| 00:00.27 | [TK]D-Fender | rob_rf: YouR side may be all IP... but someone's hardware is hitting the PSTN at some point. |
| 00:13.31 | volga629 | I am looking for command that I can pull realtime ldap mailbox info |
| 00:20.34 | *** join/#asterisk psilvao (~psilvao@190.22.111.218) |
| 00:34.10 | *** join/#asterisk psilvao (~psilvao@190.22.111.218) |
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| 00:38.42 | volga629 | ok found voicemail show users for realtime-ldap |
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| 01:26.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:50.39 | volga629 | srtp still really unstable :-( |
| 01:51.06 | volga629 | keep crashing asterisk can't make working properly |
| 01:51.29 | WIMPy | Do you have a current libsrtp? |
| 01:52.27 | volga629 | srtp-1.4.4-1 |
| 01:53.20 | WIMPy | When I fiddled around with it, released versions were buggy. |
| 01:54.41 | volga629 | hmm |
| 01:54.52 | volga629 | Do I need srtp-static.x86_64 ? |
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| 01:55.51 | volga629 | yes, I have collections of backtrace files |
| 01:55.57 | WIMPy | It stopped crashing when I used the CVS. |
| 01:56.20 | volga629 | Is possible get source ? |
| 01:56.25 | WIMPy | I've used that since. |
| 01:57.11 | volga629 | can you tell which version ? |
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| 01:57.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:59.29 | WIMPy | Hmm. Seems rather meaningless. It says 1.3.0.2, but that's from yesterday. |
| 02:02.21 | volga629 | right need look where find the package if exist at all |
| 02:02.25 | volga629 | now |
| 02:03.13 | WIMPy | :pserver:anonymous@srtp.cvs.sourceforge.net:/cvsroot/srtp |
| 02:03.32 | WIMPy | That's the one I'm using. |
| 02:03.57 | WIMPy | Or have been using. I disabled it a few weeks ago. |
| 02:04.44 | WIMPy | G.722, 10ms frames, SRTP and conferences seemed to be a litle too much for my phone. |
| 02:05.15 | WIMPy | wants switch based conferences back. |
| 02:05.22 | WIMPy | And a whole lot of other stuff. |
| 02:06.02 | volga629 | the link is ftp ? |
| 02:06.22 | WIMPy | No, cvs. |
| 02:07.02 | volga629 | ok |
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| 02:08.36 | WIMPy | http://sourceforge.net/scm/?type=cvs&group_id=38894 |
| 02:13.14 | volga629 | still give 1.4.4 |
| 02:13.36 | WIMPy | Where? |
| 02:14.14 | volga629 | when fetch the tar |
| 02:14.21 | WIMPy | Not the tar. |
| 02:14.35 | WIMPy | That's from 2007 and bad. |
| 02:14.58 | WIMPy | Follow the instructions on the page to fetch via CVS. |
| 02:15.45 | volga629 | ok |
| 02:21.14 | volga629 | CVS password: ? |
| 02:21.20 | jpsharp | press enter |
| 02:25.34 | volga629 | cvs -z3 -d:pserver:anonymous@srtp.cvs.sourceforge.net:/cvsroot/srtp co -P srtp |
| 02:25.54 | volga629 | I used this command and it fetched 1.4.4 |
| 02:27.00 | WIMPy | Are you referring to the content of the VERSION file? |
| 02:27.36 | volga629 | yes |
| 02:27.48 | WIMPy | Looks like noone bothered to put anything else there. |
| 02:28.12 | volga629 | so it right one ? |
| 02:28.18 | WIMPy | yes |
| 02:28.37 | volga629 | ok let me try install |
| 02:29.12 | WIMPy | Make sure to uninstall any other version forst. |
| 02:29.15 | WIMPy | first. |
| 02:29.50 | WIMPy | Then after installing, re-make Asterisk and install. |
| 02:30.48 | volga629 | ok |
| 02:31.10 | volga629 | is special CFLAGS need for x86_64 ? |
| 02:31.53 | WIMPy | Unless you set them to something special, gcc should output something suitable for the system it's running on. |
| 02:33.30 | volga629 | I see let me try |
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| 02:35.18 | volga629 | I have rpm for srtp module, remove this one ? |
| 02:36.09 | WIMPy | I guess, you should build Asterik yourself. |
| 02:36.54 | WIMPy | Packages always seem rather unpredictable to me, but I'm sure someone will dispute that. |
| 02:37.51 | volga629 | ok I will remove |
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| 02:38.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:42.55 | volga629 | ok srtp done re make asterisk |
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| 02:58.45 | volga629 | Look ok I placed call to *98 and crashed :-( |
| 02:59.50 | WIMPy | I guess you found a new bug then. |
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| 03:01.57 | volga629 | look like :-( |
| 03:02.08 | *** join/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 03:02.31 | slicknick5181 | I am having issues getting chan_mobile to work |
| 03:02.42 | slicknick5181 | I am using Ubuntu 12.04 with asterisk 1.8 |
| 03:03.05 | slicknick5181 | was not compiled with the same compile-time options as this version of Asterisk. |
| 03:03.22 | slicknick5181 | loader.c:777 inspect_module: Module 'chan_mobile.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 03:03.59 | WIMPy | Build it for your running Asterisk version. |
| 03:04.35 | slicknick5181 | I ahve seen a lot of people saying they recompiled it but I don't know how to do that |
| 03:05.01 | WIMPy | How did you get it? |
| 03:05.50 | slicknick5181 | when I downloaded Asterisk I did not have chan_mobile.so.. So I went and downloaded an rpm and used that 1 |
| 03:06.06 | slicknick5181 | I figured thats my problem but idk how to fix it |
| 03:06.25 | WIMPy | When you use packages, you must make sure they fit together. |
| 03:06.44 | WIMPy | If you build the stuff yourself, they will fit. |
| 03:07.06 | slicknick5181 | I kinda figured it was my screw up.. |
| 03:07.09 | slicknick5181 | can u give me some steps |
| 03:07.54 | slicknick5181 | I've used terminal to install things before with the "build" and "configure" commands but I mostly copyed and pasted |
| 03:08.44 | WIMPy | more like configure;mak;make install. |
| 03:09.11 | WIMPy | But (almost) all sources contain a INSTALL file with instructions. |
| 03:09.37 | slicknick5181 | that sounds correct.. sould I use the rpm that I downloaded? |
| 03:09.57 | WIMPy | You need the source to build yourself. |
| 03:10.13 | WIMPy | That may be available as rpm as well. |
| 03:12.43 | slicknick5181 | ok I will try this |
| 03:23.33 | volga629 | for lack of RTP activity in 31 seconds I did again with version 1.4.2 and right now I get |
| 03:24.16 | *** part/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 03:24.55 | volga629 | and after right the way 8703 Segmentation fault :-) |
| 03:32.25 | volga629 | WIMPy thank you help and suggestions, srtp didn't worked, but I will continue look on it |
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| 04:21.53 | volga629 | recompiling asterisk rpms again try make SRTP happy |
| 04:52.54 | volga629 | signal 7, Bus error is that mean might hardware problem ? |
| 04:53.14 | volga629 | that from last backtrace |
| 04:54.45 | jpsharp | No, bad code can create a SIGBUS |
| 04:55.01 | jpsharp | Much more rare than a SIGSEGV, but still possible. |
| 04:55.42 | volga629 | hmm |
| 04:56.55 | volga629 | yes srtp it just don't want to work, I already downgraded libsrtp and rebuilt rpms, but asterisk continue to crash :-) |
| 04:56.58 | volga629 | :-( |
| 04:57.40 | jpsharp | I've had SRTP running, but I built everything from straight source. |
| 04:58.00 | jpsharp | I no like RPMs. They make me cry. |
| 04:58.49 | volga629 | in ast_unregister_file_version () ??? |
| 04:59.09 | volga629 | yes, exactly cry |
| 05:00.09 | volga629 | might you know what it is the function |
| 05:00.43 | volga629 | almost all module get this No symbol table info available. |
| 05:14.17 | jpsharp | Not off the top of my head. |
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| 05:19.18 | volga629 | I am going disable SRTP until I will find way to implement it |
| 05:19.21 | volga629 | :-( |
| 05:27.28 | jpsharp | I suggest building everything from straight source. No RPMs, no SRPMS, just good oldfashioned tar.gz |
| 05:29.11 | volga629 | yes that look like the way |
| 05:29.59 | volga629 | I know it should work |
| 05:32.54 | volga629 | is which version is better to use 10 or 1.8 ? |
| 05:36.26 | jpsharp | I use 10 myself, but 1.8 is a long term release so it should be just as good. |
| 05:36.34 | jpsharp | It wont be as bleeding edge as 10 is, though. |
| 05:41.00 | volga629 | hmm make sense |
| 05:41.50 | volga629 | will be the same trouble with srtp on 10 ? |
| 05:55.21 | volga629 | Do you know about imap storage which patches need to use ? |
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| 06:14.54 | jpsharp | No, I've not used IMAP storage. |
| 06:20.12 | volga629 | I assume that in ver 10 all previous patches already in |
| 06:37.29 | volga629 | I tried and it is patched, I started build right now see how is go's |
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| 07:02.04 | volga629 | app_voicemail.c:90:21: error: linkage.h: No such file or directory |
| 07:02.24 | volga629 | I checked dev package installed |
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| 07:17.36 | slicknick5181 | issues with chan_mobile.... I have the same issue as the rest of the world my phone connects and then disconnects immediately... I keep reading that I need to recompile.. what is that and how do I do it |
| 07:17.49 | slicknick5181 | I am running ubuntu 12.04 asterisk 1.8 |
| 07:26.35 | *** join/#asterisk slicknick5181 (~patti@204.195.131.94) |
| 07:26.55 | slicknick5181 | Using Ubuntu 12.04 asterisk 1.8 |
| 07:27.11 | slicknick5181 | I need to recompile asterisk due to bluetooth issues |
| 07:27.19 | slicknick5181 | Can anyone help me do this? |
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| 07:37.53 | kaldemar | slicknick5181: just recompiling is unlikely to help you. how did you install asterisk in the first place? |
| 07:38.25 | slicknick5181 | sudp apt-get install asterisk asterisk_mobile |
| 07:39.20 | slicknick5181 | kaldemar: there is an issue with bluetooth and apparently if you change a few lines of source and recompile all is supposed to be well |
| 07:39.21 | kaldemar | you'd need to get the source code, do some modification that fixes your issue and then compile the source code to runnable binaries. |
| 07:39.49 | slicknick5181 | Correct, Thats what I have been reading but thats where I'm lost |
| 07:40.02 | kaldemar | on which part? all of those? |
| 07:40.07 | slicknick5181 | Where do I get that source code and then after I edit it what do I do |
| 07:40.56 | slicknick5181 | I have googled for the source code.. I think I know what I'm looking for.. and I hav not found it |
| 07:41.36 | kaldemar | you better get the source package for asterisk_mobile from ubuntu if that's what you use. |
| 07:44.57 | slicknick5181 | ok I think I found that |
| 07:48.36 | kaldemar | #ubuntu would really be a more suitable place to ask on directions for compiling the source into a package when you've modified the source. |
| 07:50.13 | kaldemar | which source are you supposed to modify? |
| 07:54.46 | slicknick5181 | asterisk-mobile |
| 07:55.33 | kaldemar | which file in it? |
| 07:57.56 | slicknick5181 | go to forums.digium.com/viewtopic.php?f=1&t=78042 |
| 07:58.05 | slicknick5181 | read the answer that says I win |
| 07:58.20 | slicknick5181 | from ellafino |
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| 08:00.07 | kaldemar | "sudo apt-get build-dep asterisk-mobile && sudo apt-get source asterisk-mobile" will get you started. |
| 08:00.49 | slicknick5181 | ok :) |
| 08:05.34 | slicknick5181 | terminal is unpacking a bunch of stuff |
| 08:08.00 | slicknick5181 | kaldemar: now it's finished |
| 08:09.27 | kaldemar | the first part of the command installed build-time dependencies for asterisk-mobile, and the second part got you the source code. next step is to modify chan_mobile.c |
| 08:10.46 | slicknick5181 | so this is where I find the tar.gz and find chan_mobile.c correct |
| 08:10.50 | slicknick5181 | ? |
| 08:11.58 | kaldemar | nope. this is where you cd to asterisk-addons-<version>, the "apt-get source" command got you the tar's but it also extracted them. |
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| 08:13.17 | slicknick5181 | ahh ok I found that file |
| 08:13.30 | slicknick5181 | Thats where I edit the source |
| 08:15.37 | slicknick5181 | kaldemar: ok I have edited the file chan_mobile.c and saved it |
| 08:17.03 | slicknick5181 | kaldemar: I have asterisk currently installed on this system.. do I need to uninstall it before going any further |
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| 08:17.18 | kaldemar | no, you don't need to uninstall. |
| 08:17.41 | slicknick5181 | ok |
| 08:17.42 | kaldemar | then "sudo ./configure && sudo make" will compile it for you |
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| 08:21.34 | slicknick5181 | kaldemar: thats running through it's processes now |
| 08:22.00 | slicknick5181 | What will be my next step |
| 08:24.18 | kaldemar | slicknick5181: when the compile is done, you can cut corners in an ugly way and replace your current /usr/lib/asterisk/modules/chan_mobile.so with the compiled channels/chan_mobile.so or do it the proper way and build a package with "sudo apt-get install checkinstall && sudo checkinstall". that will make you a .deb file that can be installed with "sudo dpkg -i X.deb" where you replace X.deb with the filename that you have. |
| 08:26.04 | slicknick5181 | kaldemar: ahh ok It's making sense to me now |
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| 12:26.56 | JamKo | Is there a type of SIP attack, that behaves like a smurf attack? I have been noticing attackers directing the SIP attacks at my broadcast IP. When this happens, all of my |
| 12:27.41 | JamKo | routers and switches go crazy. I can see the attack from any machine with wireshark, eventhough the attack is not going directly to that machine. |
| 12:29.10 | JamKo | My asterisk box shows the attack at the CLI. Asterisk sits on a public IP, yet my entire lan, which is on a different internet connection than asterisk, is affected. But all isp gateways are connected on the lan side, which allows |
| 12:29.13 | JamKo | the crossover. |
| 12:29.52 | JamKo | But still, why are all of my machines seeing this attack, and why is the attacker directing it at the broadcast IP? |
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| 15:37.08 | uskerine | hi |
| 15:37.26 | uskerine | i am trying to setup a digium tdm401 card following "asterisk: the future of the telephony" |
| 15:37.39 | uskerine | ztcfg and zttool are no longer available |
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| 15:37.52 | uskerine | and while installing "zaptel" package it says it is deprecated |
| 15:37.55 | uskerine | i am using ubuntu |
| 15:38.05 | uskerine | any tip on replacements for these commands? |
| 15:38.32 | WIMPy | It's been called dahdi fao some years. |
| 15:39.38 | uskerine | dahdi is installed |
| 15:39.43 | uskerine | how can i check hardware status? |
| 15:39.58 | uskerine | "/sbin/ztcfg -vv and zttool |
| 15:40.04 | uskerine | which are the equivalent ones in dahdi? |
| 15:42.00 | WIMPy | replace zt with dahdi_ |
| 15:43.21 | uskerine | ok thanks |
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| 15:44.44 | uskerine | which is the deault signalling for FXO, "loopstart protocol"? |
| 15:48.30 | uskerine | dahdi_tool shows TDM410P as OK |
| 15:48.41 | uskerine | but dahdi_cfg -vv shows nothing |
| 15:48.53 | uskerine | should it show channel map? I have already configured _/etc/zaptel.conf |
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| 15:51.57 | WIMPy | There's no zaptel any more. |
| 15:52.12 | WIMPy | /etc/dahdi/system.conf |
| 15:52.56 | WIMPy | Try dahdi_genconf |
| 15:57.55 | uskerine | ok thanks |
| 15:58.31 | uskerine | is there any good tutorial on how to set up fxo and fxs ports somewhere? |
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| 16:01.10 | WIMPy | The book, pereffably in the current version. |
| 16:01.14 | WIMPy | ~book |
| 16:01.14 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 16:12.31 | uskerine | thanks, i was reading the old one |
| 16:12.55 | uskerine | is it free? |
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| 16:30.42 | uskerine | i have modified chan_dahdi.conf and system.conf |
| 16:30.52 | uskerine | how can i make asterisk "reload" the configuration changes? |
| 16:31.02 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 16:31.06 | uskerine | i have modified chan_dahdi.conf and system.conf |
| 16:31.07 | uskerine | how can i make asterisk "reload" the configuration changes? |
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| 16:40.21 | uskerine | optiplexgamma*CLI> dahdi restart |
| 16:40.21 | uskerine | No such command 'dahdi restart' (type 'core show help dahdi restart' for other possible commands) |
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| 16:49.13 | volga629 | <PROTECTED> |
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| 17:14.31 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 17:14.31 | *** mode/#asterisk [+o ChanServ] by kornbluth.freenode.net |
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| 17:31.34 | uskerine | i have configured TDM410P from digum |
| 17:31.49 | uskerine | it picks up the phone and it is expected to echo caller voice, but it doesn't |
| 17:31.52 | uskerine | any idea on why? |
| 17:31.58 | uskerine | -- Executing [s@from-pstn:1] Answer("DAHDI/1-1", "") in new stack |
| 17:31.59 | uskerine | <PROTECTED> |
| 17:32.06 | uskerine | i can see it actually run the "echo command" |
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| 17:52.44 | jacobw | uskerine: reload |
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| 18:19.12 | uskerine | what should i reload? |
| 18:19.16 | uskerine | jacobw |
| 18:19.34 | uskerine | how is that related to echo? |
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| 18:39.02 | uskerine | i have configured asterisk with a digium TDM410P (analog 4xPOTS) |
| 18:39.04 | uskerine | FXOs |
| 18:39.21 | uskerine | asterisk answers but I can not hear myself (even when I see "echo" is executed) |
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| 19:04.21 | uskerine | TDM410P does not detect when caller finish the call and it is onhook |
| 19:04.24 | uskerine | any tip? |
| 19:08.39 | *** join/#asterisk i-ahmed (~i-ahmed@6-14.ft.nsrv.net) |
| 19:10.08 | WIMPy | That's perfectely normal unless you have disconnect supervision all the way. |
| 19:10.26 | volga629 | Multiple audio streams are not supported |
| 19:10.48 | volga629 | for SRPT is anybody know what work around ? |
| 19:11.06 | WIMPy | Don;t make it optional on the phone. |
| 19:11.27 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:11.40 | volga629 | If I set mandatory it crash SRTP if off nothing not happened |
| 19:12.00 | volga629 | We are requesting SRTP, but they responded without it! |
| 19:12.18 | WIMPy | Newer Asterisk versions don't like the optional offering. |
| 19:12.43 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 19:13.13 | WIMPy | But if it still crashes on you, the only option is to disable it or find the bug. |
| 19:15.40 | volga629 | yes, I build asterisk from tar source and it 10.5 |
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| 19:27.29 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:27.36 | volga629 | I checked back trace and look like the same crash which I started report from march 2012 |
| 19:27.40 | volga629 | https://issues.asterisk.org/jira/browse/ASTERISK-19508?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel#issue-tabs |
| 19:29.18 | WIMPy | I've been using it with the 360. |
| 19:29.47 | WIMPy | It crashed for me as well until I changed to the CVS version of libsrtp. |
| 19:30.07 | WIMPy | But interestinlgy only if the phone had come over a certain uptime. |
| 19:30.30 | WIMPy | With a freshly booted phone it was ok with the release version. |
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| 19:34.16 | volga629 | I changed firmware on snom to make clean |
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| 19:39.43 | *** join/#asterisk Node_403 (b4ea60f9@gateway/web/freenode/ip.180.234.96.249) |
| 19:40.09 | Node_403 | hi all |
| 19:40.32 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:41.38 | Node_403 | My GSM-VoIP gateway is behind firewall. I need to optimize my Bandwidth for Gateway use. 300kbps available. Need to use it for 32 channels |
| 19:42.37 | WIMPy | Wow. Definitely no go with RTP. |
| 19:42.40 | Node_403 | My GSM-VoIP gateway is behind firewall. I need to optimize my Bandwidth for Gateway use. 300kbps available. Need to use it for 32 channels |
| 19:43.14 | Gugge | 32 channels on 300kbps ?!? |
| 19:43.25 | Node_403 | hmm |
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| 19:44.10 | drmessano | lol |
| 19:44.21 | volga629 | I don't know match about asterisk development,but the module should be more stable |
| 19:44.54 | Node_403 | any idea? |
| 19:45.02 | Gugge | Node_403: get more bandwidth :) |
| 19:45.04 | WIMPy | Are you sure you don't have any old stuff getting in the way. |
| 19:45.06 | WIMPy | ? |
| 19:45.35 | WIMPy | Node_403: With IAX and a really low bandwidth codec, it might be possible. |
| 19:45.38 | Node_403 | thats not possible Gugge |
| 19:46.03 | Gugge | Node_403: neither is 32 channels on 300kbps |
| 19:46.21 | WIMPy | It should be possible. |
| 19:46.29 | Node_403 | Its happening. |
| 19:46.33 | WIMPy | But probably hard to understand. |
| 19:46.47 | Node_403 | :) |
| 19:48.09 | Node_403 | I also dont have public ip |
| 19:48.17 | drmessano | I can see doing a little less than 1/2 of that with G729 |
| 19:49.27 | Node_403 | drmessano do u have experience with this? |
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| 19:50.01 | WIMPy | G.729 is still too big. |
| 19:50.16 | drmessano | For 32 calls, yes |
| 19:50.18 | drmessano | Thats what I said |
| 19:51.13 | drmessano | Roughly 13 calls if you use G729. You're going to need to develop a new codec. I suspect paying for more bandwidth will be cheaper |
| 19:51.27 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:51.32 | WIMPy | There alre suitable codecs. |
| 19:51.40 | drmessano | Node_403, this isnt about experience, its about math. |
| 19:51.49 | Node_403 | okk |
| 19:51.50 | WIMPy | Speex, ilbc, celt, ... |
| 19:52.23 | Node_403 | My GSM-VoIP gateway doesnot support those |
| 19:52.30 | WIMPy | But at max 8kbps the sound will not be good. |
| 19:52.36 | Node_403 | My GSM-VoIP gateway doesnot support those codecs |
| 19:52.57 | drmessano | Speex and iLBC are both going to be roughly the same as G729 |
| 19:53.16 | WIMPy | You can have them at lower bit rates. |
| 19:53.21 | WIMPy | So what does your GW support? |
| 19:53.38 | Node_403 | g729. |
| 19:53.52 | Node_403 | amoung lower rates |
| 19:53.59 | WIMPy | The bad thing is that you will have to use RTP with it, so that already wastes half the bandwidth. |
| 19:55.02 | WIMPy | Can it do G.729 at 6.4kbps? |
| 19:55.31 | Node_403 | g729d? |
| 19:55.43 | WIMPy | yes |
| 19:55.48 | Node_403 | no |
| 19:56.35 | Node_403 | drmessano IAX possible? |
| 19:56.46 | WIMPy | Should make about 12 calls possible. |
| 19:57.03 | WIMPy | IAX helps, but I'd be surprised if you GW can do it. |
| 19:57.37 | Node_403 | i can have asterisk setup |
| 19:57.49 | WIMPy | on both ends? |
| 19:58.02 | Node_403 | hmm |
| 19:58.33 | drmessano | IAX will help with the overhead, but you're talking about maybe a difference of 1 call. 12 or 13 calls is still far from your 32 |
| 19:59.06 | WIMPy | No, with an IAX2 trunk it should be abour twice as much. |
| 19:59.06 | Node_403 | will new codec help? |
| 19:59.25 | jpsharp | You can also set your codec to higher packet sizes to help offset overhead. |
| 19:59.28 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:59.51 | WIMPy | According to the asteriskguru calculator that should fit easily. But that seems a little optimistic. |
| 20:00.26 | Node_403 | WIMpY this is happening really |
| 20:00.35 | Node_403 | you may be right |
| 20:01.03 | WIMPy | Yes, I think it would really fit. |
| 20:01.26 | drmessano | The calculator doesn't take overhead into account. You're dividing 24 into 300, not 8 into 300 |
| 20:02.06 | WIMPy | It does. |
| 20:02.45 | drmessano | According to my calculator, 24 fits into 300 about 12 times |
| 20:02.49 | WIMPy | What's that 24? |
| 20:03.33 | drmessano | 24 is the bandwith consumed by a G729 call at 8/8, which is how it is implemented in Asterisk |
| 20:04.04 | WIMPy | We already found out it's a no go thing with RTP. But with an IAX2 trunk it should be ok. |
| 20:07.06 | Node_403 | WIMPy said hi |
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| 20:08.33 | drmessano | So you have 300K up and 300K down? |
| 20:09.54 | WIMPy | I actually wonder why you connect a GSM gateway to such a line. |
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| 20:27.13 | uskerine | hi, exten => s,2,Dial(DAHDI/g1,20,rt) |
| 20:27.23 | uskerine | which is the equivalent action to Dial a SIP extension? |
| 20:28.22 | jpsharp | Dial(SIP/peername) |
| 20:28.40 | jpsharp | or Dial(SIP/extenstion@peername) |
| 20:29.40 | uskerine | if the SIP exteions hangup, it does not dial |
| 20:29.44 | uskerine | sorry |
| 20:29.54 | uskerine | if the SIP extensions hangs up, the call is not finished |
| 20:30.06 | uskerine | what should I do? -I am dialing from external POTS) |
| 20:31.28 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 20:32.48 | jpsharp | If the SIP extension hangs up, then either the call can disconnect or it will fall through to the next step in the dialplan, depending on how you have autofallthrough configured. |
| 20:33.31 | uskerine | so if autofallthrough is equal to "no" |
| 20:33.38 | uskerine | then if SIP extensions hangs up the call should end? |
| 20:34.41 | uskerine | it does not end the call |
| 20:34.47 | uskerine | i have the following setup |
| 20:35.02 | uskerine | PSTN ----> Digium TDM410 analog card ---> SIP extensions |
| 20:35.18 | uskerine | exten => s,1,Answer() |
| 20:35.22 | uskerine | exten => s,n,Wait(2) |
| 20:35.26 | uskerine | exten => s,n,Playback(vm-tempgreeting) |
| 20:35.30 | uskerine | exten => s,n,Dial(SIP/30,30) |
| 20:35.36 | uskerine | exten => s,n,Hangup() |
| 20:35.47 | uskerine | i would like once the SIP/30 hangs up, then the call is droped |
| 20:36.04 | uskerine | SIP/30 is the called party |
| 20:38.08 | uskerine | also i have another question, how should I set up the Dial action to allow SIP extensions use POTS Digium line? |
| 20:38.17 | uskerine | exten => _9.,1,Dial(DAHDI/g0/www${EXTEN:1}) |
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| 20:40.57 | jpsharp | The call should disconnect after SIP/30 hangs up. |
| 20:41.53 | jpsharp | And that dial string for DAHDI/g0 is the right thing. |
| 20:46.56 | uskerine | what does g0 and www means? |
| 20:46.58 | uskerine | what about EXTEN:1? |
| 20:47.03 | uskerine | what about _9? |
| 20:47.27 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 20:47.34 | uskerine | i would like to make the typical 0 or 9 prefix for calling outside |
| 20:47.48 | uskerine | so you press 0 or 9 and you get dial tone |
| 20:47.55 | WIMPy | That are typical use cases for the book. |
| 20:48.15 | WIMPy | Can your phone do that? |
| 20:48.38 | WIMPy | Most SIP phones seem to only be able to dial en-block. |
| 20:49.02 | uskerine | i don't know |
| 20:49.05 | uskerine | i am using a softphone |
| 20:49.16 | uskerine | it is ok if you have to dial with 9 as prefix |
| 20:49.24 | uskerine | and the asterisk will route the call removing that 9 or 0 prefix |
| 20:49.45 | WIMPy | Yes, but you won't get a dialtone in between. |
| 20:50.12 | WIMPy | You enter the number and then send it. |
| 20:50.56 | uskerine | it is ok if i don't get dialtone in between |
| 20:51.09 | uskerine | but that DIAL is not working |
| 20:51.38 | uskerine | i get Everyone is busy/congested at this time (1:0/0/1) |
| 20:51.39 | WIMPy | In what way? |
| 20:52.54 | uskerine | -- Executing [9nnn@phones:2] Dial("SIP/20-00000009", "SIP/9n,50") in new stack |
| 20:53.05 | uskerine | where nnn is the phone number i dialed |
| 20:53.19 | *** join/#asterisk gusto (~gusto@ip-109-84-0-68.web.vodafone.de) |
| 20:54.27 | uskerine | Executing [9610101010@phones:1] Dial("SIP/20-0000000c", "DAHDI/g09610101010") in new stack |
| 20:54.35 | uskerine | WARNING[1574]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 20:54.42 | uskerine | <PROTECTED> |
| 20:54.49 | uskerine | -- Executing [9610101010@phones:2] Congestion("SIP/20-0000000c", "") in new stack |
| 20:55.07 | uskerine | i would like SIP extension to DIAL 610101010 using POTS |
| 20:55.27 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 20:55.37 | uskerine | exten => _9.,1,Dial(DAHDI/g0${EXTEN}) |
| 20:55.37 | uskerine | exten => _9.,2,Congestion |
| 20:55.51 | WIMPy | There's a / missing between g0 and the destination. |
| 20:56.22 | uskerine | should it be exten => _9., 1, Dial(DAHDI/g0/${EXTEN}) |
| 20:56.25 | uskerine | what g0 means? group 0? |
| 20:56.37 | WIMPy | yes an yes |
| 20:56.43 | WIMPy | and |
| 20:56.43 | uskerine | ok |
| 20:56.51 | uskerine | what should i do to remove the dial prefix (9) |
| 20:56.55 | uskerine | ? |
| 20:57.11 | WIMPy | :1 |
| 20:57.21 | WIMPy | Or better read the book. |
| 20:57.54 | uskerine | i will also do it |
| 20:57.55 | uskerine | :) |
| 20:58.59 | uskerine | -- Executing [9610101010@phones:1] Dial("SIP/20-0000000f", "DAHDI/g0/610101010") in new stack |
| 20:59.03 | uskerine | [Jun 17 22:58:45] WARNING[1580]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 20:59.14 | uskerine | <PROTECTED> |
| 20:59.14 | uskerine | <PROTECTED> |
| 20:59.32 | uskerine | why does asterisk detect busy/congestion? |
| 21:00.02 | Chainsaw | uskerine: You need to dial the verbosity all the way up. |
| 21:00.04 | WIMPy | Does group 0 exist? |
| 21:00.37 | uskerine | i have group 0 configured in dahdi-channels.conf |
| 21:01.00 | uskerine | wow |
| 21:01.03 | uskerine | it was group 1 |
| 21:01.04 | uskerine | hold on |
| 21:01.05 | uskerine | i will retry |
| 21:01.56 | uskerine | it works now :) |
| 21:01.58 | uskerine | thanks a lot |
| 21:02.12 | uskerine | one question |
| 21:02.17 | uskerine | if somebody calls from a POTS |
| 21:02.26 | uskerine | then asterisk answers and redial to an internal extension |
| 21:02.30 | uskerine | and during the call |
| 21:02.40 | uskerine | the called SIP extension hangs up |
| 21:03.04 | uskerine | should the call be droped or not (i remember the caller was the one who could actually drop the call when using POTS to POTS call) |
| 21:03.28 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 21:04.33 | Chainsaw | uskerine: That way round should be easy. Once the called internal party hangs up, the call is terminated. |
| 21:04.48 | jpsharp | It will hang up unless you pass the "g" option to Dial. |
| 21:04.51 | Chainsaw | uskerine: It's when the remote POTS caller hangs up and you don't notice that things get interesting. (And that will only happen on analog lines without proper disconnect supervision) |
| 21:06.03 | uskerine | so you think it should be noticed all time |
| 21:06.07 | uskerine | right? |
| 21:06.27 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 21:06.53 | Chainsaw | uskerine: It depends, are your lines analog or ISDN? |
| 21:06.57 | uskerine | analog |
| 21:07.04 | uskerine | POTS! |
| 21:07.05 | uskerine | :) |
| 21:07.20 | Chainsaw | uskerine: POTS is a meaningless abbrevation. |
| 21:07.39 | uskerine | pots means analog |
| 21:07.42 | Chainsaw | uskerine: Some call ISDN POTS. As they apply it to anything non-VoIP. |
| 21:08.12 | uskerine | isdn is isdn |
| 21:08.15 | Chainsaw | uskerine: If the caller hangs up before your internal caller, you need disconnect supervision (such as a polarity switch from the exchange) to make you aware. |
| 21:08.17 | uskerine | pots is plain old telephony system |
| 21:08.22 | uskerine | for me POTS is analog, anyway |
| 21:08.24 | uskerine | i have analog |
| 21:08.35 | Chainsaw | uskerine: And since you are stuck on arguing terminology, I have now lost interest. Next. |
| 21:08.41 | uskerine | i have configured polatiry switch, but i am not sure my operator imlpements it |
| 21:08.46 | uskerine | i read on the web it does (it is telefonica spain) |
| 21:08.59 | WIMPy | "POTS" was invented to make the difference from ISDN, so calling ISDN POTS is pretty silly. |
| 21:09.24 | uskerine | i will later move into ISDN, (PRI) |
| 21:09.32 | uskerine | but for now i am configuring a simple card with 4 analog lines |
| 21:09.35 | Chainsaw | WIMPy: Yet some insist on doing it, so one best verifies such assumptions. |
| 21:09.37 | uskerine | as PRI will take long time to arrive |
| 21:10.32 | uskerine | if you want to make me happy, how could i implement a group call? so 4 or 5 SIP extensions will ring -unless they are busy- until one of them answers? |
| 21:10.37 | uskerine | :) |
| 21:10.45 | WIMPy | I haven't even seen it used that way in here. And terminology is often very interesting in #asterisk. |
| 21:10.47 | uskerine | and i promise to read the book too:) |
| 21:11.04 | Chainsaw | uskerine: Queues should do what you want. |
| 21:11.18 | WIMPy | Find the chapter where you find & ind the Dial parameter. |
| 21:11.22 | Chainsaw | uskerine: You can pick your algorithm, ring each in term until someone answers, ring them all at once, etc. |
| 21:11.30 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:11.32 | *** join/#asterisk psilvao (~psilvao@190.22.86.2) |
| 21:11.42 | uskerine | all of them should ring all at once |
| 21:11.45 | uskerine | but before moving into that |
| 21:11.58 | uskerine | i noticed that when i dial from the SIP extension towards the POTS analog line |
| 21:12.03 | uskerine | i can't hear the dial tone |
| 21:12.05 | Chainsaw | uskerine: (Downside of ringing them all at once, particularly on actual SIP handsets, is that it leaves "missed calls" on the menu screen) |
| 21:12.31 | uskerine | it will be a small call center with 5 guys, what do you recommend? |
| 21:12.34 | WIMPy | Err, that has been fixed many versions ago. |
| 21:12.57 | WIMPy | Must have been in the beginning of 1.6 I think. |
| 21:12.58 | uskerine | WIMpy maybe it is an issue with the softphone, rather than with asterisk |
| 21:13.07 | uskerine | i will try with different one |
| 21:13.19 | WIMPy | was referring to the missed calls. |
| 21:13.42 | Chainsaw | WIMPy: Happens to this day on Asterisk 10.5.1 with Polycom 670 handsets. |
| 21:13.44 | uskerine | ok |
| 21:14.02 | WIMPy | And with SIP you can;t get dialtone, as you (usually) open the channel after you finished dialling. |
| 21:14.04 | Chainsaw | WIMPy: So whatever "bug" you think you've "fixed"... I do not agree with you. |
| 21:14.14 | uskerine | so when i dial from softphone (sip extension) to an external POTS number, i can not hear the dialing tone on the SIP phone |
| 21:14.18 | WIMPy | It has been fixed in Asterisk. |
| 21:14.21 | uskerine | exten => _0.,1,Dial(DAHDI/g1/${EXTEN:1}) |
| 21:14.27 | uskerine | should something be added to the Dial action? |
| 21:14.33 | WIMPy | If Polycoms have a bug, that's another thing. |
| 21:14.44 | Chainsaw | WIMPy: There is no bug. A call came in that was not answered. That is a missed call. |
| 21:14.54 | Chainsaw | WIMPy: Which the phone will store. |
| 21:15.21 | WIMPy | If it is anwered someewhere it has been answered. |
| 21:15.47 | Chainsaw | WIMPy: You are confusing in-Asterisk signalling with on-phone UI actions. |
| 21:16.09 | WIMPy | Asterisk tells the phone. |
| 21:16.18 | WIMPy | If the phone doesn;t understand, replace it with a decent one. |
| 21:16.48 | Chainsaw | WIMPy: There is no status header in SIP for queue calls as opposed to regular calls. It rang, it stopped ringing and in that time, you did not answer it. That is a missed call. |
| 21:17.04 | Chainsaw | WIMPy: But you're getting side-tracked from answering uskerine's problem. I would suggest you get back on track. |
| 21:17.16 | uskerine | thanks to both of you! |
| 21:17.24 | uskerine | so i call from SIP extension, the call is routed |
| 21:17.34 | uskerine | but i can't hear anything in the SIP phone |
| 21:17.42 | WIMPy | It works with other phones. |
| 21:17.42 | uskerine | so my users will get puzzled |
| 21:17.43 | Chainsaw | uskerine: Right. So is it that second dial-tone after hitting the initial 9 or 0 that you wanted? |
| 21:17.50 | uskerine | no no |
| 21:17.51 | uskerine | i am fine |
| 21:17.51 | Chainsaw | uskerine: Or are you missing the progress tones? |
| 21:17.56 | uskerine | with the 0 or 9 prefix |
| 21:18.01 | uskerine | i miss the progress tones |
| 21:18.07 | uskerine | sorry i was looking for English word |
| 21:18.10 | Chainsaw | uskerine: You may need progressinband=yes |
| 21:18.25 | Chainsaw | uskerine: Normally the softphone would create these tones locally, based on SIP status codes. |
| 21:18.39 | uskerine | ok i will try that |
| 21:18.40 | Chainsaw | uskerine: But if this one doesn't, progressinband=yes on the relevant SIP peer(s) will have Asterisk generate them for you. |
| 21:18.54 | Chainsaw | uskerine: Make sure your indications.conf is set up for the right country, or the tones are going to sound all American and thus confusing. |
| 21:19.29 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:19.45 | uskerine | so progressinband=yes should be applied to sip.conf? |
| 21:19.57 | Chainsaw | uskerine: Yes, but not globally. Just to the SIP peers that need it. |
| 21:20.03 | uskerine | all of them |
| 21:20.04 | Chainsaw | uskerine: It takes CPU time on your Asterisk server to generate these. |
| 21:20.08 | uskerine | i have only one softclient |
| 21:20.10 | uskerine | no ip phones |
| 21:20.17 | uskerine | i have a decent server |
| 21:20.25 | uskerine | it should not be an issue |
| 21:20.30 | Chainsaw | uskerine: Okay. Up to you. |
| 21:20.38 | uskerine | but thanks for the advice |
| 21:21.01 | WIMPy | The audio should come from the dahdi channel, shouldn't it? |
| 21:21.10 | Chainsaw | WIMPy: Once the call is established, yes. |
| 21:21.20 | WIMPy | And also before. |
| 21:21.25 | Chainsaw | WIMPy: We are expecting *dial tone* *number entry* *progress tones* *called party answers* |
| 21:21.47 | WIMPy | The dialtone has to come from the client in any case. |
| 21:22.03 | WIMPy | Everything after dialling should come from the line. |
| 21:22.24 | WIMPy | Otherwise you'd never get any announcements. |
| 21:22.49 | WIMPy | Although that can be some sort of issue anyway. |
| 21:23.06 | Chainsaw | WIMPy: Well, either you need to generate your progress tones in-band or you have a NAT problem. |
| 21:23.28 | WIMPy | huh? |
| 21:24.26 | Chainsaw | WIMPy: Either I am right and the progress tones need to be generated locally, in which case progressinband=yes will fix it. Or you are right and all has to come from the analog line immediately. |
| 21:24.49 | Chainsaw | WIMPy: In which case you are losing audio streams, and that is a NAT problem so often that I can't even be bothered to express that mathematically. |
| 21:25.08 | uskerine | [general] |
| 21:25.12 | uskerine | progressinband=yes |
| 21:25.13 | WIMPy | I'm not saying you don't need progressinband, but I don't see Asterisk having to generate anything. |
| 21:25.17 | uskerine | and sip.reload in cli |
| 21:25.21 | Chainsaw | WIMPy: You could also fix it by switching to IAX2, if it is a NAT problem. |
| 21:25.35 | Chainsaw | WIMPy: SIP and cheap nasty routers doesn't really mix, after all. |
| 21:25.39 | uskerine | but i can not hear progression tones |
| 21:25.47 | uskerine | no NAT issue |
| 21:25.55 | Chainsaw | You know what... |
| 21:25.56 | uskerine | everything is on the same subnet |
| 21:25.58 | uskerine | :) |
| 21:26.11 | Chainsaw | Since uskerine & WIMPy are both so sure that they can second-guess me on every suggestion... |
| 21:26.21 | Chainsaw | I am going to watch Inception like I originally planned and leave it with you two. |
| 21:26.26 | Chainsaw | Enjoy. |
| 21:26.36 | uskerine | chainsaw please do not leave, i really thank you for your help! :! |
| 21:26.40 | uskerine | just trying to debug this |
| 21:26.45 | uskerine | omg |
| 21:27.04 | uskerine | WIMPy, "progressinband=yes" does not work |
| 21:27.15 | WIMPy | thinks SIP only really mixes well with a good rant. |
| 21:27.27 | uskerine | what is the meaning of rant? |
| 21:27.28 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:27.49 | WIMPy | Expressing some strong feelings about it. |
| 21:28.10 | uskerine | hehe |
| 21:28.31 | uskerine | actually, the only two things i am missing now is the call progress tone |
| 21:28.49 | uskerine | and the group call, but that will come later (once i read something about queing as you suggested) |
| 21:29.02 | uskerine | no progress tone will definitively puzzle users |
| 21:29.20 | WIMPy | That was Chainsaw. i suggested a list to Dial. |
| 21:29.53 | WIMPy | There's a lot of things to get used to when using VOIP and/or Asterisk. |
| 21:30.25 | uskerine | i added "r" to Dial |
| 21:30.28 | uskerine | i will give it a try |
| 21:30.50 | WIMPy | That will generate a ringback tone. |
| 21:32.29 | uskerine | Dial(SIP/maxotel/0746000000,120,r) |
| 21:32.30 | uskerine | (This calls a number for 120 seconds and plays a fake ring tone to your phone while its ringing) |
| 21:34.26 | uskerine | now it works |
| 21:34.37 | uskerine | but the progress tone is very low |
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| 21:39.00 | *** join/#asterisk Neptu (~Neptu@41.141.26.27) |
| 21:39.34 | volga629 | Some message about imap storage app_voicemail_imapstorage.c:1800 vm_imap_delete: msgnum 0, mailbox message 0 is zero. ???? |
| 21:40.05 | volga629 | It delivery only to INBOX, but set to VoiceMail |
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| 21:44.08 | uskerine | WIMPy |
| 21:44.20 | uskerine | terminating the Dial function with 30,r (30 seconds, ring tone) |
| 21:44.22 | uskerine | works like a charm |
| 21:44.28 | uskerine | now i can hear the calling progress tone |
| 21:44.28 | uskerine | :) |
| 21:45.02 | gusto | hey, does some one use this codec_ilbc.so? |
| 21:45.03 | WIMPy | Might be better to hear the real thing. |
| 21:45.12 | uskerine | yes i agree |
| 21:45.35 | WIMPy | There's probably somethign to enable in chan_dahdi.conf. There's always at least one catch in that file. |
| 21:49.16 | uskerine | i will investigate :) |
| 21:49.19 | uskerine | thanks WIMPy |
| 21:49.28 | uskerine | for today this is enough, at least i can receive and make calls |
| 21:49.31 | uskerine | things are progressing |
| 21:49.32 | uskerine | :) |
| 21:49.32 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:49.41 | uskerine | tomorrow i will fine tune |
| 21:49.47 | uskerine | and in parallel i have to read the book |
| 21:49.53 | uskerine | i am progressing on that too |
| 21:50.43 | uskerine | cu, thanks for your help WIMPy |
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| 21:55.03 | volga629 | gusto thank to suggestion |
| 21:56.10 | volga629 | I set ip, but in my case SRTP constantly crash, I disabled and see what developer will say about backtrace |
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| 22:25.57 | gusto | i do not understand that codecs here |
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| 22:50.22 | dj_hamsta | i need a voip provider in spain for a free number.. or paid thats fine lol |
| 22:50.26 | gusto | does someone have an ATA with GSM codec support? or someone an idea what the GSM codec is? |
| 22:50.45 | gusto | spain is broke |
| 22:51.26 | WIMPy | The GSM codec referred to in VOIP stuff is the FR codec used in the early years of GSM. |
| 22:52.32 | gusto | does it have a G7XX number? |
| 22:53.43 | WIMPy | No |
| 22:53.52 | WIMPy | It has a GSM number: 06.10 |
| 22:54.12 | WIMPy | Or an ETSI number: 300 961 |
| 22:55.07 | gusto | tss |
| 22:55.26 | gusto | to my PAP2T and to SPA112 GSM is not supported, seems so |
| 22:55.27 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 22:56.04 | [TK]D-Fender | Extremely few ATA's support anything other than G.711/723/729 |
| 22:57.06 | gusto | well, this 723 is there |
| 22:57.19 | gusto | and on the SPA only 726 |
| 22:57.35 | gusto | but on the old pap2t i have also g723.1 |
| 22:58.28 | [TK]D-Fender | G.729 is the "light" codec of choice. |
| 22:58.43 | gusto | when it works |
| 22:59.01 | gusto | there is g729a |
| 23:00.25 | [TK]D-Fender | I've never seen it "not work" |
| 23:00.30 | gusto | well |
| 23:00.47 | gusto | g729a you can not reencode with asterisk w/o a license, right? |
| 23:01.01 | WIMPy | There are many versions of G.729. |
| 23:01.09 | gusto | so ... only pass through works and ... what *** VoIP provider supports g729a |
| 23:01.10 | gusto | ha? |
| 23:01.25 | WIMPy | sipgate |
| 23:02.07 | [TK]D-Fender | That isn't "doesn't work |
| 23:02.25 | [TK]D-Fender | <PROTECTED> |
| 23:02.41 | [TK]D-Fender | The tmers of conversion are clear |
| 23:02.44 | [TK]D-Fender | terms* |
| 23:03.12 | [TK]D-Fender | Any real provider supports G.729 |
| 23:03.21 | volga629 | ast_unload_resource: Firm unload failed for codec_g729a ? |
| 23:03.25 | [TK]D-Fender | It is standard. Several support that and NOT even G.711 |
| 23:03.27 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 23:05.04 | gusto | ah |
| 23:05.06 | gusto | cool |
| 23:05.24 | volga629 | noload => codec_g729.so is this should help ? |
| 23:07.50 | [TK]D-Fender | How does that "help"? |
| 23:08.59 | gusto | well, personal-voip supports only G.711 alaw, G.711 ulaw, GSM, iLBC |
| 23:09.10 | gusto | and that is good ;-) |
| 23:09.19 | [TK]D-Fender | So? that's why you set the codecs in YOUR TRUNK |
| 23:09.29 | gusto | but my ATA's do not support iLBC or GSM |
| 23:09.36 | [TK]D-Fender | You don't kill off a module just because ONE SERVICE doesn't need it |
| 23:09.53 | [TK]D-Fender | Configure your extensions & trunks |
| 23:09.55 | gusto | i am not killing anything, you mix me up with volga629 maybe |
| 23:10.24 | [TK]D-Fender | Sorry, yes... |
| 23:10.25 | gusto | does not even have a codec_g729 |
| 23:10.31 | [TK]D-Fender | not enough coffe and close timing. |
| 23:10.36 | [TK]D-Fender | Darn quick mixup |
| 23:10.42 | gusto | i do not drink coffee |
| 23:10.53 | volga629 | sorry interruption |
| 23:10.55 | [TK]D-Fender | Never too soon, or too late! |
| 23:10.57 | gusto | but i am weak ... maybe i should go to sleep |
| 23:11.14 | [TK]D-Fender | volga629: I still don't see a problem.... |
| 23:11.27 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 23:11.39 | volga629 | can load properly g729 |
| 23:11.43 | gusto | [TK]D-Fender: i see you have still yeasterday |
| 23:11.48 | volga629 | I have license |
| 23:11.57 | [TK]D-Fender | sut? |
| 23:12.09 | volga629 | and it was loading, but today complain that can't reload |
| 23:13.49 | [TK]D-Fender | volga629ast_unload_resource: Firm unload failed for codec_g729a ? |
| 23:13.53 | [TK]D-Fender | that is an unload. |
| 23:14.06 | [TK]D-Fender | "core show translation" |
| 23:14.08 | [TK]D-Fender | ? pb |
| 23:14.13 | [TK]D-Fender | ~pb |
| 23:14.14 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:14.15 | [TK]D-Fender | ^^^ |
| 23:14.37 | volga629 | that I get the message |
| 23:15.26 | WIMPy | When do you get that message? |
| 23:15.57 | volga629 | http://fpaste.org/URYn/ |
| 23:16.22 | volga629 | I so translation stop working and tried reload g729 |
| 23:16.58 | WIMPy | How? |
| 23:17.05 | WIMPy | Did you try to unload it? |
| 23:17.09 | volga629 | yes |
| 23:17.39 | volga629 | http://fpaste.org/3qPk/ |
| 23:17.55 | volga629 | And tell that module loaded |
| 23:18.34 | volga629 | I remove any settings related to g729 |
| 23:19.30 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 23:19.44 | volga629 | Only license file shows ok |
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| 23:30.27 | volga629 | yes it working again I was need put unload and move file out reboot and reload again strange but it worked |
| 23:34.19 | volga629 | Still see channel.c:5218 set_format: Unable to find a codec translation path from (g729) to (slin) |
| 23:35.20 | WIMPy | TOo many channels? |
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| 23:35.29 | volga629 | need reboot again and don't see in translations ? |
| 23:35.34 | volga629 | no only one |
| 23:36.30 | WIMPy | Maybe you should call Digium later on that one. |
| 23:37.10 | volga629 | too match staff on plate today which not working :-) all vpn user will screw up today |
| 23:41.01 | volga629 | Look like again license -- Channels: 0 (incorrect host-id) |
| 23:41.30 | WIMPy | Again? |
| 23:41.34 | volga629 | and again error of limitation, I just called them yesterday |
| 23:41.59 | volga629 | everything was working and right same story :-) |
| 23:43.27 | volga629 | Server response: 469 - Maximum Activations Reached. It just make me smile, same server same codec |
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| 23:44.47 | drmessano | So its already been activated |
| 23:46.10 | volga629 | yes, why it yesterday I called support asked to reactivated and today again it asking for activation, again was working :-) |
| 23:46.47 | drmessano | Can asterisk read the license file? Check the perms on /var/lib/asterisk/licenses/ |
| 23:49.49 | volga629 | yes it reading file no issue capbxsrv01*CLI> g729 show licenses 0/0 encoders/decoders of 0 licensed channels are currently in use Licenses Found: File: G729-XXXXX.lic -- Key: G729-XXXX -- Host-ID: -- Channels: 0 (incorrect host-id) |
| 23:50.31 | volga629 | ok anyway I will call support on Monday again |
| 23:50.35 | drmessano | Is this a VM? |
| 23:50.40 | volga629 | yes |
| 23:50.44 | volga629 | ESXi |
| 23:51.24 | drmessano | Is this a new VM or have you reconfigured the Network interfaces? |
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| 23:51.54 | drmessano | The activation is tied to the MAC address of the ethernet interfaces |
| 23:52.09 | drmessano | If thats changed, the license wont work |
| 23:55.40 | volga629 | yes I know that, but this is prod and we no maded any changes which I am aware off |
| 23:56.12 | volga629 | But I changed to ulaw for right now and will deal on Monday |
| 23:59.28 | volga629 | drmessano thanks for ideas and time, I have some issue with IMAP storage it not prod, but we want implement |
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