in the RTP stream when using zoiper, and audio is two-way! |
| 21:01.57 | p3nguin | The SPA-3102 gives a private IP address and no audio when using the same peer configuration on Asterisk. |
| 21:02.44 | p3nguin | So... the next question is: How the heck do I solve this? |
| 21:02.47 | *** join/#asterisk Baylink (~jra@cerberus.vicimarketing.com) |
| 21:04.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:05.32 | p3nguin | Ah, I spoke slightly too soon... and I hate to repeat, but I want [tk]d-fender to see this. |
| 21:05.39 | p3nguin | Problem with private IP address in RTP: confirmed. Loaded zoiper on a computer on the same LAN with this remote phone... rtp debug shows the PUBLIC address in the RTP stream when using zoiper, and audio is two-way! |
| 21:05.45 | p3nguin | The SPA-3102 gives a private IP address and no audio when using the same peer configuration on Asterisk. |
| 21:06.27 | niekvlessert | if i have a lot of ds_devicestate couldnt get state in astdb in the log, can this cause instability of asterisk? |
| 21:07.07 | iscario | do you know if some IAX proxy exist ? |
| 21:07.15 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 21:07.47 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 21:08.21 | *** join/#asterisk bsaxon_ (~bsaxon@12.68.234.174) |
| 21:09.04 | *** join/#asterisk halkun (~chatzilla@mke-66-97-112-72.milwpc.com) |
| 21:10.35 | halkun | Good news. I have asterisk running! I have learned how to start the CLI and start and stop the daemon. |
| 21:10.56 | halkun | FOr my next trick I'm trying to log x-lite on locally |
| 21:11.09 | [TK]D-Fender | iscario: It does not. It is not a routed protocol |
| 21:11.32 | halkun | Now I set up users in sip.conf, correct? |
| 21:12.30 | iscario | oh yeah, sometimes i become stupid .... thx anyway [TK]D-Fender |
| 21:12.32 | [TK]D-Fender | orioni: Reliably Transmitting (NAT) to 80.91.120.50:5060: <----- Again, THEY are not behind NAT. |
| 21:12.52 | halkun | the thing is, I don't think x-lite is seeing the asterisk server |
| 21:13.16 | halkun | I'm getting a 404 - not found error |
| 21:13.49 | [TK]D-Fender | halkun: depends what its in response to |
| 21:13.53 | orioni | [TK]D-Fender : my asterisk has an private ip of 192.168.8.21 and a public static ip of 80.91.125.21 |
| 21:14.02 | Baylink | [TK]D-Fender: "IAX is not a routed protocol"? |
| 21:14.04 | [TK]D-Fender | halkun: And it clearly CAN see *... its getting wa refusal <- |
| 21:14.10 | orioni | i dont understand when you say they are not behind nat |
| 21:14.22 | [TK]D-Fender | Baylink: SIP can be proxy'd, routed, etc. |
| 21:14.31 | [TK]D-Fender | Baylink: IAX2 = P2P only |
| 21:15.29 | halkun | Ok, in that case I'll set a user up in sip.conf and see if it sees that :) I'm on the right track then. I did't want to go further I wasn't even connecting to the server |
| 21:15.34 | Baylink | Ah. For future reference, "IP is a 'routable protocol'", so so is anything on top of it. "IAX must directly connect 2 end points" isn't *quite* the same thing. :-) |
| 21:15.37 | orioni | [TK]D-Fender : can you elaborate more |
| 21:16.52 | [TK]D-Fender | Baylink: UDP encapsulates it. Routing can occur on multiple layers. |
| 21:17.10 | Baylink | Oh, shit. IAX is over UDP, isn't it. |
| 21:17.18 | Baylink | Nevermind. Last day before vacation. :-) |
| 21:17.28 | [TK]D-Fender | orioni: The service you are calling is NOT BEHIND NAT. They have public IP's. You have configured your peer wrong |
| 21:18.33 | orioni | [TK]D-Fender : so putting nat=no will resolve this part? |
| 21:18.41 | [TK]D-Fender | orioni: yes |
| 21:19.30 | orioni | indeed now it says transmitting no nat to ... |
| 21:19.39 | orioni | but still busy circuit |
| 21:19.47 | *** join/#asterisk rdircio (~admin@189.137.224.205) |
| 21:20.32 | [TK]D-Fender | orioni: Then something else is still wrong. |
| 21:20.46 | orioni | Transmitting (no NAT) to 80.91.120.50:5060 the correct line |
| 21:20.55 | [TK]D-Fender | orioni: pastebin your peer entry masking only the secret |
| 21:21.51 | orioni | http://pastebin.com/fsyhDfeN |
| 21:22.12 | orioni | secret changed ;) |
| 21:22.42 | Qwell | is it 'tok' now? |
| 21:22.52 | Qwell | or no, wait. 1234? |
| 21:23.33 | orioni | nope its secret |
| 21:24.53 | p3nguin | Does zoiper do something fancy with NAT that an SPA-3102 doesn't? |
| 21:25.03 | [TK]D-Fender | orioni: username=355880211 |
| 21:25.11 | [TK]D-Fender | orioni: From: "35588800211" <sip:35588800211@80.91.125.21>;tag=as4da06c26 |
| 21:25.21 | [TK]D-Fender | orioni: these 2 #'s are different. Why? |
| 21:25.27 | [TK]D-Fender | orioni: there is an extra 0 in one |
| 21:25.29 | orioni | i have to change that on the pastebin |
| 21:25.40 | [TK]D-Fender | orioni: do NOT fuck with the numbers! |
| 21:25.41 | orioni | on the system they are both the same, sure |
| 21:25.51 | [TK]D-Fender | orioni: We won't know what's wrong if you screw with the evidence |
| 21:26.09 | orioni | i understand man , but i had to change that |
| 21:26.29 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 21:26.36 | orioni | its my real number , dont wana get marketing calls etc |
| 21:33.25 | orioni | any idea ? |
| 21:33.52 | [TK]D-Fender | onI'm not wasting my time when the pieces don't add up. |
| 21:36.27 | *** part/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 21:42.30 | megalomano | hi people , I'm trying to automate my dialplan , with calls that be doing with numbers from a list , I want to make it with AGI ( bash ) , someone have any documentation to make this ?... only I need the variables that I can manipulate |
| 21:42.45 | *** join/#asterisk disposable (disposable@blackhole.sk) |
| 21:43.57 | megalomano | and some examples :s |
| 21:46.01 | halkun | Hey, if I get a recording that "the person you are calling is unavailable, please try again".. Is that coming from x-lite or asterisk? |
| 21:47.27 | [TK]D-Fender | megalomano: Your use of "automate" is rather vague. Where are calls coming in from to start with? What is "automate" vs what I suppose could only be called "manual" now? |
| 21:47.31 | halkun | I'm wondering if I successfully was able to get asterisk to "talk" to me. |
| 21:47.45 | [TK]D-Fender | halkun: What does * CLI at verbose 10 have to say? |
| 21:49.03 | halkun | asterisk -vvvvvvvvvv says it's running |
| 21:49.17 | halkun | or do I need to log into CLI |
| 21:50.08 | halkun | Oh. It logged it :) |
| 21:51.12 | halkun | Cool! so I can log x-lite into * new have an account set up... |
| 21:51.17 | halkun | baby steps |
| 21:51.49 | halkun | baby steps..... now to get the server to talk to the 3102 |
| 21:52.20 | megalomano | [TK]D-Fender:sorry , the word "automate" is not right , only , I want to make a calls , with numbers that are on a list |
| 21:53.25 | [TK]D-Fender | megalomano: AGI doesn't start calls. It is an application called ... BY a call. |
| 21:53.41 | [TK]D-Fender | megalomano: as a way to choose how to interact with it instead of just plain dialplan apps. |
| 21:55.02 | p3nguin | http://pastebin.com/1w3T3PKt |
| 21:56.36 | megalomano | [TK]D-Fender: understand , I don't want to start a call , simply , redirects the incoming call to other destination " EXEC TRANSFER " |
| 21:58.31 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:59.03 | [TK]D-Fender | megalomano: There is also no "transfer" per-se. You are just jumping to some other point in dialplan, etc. |
| 21:59.25 | [TK]D-Fender | megalomano: Also, what aspects of the decision process validate the use of AGI for processing? |
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| 22:00.39 | megalomano | [TK]D-Fender: before to jump the AGI , I have Answer() |
| 22:00.50 | [TK]D-Fender | megalomano: .... ok.... and? |
| 22:02.07 | megalomano | [TK]D-Fender: Answer() -> AGI(someagi.agi,${EXTEN} - |
| 22:02.40 | megalomano | [TK]D-Fender: only, I have these 2 options to make a test |
| 22:04.45 | *** join/#asterisk Chilling_Silence (~Chilling_@203-167-138-160.dsl.telstraclear.net) |
| 22:05.24 | p3nguin | This is the collected RTP evidence: http://pastebin.com/1w3T3PKt |
| 22:05.54 | Chilling_Silence | I'm getting X-Asterisk-HangupCauseCode: 16, and it's dropping my calls randomly. I've got the sip debug trace from a call which just "dropped". Happens for ISDN and SIP calls too. Is anybody able to assist me with some debugging of my issue? I've done dozens of PBX's and never had this issue before :( |
| 22:09.32 | megalomano | [TK]D-Fender: in my dialplan I have this: exten => _X.,1,Answer() ---> exten => _X.,2,AGI(scrip.agi,${EXTEN}) ---> exten => _X.,109,Hangup |
| 22:11.47 | megalomano | [TK]D-Fender: the scrip says at the end line , echo " EXEC TRANSFER SIP/$x@X.X.X.X" , where $x variable i made it randomly |
| 22:11.57 | Chilling_Silence | Is there anybody who's able to assist me with these supposedly random call drops? please? I'm just having a bit of trouble understanding the SIP debug output :-/ |
| 22:12.54 | megalomano | [TK]D-Fender: is correct this procedure ? |
| 22:15.47 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 22:16.06 | [TK]D-Fender | megalomano: What is the point of this "transfer"? |
| 22:18.29 | *** part/#asterisk jarrod (anon@unixc.org) |
| 22:20.34 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 22:21.34 | megalomano | [TK]D-Fender: the asterisk . will receive, around 15 calls , from friend user ( into asterisk ) , the incoming calls have the way 768XXXXXXX , for this the calls will be placed in other location into the dialplan |
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| 22:25.17 | megalomano | [TK]D-Fender: is possible to make simultaneous calls to the same dialplan? |
| 22:25.20 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 22:25.49 | p3nguin | huh? |
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| 22:28.10 | [TK]D-Fender | megalomano: That isn't a "trasfer", that is a "goto" |
| 22:28.57 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 22:29.23 | megalomano | [TK]D-Fender: right , but the command that use to make the call is " EXEC TRANSFER " |
| 22:30.03 | [TK]D-Fender | megalomano: "make the call"? To what? |
| 22:32.20 | [TK]D-Fender | megalomano: I also see no AGI command "transfer" |
| 22:33.36 | megalomano | [TK]D-Fender: ok , the scrip works with this line " EXEC TRANSFER SIP/$X@GSM " |
| 22:34.21 | [TK]D-Fender | megalomano: Again... that THROWS the call outside of * completely. Where is it going? |
| 22:35.12 | boodu | hi |
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| 22:42.45 | megalomano | [TK]D-Fender: the incoming calls from some client ( type friend ) , will be redirected to some number stablished in the dialplan, i.e. -- X dial 3456 -----> asterisk ( context [redirection] exten => 3456,1,Answer; exten => 3456,2,AGI(scrip.agi); the scrip generate some number and execute this "EXEC TRANSFER SIP/$VARIABLE@xxxx") ------> the call terminate with this way ' client ---> $VARIABLE@XXXX ' |
| 22:46.55 | [TK]D-Fender | megalomano: Trasfer is like a DIAL execpt if the call is SIP it literally throws the call out of * via a REDIRECT. Why are you completely ejecting the call from *? |
| 22:47.05 | [TK]D-Fender | megalomano: Instead of DIAL()-ing somewhere? |
| 22:50.46 | megalomano | [TK]D-Fender: the traffic will come from a softswitch , without human interaction |
| 22:52.11 | megalomano | [TK]D-Fender: some like asterisk :) |
| 22:52.13 | *** join/#asterisk jhirley_ (~jhirley@adsl-6-37-114.mia.bellsouth.net) |
| 22:54.08 | [TK]D-Fender | megalomano: Ok, if all you're doing is a scripted decision and then throwing it out to another system... what's missing? |
| 22:57.18 | megalomano | [TK]D-Fender: more details about the AGI in bash , my scrip is very simple , and I don't have all details about the dialplan manipulation over AGI |
| 22:59.25 | megalomano | [TK]D-Fender: for example the variable ${DIALSTATUS} , is managed over AGI ? |
| 23:02.02 | [TK]D-Fender | megalomano: You aren't even CALLING dial from your description. So where is DIALSTATUS supposed to come from or even mean? |
| 23:02.42 | *** join/#asterisk lepine (~lmacguire@204.92.8.29) |
| 23:03.08 | *** join/#asterisk jksM (~jks@193.189.93.254) |
| 23:03.15 | [TK]D-Fender | ... |
| 23:03.15 | lepine | What are the prerequisites to route a call to an extension defined in users.conf when using WaitExten() ? |
| 23:03.22 | lepine | I'm told the extension is invalid |
| 23:03.40 | [TK]D-Fender | lepine: You require a pattern that actually matches in your dialplan |
| 23:04.16 | lepine | thus XXXX if the extensions are 4 digits ... |
| 23:05.16 | [TK]D-Fender | lepine: Something like that... the details obviously don't quite make it yet. |
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| 23:16.45 | p3nguin | RTP from a remote device has a private IP, resulting in no audio. How do I solve it? http://pastebin.com/1w3T3PKt |
| 23:19.13 | [TK]D-Fender | p3nguin: not seeing full call SIP DEBUG... or the other IP's involved... |
| 23:20.11 | p3nguin | I can't provide it because I am not at the location where that phone is and cannot make a call from it. |
| 23:21.30 | [TK]D-Fender | p3nguin: You know the drill... come back when you have EVIDENCE to look at :p |
| 23:22.33 | p3nguin | Isn't there some theory to discuss regarding the matter? We know the problem, so there's probably some fix for it. |
| 23:23.02 | [TK]D-Fender | p3nguin: We can waste hours talking about things that may not matter at all to it. I'd rather not. |
| 23:23.23 | p3nguin | You'll be here anyway, so we might as well discuss the possible options. |
| 23:24.49 | *** part/#asterisk jasonjjohnsonjr (~jjohnson@adsl-93-38-91.owb.bellsouth.net) |
| 23:25.09 | [TK]D-Fender | p3nguin: Looks likely to be a NAT/subnet issue. |
| 23:26.19 | p3nguin | On the remote network, the SPA-3102 is giving its private IP address in the RTP packets. On that same LAN, Zoiper on a computer (on the same subnet) gives is offering the public IP address in the RTP packets. |
| 23:27.46 | [TK]D-Fender | p3nguin: Specify nat=yes for the peer, and that the subnet is still local |
| 23:28.27 | p3nguin | I have nat=yes, and I'll check my localnets setting now. |
| 23:29.33 | p3nguin | Okay, I added localnet=172.16.0.0/12 to sip.conf. |
| 23:30.24 | p3nguin | Now as soon as I can find someone over there to make a call or receive one, we'll see if it makes any difference. I'll also be able to go ahead and grab a full sip debug just in case. |
| 23:30.53 | p3nguin | Would nat=route be anything useful in this particular scenario? |
| 23:31.41 | [TK]D-Fender | p3nguin: No. You KNOW where that device is supposed to be. This isn't a quessing game |
| 23:31.49 | [TK]D-Fender | guessing* |
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