| 00:05.26 | *** join/#asterisk Vann (~manny@c-76-97-51-58.hsd1.ga.comcast.net) |
| 00:28.07 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
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| 00:46.28 | hebber | Hi, I have a problem with registering to a trunk using callbackextension using realtime peers. I believe the problem could be the name of auth or authuser or authname. Documentation is confusing. It does work if I add a register in sip.conf. Could anyone point me to an working example or documentation? |
| 00:47.43 | hebber | I end up with state=unregistered in sip show registry |
| 00:48.57 | hebber | callbackextension works as intended, just it don't want to register without the register in sip.conf |
| 00:49.09 | Penguin | You don't register to a trunk. You register to a peer. |
| 00:49.22 | Penguin | The only way to register to a peer is with a register statement. |
| 00:50.10 | hebber | no - from version 10 upwards, then need of register in sip.conf is not needed |
| 00:50.38 | Penguin | What you're saying is that the realtime register statement doesn't work? |
| 00:50.38 | hebber | you can initialize it with a callbackextension in sippeers |
| 00:51.02 | hebber | yes, maybe I was unclear |
| 00:52.04 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-104-246-231.ph.ph.cox.net) |
| 00:52.42 | hebber | hehe no - there is no need to register=>, but use a callbackextension instead. Let me find the documenation I have |
| 00:56.00 | hebber | penguin: I don't find the docs from Asterisk now - but here is a guide that points it out: http://www.fussylogic.co.uk/blog/?p=441 |
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| 00:57.12 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:58.50 | Penguin | I don't see anything there that mentions what you were talking about. Additionally, I can't trust the author of that to be accurate. |
| 01:00.30 | hebber | hmm - I will dig harder - from sip.conf v11.5 - ;callbackextension=123 ; Register with this server and require calls coming back to this extension |
| 01:05.35 | [TK]D-Fender | that has its downsides as well |
| 01:05.40 | WIMPy | Both ways have issues. |
| 01:06.35 | [TK]D-Fender | Generally more negative than positive |
| 01:07.02 | *** join/#asterisk CeBe (~CeBe@port-92-206-207-153.dynamic.qsc.de) |
| 01:09.01 | hebber | If you referred to my case, whats the advised way to go, when I have a SIP trunk provider with multiple accounts and same IP address. |
| 01:09.36 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-15-91.dynamic.qsc.de) |
| 01:09.55 | hebber | I liked the callbackextension approach as you write the config once - not twice |
| 01:09.57 | [TK]D-Fender | hebber: And if you knew how calls were authed you would know that it doesn't amke a difference HOW you register to them |
| 01:10.33 | [TK]D-Fender | There is no "twice" with a separate register either. |
| 01:10.51 | crumb | /usr/lib/asterisk/modules/codec_g729.so: cannot open shared object file: No such |
| 01:10.54 | crumb | <PROTECTED> |
| 01:11.06 | [TK]D-Fender | crumb: If it says it's not there ... then it's not there |
| 01:11.08 | hebber | Thats is exactly my problem, I'm confused with user, peer and friends |
| 01:11.14 | crumb | it is there |
| 01:11.19 | [TK]D-Fender | crumb: Show us |
| 01:11.29 | [TK]D-Fender | crumb: Also with all its neighbours |
| 01:11.31 | WIMPy | permissions? |
| 01:11.33 | crumb | ls -al /usr/lib/asterisk/modules/codec_g729.so |
| 01:11.44 | [TK]D-Fender | WIMPy: Shush you! I'm giving rope! |
| 01:11.48 | crumb | -rw-r--r-- 1 root root 99172 Sep 22 20:17 /usr/lib/asterisk/modules/codec_g729.so |
| 01:11.55 | Penguin | user, peer, friend... that's pretty easy to understand. I'll explain it in simple form. |
| 01:12.08 | [TK]D-Fender | crumb: ls -la /usr/lib/asterisk/modules/ |
| 01:12.11 | [TK]D-Fender | crumb: the WHOLE folder |
| 01:12.23 | [TK]D-Fender | crumb: AND ... "core show settings" |
| 01:12.24 | WIMPy | No need |
| 01:12.30 | WIMPy | It's not executable. |
| 01:12.35 | [TK]D-Fender | crumb: indeed ROOT should be a cliue |
| 01:12.40 | [TK]D-Fender | as well |
| 01:13.07 | Penguin | Using type=peer causes matching by IP/port. user matches by username. friend creates both a peer and a user and attempts to match incoming calls by username first; outbound calls will be treated like type=peer. |
| 01:13.21 | crumb | http://pastebin.ca/2457059 |
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| 01:14.42 | crumb | http://pastebin.ca/2457060 |
| 01:16.11 | crumb | wtf |
| 01:16.17 | crumb | the whole thing didn't paste |
| 01:16.31 | hebber | penguin: so for a case of a SIP provider with same IP and I need to distinguish several account I would need to make type=friend so it matches by username first to separate the accounts. If an account have only type=user you cannot receive calls? |
| 01:16.37 | crumb | nvm it did |
| 01:16.49 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-55-108.dynamic.qsc.de) |
| 01:17.03 | Penguin | The chance of them authenticating calls to your asterisk with a username is pretty low. |
| 01:17.09 | *** join/#asterisk robert___ (~hellspawn@objectx/robert) |
| 01:17.11 | Penguin | Basically, it's not going to happen. |
| 01:17.21 | WIMPy | hebber: No, ITSPs usually don;t authenticate to you. You register different extensions. |
| 01:18.29 | Penguin | But having to user one extension per account is ridiculous. This is a major problem for a long time. People have been wanting to use multiple accounts for a long time. |
| 01:18.49 | hebber | ok, so the don't authenticate on INVITES, so I would use callbackextensions to separate the accounts? |
| 01:19.21 | crumb | [TK]D-Fender: would you like root access to the box? :/ |
| 01:19.25 | hebber | I know, but that is out of my control, thats how that provider operates |
| 01:19.38 | *** join/#asterisk suneye (~atcmmi@116.25.194.34) |
| 01:20.06 | WIMPy | hebber: That's the way, yes. |
| 01:20.25 | crumb | i don't understand why it says no such file when it exists |
| 01:20.36 | crumb | is it possible to get more verbosity from asterisk? |
| 01:21.06 | [TK]D-Fender | crumb: where's the other half of what I asked you for? |
| 01:21.18 | crumb | http://pastebin.ca/2457060 |
| 01:21.50 | hebber | ok thanks, then I just need to play with the settings |
| 01:26.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:30.41 | crumb | [TK]D-Fender: i didn't change anything else, except place the codec in the /usr/lib/asterisk/modules |
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| 01:30.53 | crumb | g723 and g729 |
| 01:31.00 | *** part/#asterisk imcdona1 (~Thunderbi@2001:470:e8f1:2:7847:3c92:2414:bafc) |
| 01:31.16 | *** join/#asterisk imcdona1 (~Thunderbi@2001:470:e8f1:2:7847:3c92:2414:bafc) |
| 01:31.31 | crumb | i get the same error with codec_g723.so as well |
| 01:31.36 | *** part/#asterisk imcdona1 (~Thunderbi@2001:470:e8f1:2:7847:3c92:2414:bafc) |
| 01:32.14 | WIMPy | Try ldd |
| 01:33.01 | crumb | not a dynamic executable :/ |
| 01:33.18 | iq | l |
| 01:33.19 | *** join/#asterisk imcdona (~Thunderbi@gateway.defiancedc.com) |
| 01:33.30 | WIMPy | That's not a good start. |
| 01:33.42 | WIMPy | What does file think it is? |
| 01:33.51 | crumb | i used a cross-compiler :/ |
| 01:33.57 | file | I think it's Sunday. |
| 01:34.05 | dlynes | Almost Monday |
| 01:34.09 | WIMPy | I guess you used it wrong. |
| 01:34.44 | Penguin | ldd tells me all about my codec g729.so |
| 01:35.52 | Penguin | And it's not an executable, it's an so. |
| 01:35.53 | dlynes | crumb, is it compiled for the same version of asterisk? is it compiled for the correct platform? 32-bit vs 64-bit? x86/x86_64 vs SPARC vs MIPS vs ARM vs ...? |
| 01:36.09 | dlynes | Penguin, a shared object is a special kind of executable |
| 01:36.20 | Penguin | It doesn't SAY it's an executable. |
| 01:36.33 | dlynes | Penguin, It speaks to you? |
| 01:36.40 | Penguin | It clearly says ELF 32-bit LSB shared object. |
| 01:36.47 | WIMPy | That's why It irritates me that they aren't executable. |
| 01:37.14 | Technodrome | is 20 ms jitter up bad? |
| 01:37.28 | dlynes | as i said...special kind of executable...you can't run it by itself, but it's still ELF format |
| 01:37.42 | *** join/#asterisk CeBe (~CeBe@port-92-206-132-104.dynamic.qsc.de) |
| 01:37.52 | Penguin | `file' would tell you what kind of file it is. |
| 01:38.01 | dlynes | And they should be chmod 0755 |
| 01:38.02 | WIMPy | Or AOF or whatever. |
| 01:38.07 | Penguin | It's not executable at all. It's a shared object. |
| 01:38.38 | WIMPy | Sure it's executable, just not on its own. |
| 01:38.56 | Penguin | Saying they need to be 755 would assume a few things that we didn't even talk about. |
| 01:40.04 | dlynes | crumb, you can do an strace against asterisk to see what it's looking for, and then see if that matches what you're providing |
| 01:40.23 | dlynes | crumb, maybe it's making some assumptions that don't match with what you're providing |
| 01:40.55 | WIMPy | If ldd can't handle the lib, how could Asterisk? |
| 01:40.58 | dlynes | crumb, and watching it with strace might make the light bulb go off in your head |
| 01:41.28 | Penguin | Usually, the light bulb needs to come ON. |
| 01:41.30 | dlynes | WIMPy, the /usr/lib/asterisk/modules files are not bound to asterisk...asterisk discovers them |
| 01:41.37 | Penguin | When it's off, there's a problem. |
| 01:41.40 | dlynes | WIMPy, so ldd won't show them |
| 01:42.07 | WIMPy | ldd on the .so |
| 01:42.09 | Penguin | ldd will show what the so is linked to. |
| 01:42.15 | dlynes | WIMPy, that's true |
| 01:42.18 | Penguin | If it doesn't, something is broken. |
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| 02:00.20 | crumb | WIMPy: sorry, misunderstood your question.. file /usr/lib/asterisk/modules/codec_g729.so |
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| 02:00.22 | crumb | /usr/lib/asterisk/modules/codec_g729.so: ELF 32-bit LSB shared object, ARM, version 1 (SYSV), dynamically linked, stripped |
| 02:01.00 | Penguin | The first question was what ldd says about it. |
| 02:01.16 | crumb | "not a dynamic executable" |
| 02:01.47 | Penguin | That's pretty weird. For me, ldd says the files that it is linked to. |
| 02:02.18 | Penguin | libc.so, ld-linux.so, etc. |
| 02:02.27 | crumb | yeah, the default codecs say so as well |
| 02:02.43 | crumb | do you happen to have a g729 codec on your system? |
| 02:03.03 | Penguin | Yes. |
| 02:03.13 | crumb | and it's linked? |
| 02:03.15 | crumb | k |
| 02:03.16 | Penguin | Yes. |
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| 02:48.34 | matrix1233 | hello |
| 02:48.47 | matrix1233 | am installing a realtime asterisk |
| 02:49.17 | matrix1233 | with res_config_mysql |
| 02:49.44 | matrix1233 | on localhost it"s work bun when i change the ip adress to a remote server i cant connect |
| 02:50.01 | matrix1233 | any one can tel me where is the problem ? |
| 02:55.55 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 02:58.23 | *** join/#asterisk Technodrome (~asdfasdf@unaffiliated/technodrome) |
| 03:18.06 | hebber | First you should check the network. telnet remoteip port etc. If no connection its pointless to check configuration in Asterisk. |
| 03:18.07 | Nugget | telnet is eeeeeeevil! |
| 03:39.18 | *** join/#asterisk jimfriendo (~jimtendo@sav1217145.lnk.telstra.net) |
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| 03:48.49 | jimfriendo | Hey guys. Does anyone know of any service which might allow me to connect my Asterisk PBX to the PSTN without requiring me to purchase and run a local Gateway/FXO? |
| 03:49.25 | WIMPy | ~itsp |
| 03:49.26 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 03:52.00 | jimfriendo | Thanks heaps WIMPy |
| 04:02.34 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-uovittbgkdlfircp) |
| 04:06.23 | Penguin | I wish someone would have told me about an ITSP back ten years ago. |
| 04:07.18 | WIMPy | Here's a deal: You get the time machine and I tell you. |
| 04:09.52 | Penguin | If you'll build it for me, I'll take a ride in it. |
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| 05:26.49 | ChannelZ | I wish someone told me just because it has lady-parts now, doesn't mean it started out as a lady |
| 05:27.20 | WIMPy | Found a surprize? |
| 05:29.31 | [TK]D-Fender | Had the body of a venus... |
| 05:29.32 | ChannelZ | What happens in Rio de Janeiro stays in Rio de Janeiro |
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| 05:48.05 | phix | back |
| 05:48.06 | carrar | heh |
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| 08:32.52 | phix | umm looks like you are dialing console instead of a SIP channel |
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| 08:36.04 | bulkorok | hi |
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| 09:01.59 | Scapal | Hi, anyone here already played with app.Originate() ? |
| 09:03.04 | Scapal | I am trying to use it with an app but can't figure out how to pass multiple params to that app |
| 09:06.36 | wdoekes | Scapal: no one uses pbx_lua |
| 09:07.14 | Chainsaw | wdoekes: That's not true, Scapal does! |
| 09:09.00 | Scapal | wdoekes: yes, and it works really fine. Why wouldn't someone use pbx_lua, it is fast and easy to maintain? |
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| 09:11.21 | Scapal | wdoekes: even so, the Originate application stays the same, I still have a problem to pass multiple params since the 4th arg is for app param but the 5th is ignored |
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| 09:12.08 | Scapal | is there a way to escape a ',' so that it is used by the called app, not the Originate |
| 09:15.14 | wdoekes | have you tried one or more backslashes? |
| 09:16.56 | wdoekes | Scapal: because using rarely used bits in asterisk always bites you |
| 09:17.32 | Scapal | wdoekes: well I tried one bs... and it needed a second one, it's working now :-) |
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| 09:20.01 | Scapal | wdoekes: I encourage you to use pbx_lua, I'm using it for a Business Center hosting 80 companies, everything is configured dynamically through a database, with on the fly custom dialplan loading for selected customers... And it is running for months without a single crash. How stable is that. |
| 09:22.22 | Scapal | wdoekes: what do you use for you dialplan? pbx_config, pbx_ael? |
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| 09:42.17 | wdoekes | regular pbx_config over static realtime. I use func_odbc for all my "dynamic" needs. we don't do customer specific dialplans as a rule |
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| 09:54.48 | eject_ck | Hi all |
| 09:55.26 | eject_ck | I'm trying to configure call forwarding calls to asterisk from tollfreeforwarding.com, does anybody have experience with it? |
| 09:56.25 | eject_ck | My issue is that I see calls to asterisk from multiple IP addresses, so I've added extensions to sip.conf with all IPs I see |
| 09:56.42 | eject_ck | However I see this in verbose console output |
| 09:56.44 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.60.234) |
| 09:56.46 | eject_ck | [Sep 23 12:47:06] NOTICE[20667]: chan_sip.c:22518 handle_request_invite: Sending fake auth rejection for device "TollFreeForwarding" |
| 10:03.28 | Scapal | wdoeskes: I have something like a thousand lines of lua code, also using odbc. If I had to maintain those in regular pbx_config, it would be a nightmare. Structured language vs C64 basic :-P |
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| 11:31.43 | *** join/#asterisk dose_ (54cf1787@gateway/web/freenode/ip.84.207.23.135) |
| 11:31.53 | dose_ | hi all |
| 11:33.40 | Greenlight | Howdy |
| 11:35.06 | *** join/#asterisk Technodrome (~asdfasdf@unaffiliated/technodrome) |
| 11:36.06 | dose_ | [asterisk 1.8.15] small issue maybe one of you faced, I'm trying to hide my callerid when calling from SIP through H323 trunk, but unfortunately my callerid is always sent, in my dialplan I use Set(CALLERID(num-pres)=prohib). I check tcpdump traces and saw that presentation is allowed. So it looks that prohib command are not passed to H323 protocol, any idea would be appreciated? |
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| 11:52.46 | Milos | How do I check the return value of Queue()? |
| 11:52.49 | Milos | in extensions.conf? |
| 11:52.57 | Milos | I've been googling but it doesn't seem to be stated anywhere... |
| 11:54.27 | kaldemar | what information are you really after? |
| 11:55.05 | Milos | the return value |
| 11:55.12 | Milos | This application returns -1 if the originating channel hangs up, or if the call is bridged and either of the parties in the bridge terminate the call. Returns 0 if the queue is full, nonexistent, or has no members. |
| 11:55.24 | Milos | I want to play a message if there are no available phones for the call |
| 11:55.29 | Milos | rather than just hang up abruptly |
| 11:55.37 | Milos | similarly I don't want the message to be heard if the call was answered |
| 11:55.49 | Milos | so I'm using GoToIf but I don't know what variable to check for the return value |
| 11:56.20 | Milos | I have exten => ...,n,Queue(default) |
| 11:56.28 | Milos | then after that I need to check the return value of Queue() |
| 11:56.30 | Milos | so how do I dothat |
| 11:56.32 | kaldemar | where did you copy that return value statement from? see "core show application Queue" for what QUEUESTATUS offers. |
| 11:56.33 | Milos | s/dothat/do that/ |
| 11:56.40 | Milos | http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue |
| 11:57.05 | Milos | core show application Queue look really helpful I'll have a read thanks |
| 11:58.03 | kaldemar | avoid voip-info when possible. |
| 11:58.06 | kaldemar | ~book |
| 11:58.06 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 11:58.13 | kaldemar | ~asteriskwiki |
| 11:58.13 | infobot | i guess asteriskwiki is http://wiki.asterisk.org |
| 11:58.38 | kaldemar | and of course your own asterisk provides documentation for your versions of applications and functions. |
| 12:02.33 | *** join/#asterisk danjenkins (~danjenkin@213.106.234.250) |
| 12:03.59 | *** join/#asterisk danjenkins_ (~danjenkin@213.106.234.250) |
| 12:06.08 | Milos | thanks kaldemar I will get that book then :) |
| 12:06.12 | Milos | I have another question while I'm here |
| 12:06.15 | Milos | how do I realod queues.conf? |
| 12:06.29 | Milos | when I try reload it it only seems to look at queuerules.conf |
| 12:06.39 | Milos | or maybe it's reloading both and only showing the warning for queuerules.conf |
| 12:06.46 | Milos | (warning is nothing changed) |
| 12:06.55 | Milos | s/realod/reload/ |
| 12:08.32 | kaldemar | Milos: queue reload all |
| 12:08.42 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 12:08.57 | Milos | if I want to find information about a command, where can I get that? |
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| 12:09.07 | Milos | gw1*CLI> queue reload all |
| 12:09.08 | Milos | [Sep 24 00:09:01] NOTICE[6465]: app_queue.c:7726 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. |
| 12:09.09 | kaldemar | Milos: core show help queue reload all |
| 12:09.13 | kaldemar | that for example. |
| 12:09.19 | Milos | as I said before - does this reload both and give warning for one? |
| 12:09.20 | kaldemar | core show help <command> |
| 12:09.50 | kaldemar | "all" reloads members, parameters and rules. |
| 12:09.59 | Milos | if I can give one complaint, it's inconsistency. why does sip reload not give any output but dialplan reload gives a success message? |
| 12:10.08 | Milos | yet they both complete successfully? |
| 12:10.31 | Milos | different module authors or what? |
| 12:10.39 | Milos | thanks for the command help |
| 12:11.31 | file | different module authors, different configuration usage system, different approach |
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| 12:12.10 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 12:38.47 | Milos | kaldemar, I still can't find information on what TIMEOUT, FULL, JOINEMPTY etc from ${QUEUESTATUS} means? |
| 12:44.12 | [TK]D-Fender | Milos: Those values sort of speak for themselves... |
| 12:45.31 | Milos | well I'm getting |
| 12:45.33 | Milos | [Sep 24 00:45:07] WARNING[6702][C-00000005]: app_queue.c:7049 queue_exec: Unable to join queue 'default' |
| 12:45.55 | Milos | I have exten => ....,n,GoToIf($[${QUEUESTATUS} = "JOINEMPTY" | ${QUEUESTATUS} = "LEAVEEMPTY"]?sorry:end) |
| 12:46.20 | Milos | I'm assuming (not really speaking for itself) that when it tries to join the queue and nobody is available that should be JOINEMPTY return status |
| 12:46.26 | wdoekes | Milos: if you use quotes in the rhs, use them on the lhs too |
| 12:46.47 | Milos | ok, is that why it's failing though? |
| 12:46.53 | wdoekes | it's expanded as a string before the comparison is made |
| 12:47.14 | kaldemar | quote is a character just like any other. |
| 12:47.46 | wdoekes | "${x}" = "123" becomes "123" = "123", and then they are compared char by char |
| 12:48.05 | Milos | right |
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| 12:48.27 | [TK]D-Fender | Milos: Quotes are literal... |
| 12:48.42 | Milos | neither of that is helpful for the issue at hand |
| 12:49.06 | wdoekes | in that case you shouldn't have pasted problem #2 simultaneously ;) |
| 12:49.07 | Milos | unable to join queue default -> what is the return value |
| 12:49.12 | [TK]D-Fender | Milos: ${QUEUESTATUS} = "JOINEMPTY" <- this will never be true |
| 12:49.23 | Milos | I understand, I have changed it |
| 12:49.26 | [TK]D-Fender | Milos: You should be DISPLAYING the variable s and seeing what it's saying |
| 12:49.56 | *** join/#asterisk danjenkins (~danjenkin@213.106.234.250) |
| 12:50.01 | kaldemar | Milos: add a Noop(${QUEUESTATUS}) after the Queue app. then you'll see what it has. |
| 12:50.05 | [TK]D-Fender | Milos: Rather than asking why your GotoIF isn't catching the right reason.. you shold actually display the vlue so you know why |
| 12:50.17 | Milos | right, that's a pretty good suggestion |
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| 12:53.01 | Milos | ah right, it was the quotes thing I just forgot to reload the dialplan after I fixed it |
| 12:53.09 | Milos | sorry about that - thanks for the advice |
| 12:53.37 | Milos | so quotes vs no quotes, any preference? |
| 12:53.54 | Milos | when there is a space I should use quotes, otherwise it's ok not to? for consistency would be best to always use quotes? |
| 12:54.15 | [TK]D-Fender | Milos: Preferrable, because if the value is actually blank it'll error out |
| 12:54.26 | Milos | ah, syntax error or something? |
| 12:54.59 | [TK]D-Fender | Milos: Yes. $[ ="HELLO"] = FAIL |
| 12:55.07 | Milos | yup |
| 12:58.01 | *** join/#asterisk ocholetras (~alorenzo@taburiente.ccti.ull.es) |
| 12:58.06 | ocholetras | hi all! |
| 12:58.12 | phix | werd up ocholetras |
| 12:58.21 | ocholetras | im going to fire! |
| 12:58.26 | ocholetras | i have g723 installed |
| 12:58.36 | ocholetras | its showed up on: core show codecs |
| 12:58.45 | ocholetras | but no translation availables on "core show translations" |
| 12:59.19 | ocholetras | on menuselect cant see a translations section, only codecs and g723 is installed :/ |
| 12:59.50 | [TK]D-Fender | ocholetras: Asterisk has no native G.723 transcoding support aside from the hardware decoder cards out there |
| 13:00.13 | ocholetras | aha |
| 13:00.16 | ocholetras | ok |
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| 13:00.20 | ocholetras | so i need a card.. |
| 13:00.28 | ocholetras | old server has a analog one |
| 13:00.38 | ocholetras | may be thats the way it decodes |
| 13:00.43 | ocholetras | now im using SIP gateways |
| 13:00.47 | ocholetras | to talk with PSTN |
| 13:00.51 | [TK]D-Fender | ocholetras: Or need to ensure you are running in pass-through mode only. Or need to stop using it. |
| 13:01.13 | [TK]D-Fender | ocholetras: Pick another codec |
| 13:03.10 | ocholetras | thanks! |
| 13:03.13 | ocholetras | ill do that! D |
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| 13:08.38 | dose_ | Hey guys anyone now how to hide caller id on OOH323 even if I put in my dialplan Set(CALLERID(num-pres)=prohib) callerid is sent ?? |
| 13:09.37 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:10.14 | WIMPy | dose_: That's the way it works on most channels. There's just a flag that says it must not be displayed. |
| 13:11.03 | dose_ | yes normally but it doesn't looks to works I have tried all posibility with command Set(CALLERID(num-pres)=XXXX) but no one works |
| 13:11.33 | WIMPy | So the bit isn't set? |
| 13:11.48 | dose_ | yes it doesn't change anything |
| 13:12.01 | Greenlight | Then isn't the other side to blame ? |
| 13:12.02 | WIMPy | Did you check the packets or just ome terminal? |
| 13:12.07 | WIMPy | some |
| 13:12.11 | dose_ | I made after it change into dialplan traces with tcpdump and Presentation and Screening stay the same |
| 13:12.41 | WIMPy | Smells like a missing feature then. |
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| 13:13.00 | dose_ | it looks that's the case |
| 13:13.09 | WIMPy | I think there has never gone much effort into H.323. |
| 13:13.29 | dose_ | I don't see another way to change the CALLERID by changing it into dialplan |
| 13:15.00 | WIMPy | May the source be with you. |
| 13:24.51 | [TK]D-Fender | dose_: The normal value is CALLERID(num), not num-pres |
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| 13:34.18 | phix | I am getting echo still, even after fxotune |
| 13:36.27 | WIMPy | Setting num-pres is the right way to do it. |
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| 13:38.49 | phix | I am using oslec with echocancel=yes |
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| 13:49.00 | dose_ | yes normally num-pres is the right way now |
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| 13:58.16 | phix | soooo, any other suggestions ChannelZ ? |
| 14:00.32 | [TK]D-Fender | phix: Already tried all of the native DAHDI EC's as well? |
| 14:00.48 | phix | What does that mean? |
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| 14:04.26 | *** join/#asterisk serafie1 (~erin@nat/digium/x-qnijilucakiaeosp) |
| 14:05.13 | [TK]D-Fender | phix: What else could it mean? There are MULTIPLE DAHDI SWEC algorithms. Have you gone through all of those in addition to OSLEC? |
| 14:06.33 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
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| 14:10.02 | phix | nope |
| 14:10.13 | phix | Thought oslec was the best one to use |
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| 14:12.43 | phix | [TK]D-Fender: What other ones do you recommend? |
| 14:12.50 | MLNoah | i'm trying to set up asterisk 11.5.1 on a test server, it's starting if i run asterisk -vvvc as root, but when i try to run service asterisk start (asterisk.conf is configured to use asteriskpbx:asteriskpbx, which has ownership of the 4 /var/*/asterisk directories), asterisk core dumps with an address out of bounds error in load_modules |
| 14:13.37 | MLNoah | http://pastebin.com/h3YKRpgk |
| 14:13.45 | [TK]D-Fender | phix: Whatever works. |
| 14:13.59 | MLNoah | any suggestions on where i should start digging for which module is defeating me? |
| 14:14.47 | phix | [TK]D-Fender: awesome thnx |
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| 14:30.27 | oryn | hey guys, I'm trying to setup an iax2 trunk bewteen 2 servers, but keep getting the dreaded "no authority found" any clues? |
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| 14:34.43 | MLNoah | is there a specific module required to run asterisk as a service? |
| 14:35.43 | mjordan | MLNoah: looks like libneon is misbehaving. Don't load res_calendar_caldav. |
| 14:38.51 | MLNoah | thanks |
| 14:39.12 | MLNoah | i think i'm just going to never ever try to restore backup config files onto a new server *eyeroll* |
| 14:40.51 | *** join/#asterisk CIDname (~CIDname@c-76-30-153-192.hsd1.tx.comcast.net) |
| 14:40.55 | CIDname | hi all |
| 14:41.14 | MLNoah | ok, now i have it to where it's creating /var/run/asterisk/asterisk.pid but not asterisk.ctl |
| 14:41.17 | phix | MLNoah: as a service? as in using a super server? |
| 14:41.33 | MLNoah | trying to get "service asterisk start" to work (CentOS 6.4) |
| 14:41.53 | MLNoah | which i've done a half dozen times on servers i hand-configured start to finish. but something in one of my copied over config files must be misbehaving |
| 14:41.56 | phix | ah |
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| 14:42.59 | Chainsaw | So, DAHDI is pulling the KERNEL_VERSION from /usr/src/linux/include/linux/version.h instead of from /usr/include/linux/version.h; instead of 199424, I'm getting 198144. And all the resulting compile failures that this entails, because it fails to exclude the pre-3.10 code paths. |
| 14:43.34 | Chainsaw | Do I need to specify an override on the include paths somehow? |
| 14:43.54 | newtonr | oryn, http://www.voip-info.org/wiki/view/Asterisk+No+authority+found |
| 14:44.41 | MLNoah | Chainsaw: have you updated the kernel-devel package (and possibly the kernel package) without rebooting? at least on CentOS, updated kernels don't take effect until reboot, and dahdi won't compile when you have the new kernel-devel installed but the old kernel running |
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| 14:45.07 | Chainsaw | MLNoah: No. The kernel sources are equivalent to the running kernel. |
| 14:45.13 | Chainsaw | Linux adrastea 3.11.0-10130-gd8efd82 #1 SMP PREEMPT Mon Sep 16 10:38:15 BST 2013 x86_64 Intel(R) Core(TM) i7-2620M CPU @ 2.70GHz GenuineIntel GNU/Linux |
| 14:45.34 | Chainsaw | MLNoah: 198144 is a 3.7 kernel version, and that would have broken ever since UAPI came in. |
| 14:46.11 | MLNoah | well, you're beyond my linux-fu then :) |
| 14:47.05 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
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| 14:49.06 | *** join/#asterisk rkrahe (~rkrahe@179.219.230.7) |
| 14:49.30 | pensmit | Anyone know of some tricks that can be done to improve communications with folks connecting form satellite based internet. I already increased the speed of light, I'm just looking for config tricks now. |
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| 14:49.59 | file | improve in what way? |
| 14:50.21 | Greenlight | Lower orbit of satallite? |
| 14:50.39 | Greenlight | Geosync is overrated anyway |
| 14:52.39 | [TK]D-Fender | pentYou need res_ftl so that your packets arrive BEFORE they are sent... |
| 14:52.39 | *** join/#asterisk CIDname (~CIDname@c-76-30-153-192.hsd1.tx.comcast.net) |
| 14:52.45 | [TK]D-Fender | pensmit: You need res_ftl so that your packets arrive BEFORE they are sent... |
| 14:54.59 | rkrahe | Hi, I've setup asterisk on my test server and I created an AGI to search the caller number on my CRM database (a variation of sugarlookup agi), this script gets the caller name and account and send back to asterisk, it also sets a variable with possible values 0, 1, 2, 3 where 0 is for my suppliers, 1 my customers, 2 my employees and 3 for unknown callers. I would like to route this callers according to this variable to the suppliers_IVR, customers_IVR, HR |
| 14:55.00 | rkrahe | unknown_callers_IVR respectively, does anyone knows how to do that? |
| 14:56.13 | [TK]D-Fender | rkrahe: what is this "suppliers_IVR" you're referring to? |
| 14:57.21 | rkrahe | I created different IVR's each one with options and destinations allowed according to the caller type |
| 14:58.01 | [TK]D-Fender | rkrahe: then set the exit context, exten, and priority and end your AGI |
| 14:58.16 | rkrahe | the only caller type that doesn't need an ivr would be my employees that would be redirect to HR extension. |
| 14:59.19 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 14:59.26 | rkrahe | D-Fender, what do you mean by set the exit context, exten and priority? I haven't heard about that before. |
| 14:59.27 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:01.04 | [TK]D-Fender | rkrahe: "SET CONTEXT", etc |
| 15:01.11 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
| 15:01.47 | rkrahe | I call this script from any inbound route using 'Call flow' and I understood at the end of this script the call would go back to the next line on the call flow |
| 15:02.24 | pensmit | lol@D-Fender--I totally MUST have this res_ftl |
| 15:02.39 | rkrahe | D-fender, thank very much mate, I will have a look on the documentation about that "set context". |
| 15:03.02 | pensmit | I read somewhere that you could use a jitter buffer? I just read this in passing and I'm not that bright so maybe I got confused. |
| 15:03.19 | [TK]D-Fender | rkrahe: Whatever this "inbound route" and "call flow" things you're referring to is... ok.... |
| 15:03.24 | Katty | looks in |
| 15:03.52 | [TK]D-Fender | pensmit: FYI, G.729 and iLBC both back better voice quality compensation for jitterly/lossy links |
| 15:04.45 | pensmit | Thanks Big D-Fend |
| 15:05.09 | Katty | lifts an eyebrow |
| 15:05.19 | Katty | [TK]D-Fender: i had /no/ idea. |
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| 15:26.13 | carrar | asterisk should implement codec2 |
| 15:26.14 | carrar | heh |
| 15:26.18 | carrar | opensource |
| 15:26.51 | file | ...what? |
| 15:26.58 | carrar | used in ham radio |
| 15:27.00 | carrar | http://www.rowetel.com/blog/?page_id=452 |
| 15:27.25 | file | there is nothing to say it can't |
| 15:31.55 | carrar | Looks like there already a asterisk model, but I don't think it's in ships with asterisk |
| 15:32.51 | file | it does not |
| 15:32.58 | carrar | seems only nature to put a opensource codec in asterisk |
| 15:33.02 | carrar | natural |
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| 15:54.45 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
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| 16:04.16 | *** join/#asterisk bobbyz (~bobbyz@c-67-161-141-149.hsd1.co.comcast.net) |
| 16:07.06 | bobbyz | I switched a client from meetme to confbridge and they are complaining that the join/leave sounds are too soft. Anyone know how to either extract the old meetme join/leave sounds (which are apparently not stored as separate files?), or can recommend some decent replacements that are permissively licensed? |
| 16:07.23 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:07.24 | *** join/#asterisk danjenkins (~danjenkin@213.106.234.250) |
| 16:13.55 | mjordan | bobbyz: MeetMe uses conf-userwilljoin/conf-userswilljoin. If you want to use those sounds, simply configure ConfBridge to use them. |
| 16:17.42 | bobbyz | mjordan: Thanks. I think they're actually complaining about the join/leave beep/chime noises. Apparently those chime sounds were hard-coded in header files for Meetme. I think I found some info on how to extract those from the source, though |
| 16:18.36 | Greenlight | Users find the oddest things to comlain about ^^ |
| 16:22.27 | bobbyz | no joke |
| 16:22.29 | bobbyz | :) |
| 16:23.21 | *** join/#asterisk danjenkins (~danjenkin@213.106.234.250) |
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| 16:27.10 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 16:29.16 | [404] | I was wondering if anyone has ever went from asterisk to an esi system? I find a lot of tutorials that go from ESI to sip trunk but not sip to esi. |
| 16:29.16 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:29.52 | navaismo | NOT FOUND |
| 16:30.37 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 16:34.47 | crumb | is the g729 implementation on asterisk.hosting.lv A or B annex? |
| 16:34.52 | [TK]D-Fender | [404]: Whatever it takes to go one way .. should go the other way |
| 16:34.54 | crumb | or just g729? |
| 16:35.28 | crumb | [TK]D-Fender: i got the codec working by the way :) |
| 16:35.30 | crumb | i had to use another toolchain |
| 16:45.22 | *** join/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 16:45.34 | volga629 | Hello Everyone, what is mean X-Asterisk-HangupCause: Bearer capability not available ? |
| 16:46.02 | file | codec incompatibility |
| 16:46.25 | volga629 | ok so need check both end what in use ? |
| 16:48.08 | *** join/#asterisk imcdona (~Thunderbi@gateway.defiancedc.com) |
| 16:49.16 | [TK]D-Fender | volga629: you should be looking at more than just that one line in yoru debug. |
| 16:49.21 | [TK]D-Fender | </tunnelvision> |
| 16:51.33 | volga629 | this debug call https://fpaste.networklab.ca/Ezmx/ |
| 16:53.03 | *** join/#asterisk jploh (~textual@49.144.39.51) |
| 16:54.17 | *** join/#asterisk jploh (~textual@49.144.39.51) |
| 16:56.50 | [TK]D-Fender | Whatever that XML transimission is supposed to contain is not supported |
| 16:56.50 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:56.59 | [TK]D-Fender | <--- SIP read from UDP:10.140.254.75:5060 ---> SIP/2.0 488 Not Acceptable Here |
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| 17:00.28 | volga629 | ok I will correct right now |
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| 17:15.10 | volga629 | To look like it some bug in soft client on remote end I tested with different client and was working good |
| 17:15.48 | volga629 | Hardware will be better way to go |
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| 17:34.45 | Penguin | mjordan: In asterisk 1.8, I had to manually set the join and leave variables to the sound files to get confbridge to play the sounds. Is it still like that in newer branches? |
| 17:35.28 | Penguin | same => n,Set(CONFBRIDGE_JOIN_SOUND=confbridge-join); |
| 17:35.29 | Penguin | same => n,Set(CONFBRIDGE_LEAVE_SOUND=confbridge-leave); |
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| 17:37.54 | mjordan | Penguin: Asterisk 1.8? Or Asterisk 10? |
| 17:38.07 | Penguin | I never used 10. |
| 17:38.12 | mjordan | Penguin: anything you did with ConfBridge in Asterisk 1.8 doesn't translate to 10+ |
| 17:38.35 | mjordan | It *may* carry over, but more than likely it doesn't. ConfBridge was pretty much re-written in Asterisk 10 |
| 17:38.53 | Penguin | I was sure of that, but I'm asking do I still have to manually set the variables in order to make the sounds play. |
| 17:39.05 | *** part/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 17:39.08 | mjordan | For the join/leave sounds, you can configure them in a bridge profile, or you can configure them via the dialplan by modifying the bridge profile using the CONFBRIDGE function |
| 17:39.13 | Penguin | In 1.8, if the sounds were not set using those variables, joining and leaving were silent. |
| 17:40.32 | mjordan | The variables no longer exist. You use the CONFBRIDGE function now - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONFBRIDGE |
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| 17:54.17 | Penguin | I guess that's pretty good. One day, I'll probably switch to 11. |
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| 18:18.06 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 18:19.30 | MLNoah | When I was reading the documentation for Asterisk Realtime, I saw that extensions.conf Realtime isn't really recommended. Is there a better/more supported way to dynamically create hint contexts? I have a multi-tenant system where if the customer has hints, the phones will be subscribing for ext@<custname>-hints, and Asterisk needs to make an ODBC call based on ext@<custname> to find the... |
| 18:19.31 | MLNoah | ...hint in my config database |
| 18:20.00 | MLNoah | i'm using Asterisk 11 |
| 18:21.32 | [TK]D-Fender | DB trigger -> config generator > reload |
| 18:23.24 | MLNoah | sometimes the simplest solutions are the hardest to find |
| 18:23.26 | MLNoah | thanks |
| 18:44.18 | *** join/#asterisk Grady2000 (~IceChat77@96.53.20.190) |
| 18:44.39 | Grady2000 | heya all, I'm using asterisk 1.6.2 and am getting errors on my previously supported 1.4 command: |
| 18:45.13 | Grady2000 | EXEC Set(CALLERID(all)="TEST <4442223333>") |
| 18:45.19 | Grady2000 | does anyone konw the 1.6 equivalent? |
| 18:45.40 | Grady2000 | and yes I am searching asterisk.org and trying different variations with no luck |
| 18:46.23 | [TK]D-Fender | That appears to be AGI... |
| 18:47.22 | Grady2000 | yes I am using a python agi script |
| 18:48.05 | [TK]D-Fender | I dont recall it ever being legal to put brackets after the application name |
| 19:02.06 | *** join/#asterisk apb1963 (~apb1963@174.134.98.138) |
| 19:06.45 | apb1963 | Good morning... Google voice is once again being lame. DTMF is sending double digits, which I have verified by using this test service: http://www.testcall.com/222-1111.html .... |
| 19:06.57 | apb1963 | Anyone know of a similar test site for SIP I can use where I can eliminate GV from the equation? |
| 19:07.06 | tonyclewis | not that I spam here ever but some cool news for the FreePBX world for anyone who has been around the project for years. http://www.freepbx.org/news/2013-09-23/were-getting-the-band-back-together |
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| 19:33.56 | malcolmd | tonyclewis: +1 |
| 19:34.33 | [TK]D-Fender | tonyclewis: /0 |
| 19:34.57 | [TK]D-Fender | watch tonyclewis segfault... |
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| 20:00.16 | highvoltz | Any way to link an Asterisk box with a ShoreTel server? Was looking to set up iax2 or equivalent so we can transfer calls between systems. |
| 20:00.26 | highvoltz | I've never used ShoreTel... |
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| 20:04.54 | [TK]D-Fender | highvoltz: ShoreTel supports SIP. Go for it. |
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| 20:25.33 | karl-s | Shoretel barely supports sip |
| 20:26.06 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.60) |
| 20:26.08 | karl-s | When you drill them on it, they tell you they support their own variation of SIP called ShoreSIP |
| 20:27.59 | highvoltz | that's the best route though? |
| 20:29.04 | [TK]D-Fender | Best route : Burn ShoreTel. |
| 20:29.06 | [TK]D-Fender | WITH FIRE!!! |
| 20:29.56 | jmetro | burn it with water! |
| 20:29.59 | jmetro | wait.. |
| 20:31.13 | [TK]D-Fender | jmetro: </fracking> |
| 20:32.18 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 20:33.14 | [TK]D-Fender | heads home |
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| 21:04.20 | karl-s | highvoltz, best route is to doa SIP Trunk on shoretel provided the vendor has allocated ample switch resources |
| 21:04.42 | karl-s | you are going to end up with a few warnings on Asterisk and it will take some finagling |
| 21:05.00 | karl-s | usually the fix for me has been to reboot the shoregear switch when sip stops working |
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| 22:37.16 | Grady2000 | does anyone have asterisk 1.6 set caller id functions working in agi scripts? |
| 22:37.24 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 22:37.57 | Grady2000 | I'm trying every variation of EXEC SetCallerid and it all worked in 1.4 , but not 1.6 |
| 22:39.22 | karl-s | Grady2000, i think it changed |
| 22:39.22 | [TK]D-Fender | <PROTECTED> |
| 22:39.31 | [TK]D-Fender | SetCallerID died a LONG time ago |
| 22:39.48 | [TK]D-Fender | Grady2000: And I already pointed out your syntax error several hours ago |
| 23:02.05 | Grady2000 | I appreciate the help D-fender, I misunderstood when you said remove the brackets |
| 23:02.44 | Grady2000 | so I can try: EXEC CALLERID()? with proper syntax? |
| 23:04.38 | [TK]D-Fender | Grady2000: No, you kep calling SET with BRACKETS after it |
| 23:04.51 | [TK]D-Fender | Grady2000: EXEC APP DATA |
| 23:04.59 | [TK]D-Fender | no brackets between APP & DATA |
| 23:08.48 | *** part/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 23:13.03 | Grady2000 | could you give me a syntax example? |
| 23:14.44 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+AGICommand_exec |
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