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| 01:45.09 | arcticblue | i have a question about updating a review in gerrit. when i run 'git review' again, it warns me that I'm about to submit multiple commits. Is this expected or did I screw something up? |
| 01:47.54 | arcticblue | the review in question - https://gerrit.asterisk.org/#/c/977/ I've made the requested changes and want to update, but the warning about submitting multiple commits has me hesitating. It looks like it wants to submit my original commit and the new one. |
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| 02:15.30 | arcticblue | well, maybe i should read a little closer. "This is expected if you are |
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| 02:15.32 | arcticblue | submitting a commit that is dependent on one or more in-review |
| 02:15.34 | arcticblue | commits |
| 02:15.53 | arcticblue | i'll just say "yes" and see what happens :P |
| 02:16.40 | arcticblue | ah, rejected. got to squash commits. the asterisk contributing docs seem to be a little lacking when it comes to updating a review. |
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| 02:17.22 | arcticblue | asterisk wiki says run "git rebase -i" then "git review". this doesn't work. |
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| 06:11.32 | tulga | my provider said our digium card sync out. so how to sync again? |
| 06:12.11 | tulga | sync/clock source is 0 |
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| 07:15.33 | gustopn | hi |
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| 12:08.38 | Rico | hi |
| 12:09.20 | Rico | with pjsip, is there a way to have the SIP trunk informations in pjsip.conf and informations about the endpoints (sip phones) in realtime database ? |
| 12:10.22 | file | https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
| 12:13.29 | Rico | ok got it, thanks |
| 12:13.54 | luisgrin | hi, long time i not by |
| 12:14.07 | luisgrin | but now i need to build an outbound ivr |
| 12:14.28 | luisgrin | posibly the simplest |
| 12:14.52 | luisgrin | some advice? I have 3 voip lines and ubuntu 13.04 |
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| 12:21.56 | [TK]D-Fender | luisgrin: No such thing as "outbound IVR". |
| 12:22.10 | [TK]D-Fender | IVR is a thing you can do with a channel you HAVE though. |
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| 12:25.11 | Rico | can't find a way to make an incoming call match an endpoint... |
| 12:25.23 | WIMPy | No asdvice without input. |
| 12:26.02 | Rico | WIMPy: for me ? |
| 12:26.16 | WIMPy | No, for luisgrin |
| 12:26.28 | Rico | ok |
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| 12:29.27 | luisgrin | hi |
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| 12:31.09 | luisgrin | no advice without advice? |
| 12:34.18 | [TK]D-Fender | luisgrin: rephrase your question |
| 12:34.35 | Rico | about my pjsip problem, if somebody can take a look at that... |
| 12:34.36 | Rico | http://pastebin.com/uATtbNv0 |
| 12:34.55 | Rico | when I do an incoming call, it's rejected with 488 |
| 12:35.10 | luisgrin | ok, I need advice in using asterisk to build a simple IVR, mainly for outbound calls, I have 3 voip lines |
| 12:35.14 | Rico | I guess it is because no endpoint is found |
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| 12:35.36 | Rico | but I don't understand how to solve that |
| 12:36.38 | Rico | following the wiki (https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide), I don't have the endpoitn warning in the CLI |
| 12:36.51 | [TK]D-Fender | luisgrin: ivr = context with "core show application waitexten". |
| 12:36.52 | Rico | if I don't set debog on, I have nothing in CLI |
| 12:37.13 | [TK]D-Fender | luisgrin: And clarify about the origin of these outbound calls. |
| 12:37.29 | [TK]D-Fender | luisgrin: You are putting vague words together to form this idea. |
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| 12:45.08 | file | Rico, you would be incorrect - it's matching an endpoint fine, 488 is incompatible codec |
| 12:45.35 | luisgrin | ok |
| 12:45.50 | luisgrin | ok, I need advice in using asterisk to build a simple PBX mainly for outbound calls, I have 3 voip lines |
| 12:47.52 | mjordan | "res_pjsip_sdp_rtp.c:785 negotiate_incoming_sdp_stream: Endpoint has no codecs for media type 'audio', declining stream" |
| 12:48.47 | Rico | ok thanks file |
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| 12:50.11 | Rico | file, mjordan : thanks a lot, incoming call works |
| 12:55.05 | mjordan | luisgrin: have you looked at the wiki or A:TDG? |
| 12:57.07 | luisgrin | hi, no i have not, thanks, i will |
| 12:58.36 | [TK]D-Fender | ~book |
| 12:58.37 | 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 |
| 12:58.39 | [TK]D-Fender | ~asteriskwiki |
| 12:58.40 | infobot | i guess asteriskwiki is http://wiki.asterisk.org |
| 12:59.22 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:03.27 | [TK]D-Fender | I recall there being at least two H.323 channel drivers. I've got a 1.8 system I'm testing on and has chan_ooh323.so" and I see h323.conf but somehow didn't feel like it was the matching config. Am I simply mistaken? |
| 13:04.04 | file | chan_ooh323 uses ooh323.conf |
| 13:04.08 | file | chan_h323 uses h323.conf |
| 13:04.53 | [TK]D-Fender | Was thinging that, .... |
| 13:04.58 | [TK]D-Fender | I skipped right over it... |
| 13:05.12 | [TK]D-Fender | not sufficiently caffeinated yet |
| 13:05.23 | [TK]D-Fender | All cool. thanks. |
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| 13:20.32 | GeForce111 | Good morning. I need some advice on connecting a remote user to our Asterisk PBX from him home. We have the firewall set to filter by IP to protect against anyone from connecting but since our home users don't have static IPs, we're looking for another way where the user can simply plug in the phone and have it work. Minus some steps in port forwarding on their home firewalls. |
| 13:21.55 | *** part/#asterisk esaym153 (~esaym153@75-1-182-8.lightspeed.snantx.sbcglobal.net) |
| 13:25.06 | [TK]D-Fender | Their side should never need forwarding |
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| 13:27.01 | GeForce111 | @ TKDFender, That's true. I was referring to if we had to, that would be fine. Oh and to add to the mix, we're using Polycom Soundpoint IP 335 phones with 4.08 firmware |
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| 13:27.42 | [TK]D-Fender | If you are behnd NAT then you have a number of settings to account for on your side |
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| 13:28.54 | GeForce111 | We're not behind a NAT. We're using the Firewall included with the CentOS/Asterisk. The PBX IP is open and is strickly used only for PBX traffic. |
| 13:30.33 | [TK]D-Fender | then you should be fine. |
| 13:31.10 | GeForce111 | We have the firewall turned on and were not able to connect any phone from remote. Then we turned on IP filtering and tested using the current IP of a home user. But we know home internet IPs change and when it does, the phone will lose connection unless we change the IP. That won't fly with the Executives. |
| 13:33.33 | [TK]D-Fender | So you need some way to track them. |
| 13:33.40 | [TK]D-Fender | or change your firewall strategy |
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| 13:33.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:33.54 | [TK]D-Fender | fail2ban is the usual solution |
| 13:35.16 | GeForce111 | What other firewall strategy do we have? Mac address filtering only works for us for internal, should it work for external as well? |
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| 13:38.14 | [TK]D-Fender | 1 idea : use a MicroBrowser script to hit a web page that auto-open's that IP for SIP. |
| 13:38.37 | [TK]D-Fender | And shove it on a 1 min refresh |
| 13:38.46 | [TK]D-Fender | (as the idle page) |
| 13:39.02 | [TK]D-Fender | or track attackers with fail2ban and just keep after them |
| 13:41.49 | GeForce111 | What about this? Do we know if the Asterisk/CentOS firewall can filtering by DNS, so if I got home users with a DNS name (example dyndns) that auto updates their home IP, we can filter by just the DNS name? |
| 13:44.16 | johnny_|_ | And what is the advantage of keeping track and updating DNS lookup rather than just track IP?? |
| 13:44.49 | [TK]D-Fender | You could do that too |
| 13:45.11 | [TK]D-Fender | mind you I don't think it will resolve constantly like that |
| 13:45.20 | [TK]D-Fender | usually when you add a rule it'll resolve it ONCE and that's it |
| 13:45.25 | [TK]D-Fender | You'd have to script a refresh on it |
| 13:45.34 | [TK]D-Fender | Not sure on how that can be done |
| 13:45.49 | [TK]D-Fender | BRB |
| 13:45.54 | johnny_|_ | It can be done but it makes no sense. |
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| 14:17.24 | ak77 | hello all |
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| 14:18.10 | cyford33 | hi, i reinstall asterisk 11.16 and not i dont have any sip commands |
| 14:21.26 | ak77 | in my system I have two channels configured for dialout... technology/resource: PJSIP/endpoint1 and Dongle/endpoint2 ... while I can make calls with Dongle/endpoint2/NUMBER and PJSIP/endpoint1/NUMBER@HOST, how can I determine suffix needed (@HOST) using ARI? or. how can i configure PJSIP that it won't need that suffix |
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| 14:28.40 | [TK]D-Fender | cyford33: go verify you have the module present, then reload it and watch what happens. |
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| 14:38.01 | cyford33 | module show = 0 modules loaded lol |
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| 14:48.32 | [TK]D-Fender | SUCKcess |
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| 15:36.50 | nunopereira | hi, is it possible to get the local ip (sent in the from: header) in the dialplan? |
| 15:37.28 | nunopereira | my objective is to set a P-Asserted-Identity: header |
| 15:38.54 | [TK]D-Fender | "core show functions like SIP" |
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| 15:42.20 | nunopereira | [TK]D-Fender, in that case, what sippeer do I use |
| 15:43.07 | [TK]D-Fender | non. |
| 15:43.09 | [TK]D-Fender | none. |
| 15:43.15 | [TK]D-Fender | wrong function. |
| 15:43.29 | nunopereira | sorry, for SIPPEER() function |
| 15:43.35 | [TK]D-Fender | wrong function. |
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| 15:49.05 | nunopereira | not SIPPEER()? |
| 15:52.59 | [TK]D-Fender | wrong function. <--------- |
| 15:55.50 | nunopereira | I'm using sip, not pjsip, and so that's the only one that I see available |
| 15:56.07 | nunopereira | is one of CHECKSIPDOMAIN or SIP_HEADER? |
| 15:56.19 | [TK]D-Fender | pastebin the complete output of the command I gave you |
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| 15:58.23 | nunopereira | in http://pastebin.com/g5kmXy0L |
| 16:00.02 | cyford33 | Unable to load module chan_sip.so |
| 16:00.02 | cyford33 | Command 'module load chan_sip.so' failed |
| 16:01.04 | WIMPy | Why do you have pjsip loaded if you don't use it? |
| 16:02.05 | nunopereira | didn't configured the server |
| 16:03.15 | [TK]D-Fender | SIP_HEADER SIP_HEADER(name[,number]) Gets the specified SIP header from an incoming INVITE message. |
| 16:03.23 | [TK]D-Fender | nunopereirahi, is it possible to get the local ip (sent in the from: header) in the dialplan? |
| 16:03.44 | [TK]D-Fender | <PROTECTED> |
| 16:04.02 | nunopereira | I'm using ARI and setting in the dialplan is not the preferred option |
| 16:04.10 | nunopereira | but I don't want the full header |
| 16:04.26 | [TK]D-Fender | It's text. CHOP IT UP |
| 16:04.39 | nunopereira | just the domain/ip part |
| 16:06.04 | *** join/#asterisk ryang (sid10904@gateway/web/irccloud.com/x-rydnjoddnoavbprw) |
| 16:06.31 | file | hmmm? |
| 16:06.57 | nunopereira | "Gets the specified SIP header from an incoming INVITE message." |
| 16:07.09 | nunopereira | I'm making an outgoing call, not inbound |
| 16:08.47 | timholum | <PROTECTED> |
| 16:09.14 | [TK]D-Fender | nunopereiramy objective is to set a P-Asserted-Identity: header <- then you're too late |
| 16:09.35 | timholum | but thee phone 2009 is not on the phone on eather server |
| 16:09.42 | nunopereira | [TK]D-Fender why too late? |
| 16:09.44 | [TK]D-Fender | nunopereira: You don't have a header till a call is actually happening. If the call is already ahppening.. then the packets are flying and you can't just add another header |
| 16:10.25 | nunopereira | [TK]D-Fender but I was successful in adding the header, before the Dial() call |
| 16:11.13 | [TK]D-Fender | timholum: And that queue is clearly calling a LOCAL CHANNEL. So you should be paying attention to what that's actually doing in the dialplan because there is no direct relationship to something like a SIP device |
| 16:11.24 | *** join/#asterisk LiuYan (~hola@unaffiliated/liuyan) |
| 16:11.28 | WIMPy | And the PAI that would be send from the CALLERID isn't good enough? |
| 16:11.29 | [TK]D-Fender | nunopereira: You don't have a FROM YET if this is before you call |
| 16:12.52 | nunopereira | so it's not possible to get the domain that is used in the call, BEFORE establishing the call? |
| 16:12.52 | timholum | is there a way to tie that to a SIP channel? |
| 16:13.04 | *** join/#asterisk qloogkm (sid20721@gateway/web/irccloud.com/x-kxzysxyywdjmyvtz) |
| 16:13.20 | [TK]D-Fender | nunopereira: You can't get soething FROM a call BEFORE the call. |
| 16:13.28 | nunopereira | WIMPy PAI? |
| 16:13.37 | [TK]D-Fender | nunopereira: NOT APPLICABLE |
| 16:14.02 | [TK]D-Fender | nunopereira: This is a 4th dimensional problem. |
| 16:14.13 | nunopereira | [TK]D-Fender: seems so |
| 16:14.16 | WIMPy | PAI = P-Asserted-Identity |
| 16:14.31 | [TK]D-Fender | WIMPy: He wants to know what the "From:' WILL BEL on an outgoing call he's looking to place. |
| 16:14.59 | [TK]D-Fender | WIMPy: that's the key |
| 16:15.06 | nunopereira | WIMPy: I'm trying to build it, and need to set the @host part of it (sip: format) |
| 16:15.09 | WIMPy | Does he? I read the he wants to build a PAI header. |
| 16:15.27 | [TK]D-Fender | WIMPy: yes, based on a value that does not EXIST |
| 16:15.33 | WIMPy | So what's wrong with the one that would be generated from te CALLERID? |
| 16:15.44 | [TK]D-Fender | WIMPy: He needs to base that on the FROM: Header for his call.. which does not EXIST yet |
| 16:15.58 | nunopereira | WIMPy: not usable in the remote peer |
| 16:16.09 | nunopereira | [TK]D-Fender: not necessarily |
| 16:16.19 | [TK]D-Fender | nunopereira: Yes, very necessarily |
| 16:16.23 | nunopereira | This is an ARI app |
| 16:16.51 | nunopereira | that wants to make anonymous calls, but needs to send the PAI |
| 16:17.03 | [TK]D-Fender | that's fine |
| 16:17.19 | [TK]D-Fender | it's what you are looking to PULL the information from in order to know how to GENERATE it that is the issue |
| 16:17.22 | nunopereira | if getting the @host part from the FROM: header isn't possible |
| 16:17.23 | WIMPy | So what's wrong with the generated one? |
| 16:17.29 | nunopereira | I'm open for other options |
| 16:17.56 | [TK]D-Fender | nunopereira: What do you get if you simply let "sendrpid=pai" do its job? |
| 16:17.57 | nunopereira | WIMPy: it isn't generated |
| 16:18.47 | WIMPy | Switch it on. |
| 16:18.49 | nunopereira | [TK]D-Fender: let me check the "sendrpid=pai" option |
| 16:19.18 | WIMPy | ("sendrpid") |
| 16:22.15 | nunopereira | not good |
| 16:22.46 | nunopereira | it's sending the same as in FROM:P-Asserted-Identity: "Anonymous" <sip:anonymous@anonymous.invalid> |
| 16:22.50 | nunopereira | P-Asserted-Identity: "Anonymous" <sip:anonymous@anonymous.invalid> |
| 16:23.40 | *** join/#asterisk foamz (sid25727@gateway/web/irccloud.com/x-vfpmptvksiavrmnp) |
| 16:25.58 | *** join/#asterisk Dibbler (~Dibbler@host86-150-219-212.range86-150.btcentralplus.com) |
| 16:31.06 | WIMPy | Set your CALLERID then. |
| 16:32.23 | nunopereira | I'm already setting callerid(num-pres) with prohib for anonymous calls |
| 16:33.18 | WIMPy | Yes, that's what you get. |
| 16:34.05 | nunopereira | but that is interpreted by the remote peer as anonymous, as both PAI and From: headers are anonymous |
| 16:34.42 | *** join/#asterisk nunne (sid38499@gateway/web/irccloud.com/x-nwjmviwexnyayvho) |
| 16:34.49 | nunopereira | which is not what we want |
| 16:35.04 | WIMPy | Then why do you set it that way? |
| 16:35.31 | nunopereira | because I'm a dummy in Asterisk :S |
| 16:35.57 | WIMPy | >>Set your CALLERID then. |
| 16:36.14 | nunopereira | I'm setting |
| 16:36.19 | nunopereira | both num and num-pres |
| 16:37.10 | WIMPy | to what you want? |
| 16:38.53 | nunopereira | I'm having this: |
| 16:38.54 | nunopereira | From:"anonymous"<sip:anonymous@anonymous.invalid>;tag=nnnnn Privacy:id P-Asserted-Identity: "Anonymous" <sip:anonymous@anonymous.invalid> |
| 16:39.16 | nunopereira | and need to have this |
| 16:39.18 | nunopereira | From:"anonymous"<sip:anonymous@anonymous.invalid>;tag=nnnnn |
| 16:39.18 | nunopereira | Privacy:id |
| 16:39.18 | nunopereira | P-Asserted-Identity:"XXXXXXX"<sip:XXXXXXX@host> |
| 16:39.26 | nunopereira | where XXXXXX is the calling number |
| 16:39.28 | WIMPy | >>Set your CALLERID then. |
| 16:39.59 | WIMPy | 'core show function CALLERID' if yu didn't have that idea, yet. |
| 16:40.10 | nunopereira | I said that I set it |
| 16:40.26 | WIMPy | And the wiki has a page about manipulationg caller id as well. |
| 16:40.32 | nunopereira | both CALLERID(num), and that is set inside the Stasis app |
| 16:40.35 | WIMPy | But obviousely not to what you want. |
| 16:40.49 | nunopereira | and CALLERID(num-pres) with value prohib |
| 16:41.21 | WIMPy | Any idea what prohib(ited) might mean? |
| 16:41.35 | nunopereira | I have |
| 16:41.42 | WIMPy | So? |
| 16:41.49 | tompaw | Does asterisk have any preference on number of cpu sockets? (2 vs 4) |
| 16:42.35 | nunopereira | it is what sets the number in the FROM: to anonymous |
| 16:42.44 | tompaw | i.e. if I get 4x12-cpuxHT, will asterisk be able to use all 96 logical cores? |
| 16:42.59 | WIMPy | nunopereira: As well as in the PAI. |
| 16:43.25 | nunopereira | if i set CALLERID(num-pres)=allowed_passed_screen, for example, then the number is sent in both PAI and FROM |
| 16:44.04 | WIMPy | Ok, so maybe we get to the real question now... |
| 16:44.11 | WIMPy | So you don't want it in the from? |
| 16:45.20 | nunopereira | From:"anonymous"<sip:anonymous@anonymous.invalid>;tag=nnnnn |
| 16:45.31 | nunopereira | Privacy:none |
| 16:45.41 | nunopereira | and |
| 16:45.46 | nunopereira | P-Asserted-Identity:"XXXXXXX"<sip:XXXXXXX@host> |
| 16:46.14 | WIMPy | That's the way you want it??? |
| 16:46.27 | nunopereira | WIMPy: yes |
| 16:47.04 | WIMPy | Ok, might work by (ab)using defaultuser/fromuser/fromdomain settings. |
| 16:51.06 | *** join/#asterisk Katty (sid62315@gateway/web/irccloud.com/x-drgcifcsllyhpvvu) |
| 16:54.16 | tompaw | Does asterisk prefer clock speed or number of cores? (40 x 2.5 Ghz vs 48 x 1.9 Ghz) |
| 16:55.04 | WIMPy | 40x2.5 is more than 48x1.9 anyway. |
| 16:55.52 | WIMPy | But at that numbers, kernel configuration might become interesting. |
| 16:58.14 | tompaw | Interesting as in "challenging at prone to fail unless you spend 12 hours rebuilding the thing" or interesting as in "let's see how * scales on this" |
| 16:58.17 | tompaw | ? |
| 16:58.51 | nunopereira | WIMPy: how can I do that? |
| 16:59.55 | WIMPy | Interesting as in having a few days fun trying to build with different serrings and try to measure efficiency. |
| 17:03.07 | Ice_Strike | Have any of you guys use Vagrant with Asterisk installed? |
| 17:03.59 | *** join/#asterisk ketas- (~ketas@229-211-191-90.dyn.estpak.ee) |
| 17:04.12 | *** join/#asterisk airjump (~Thunderbi@p20030070CE3251D570DEACD98728407F.dip0.t-ipconnect.de) |
| 17:04.24 | lvlinux | I need help with something: I have srvlookup set to yes in sip.conf. Callcentric requires it. When I lose internet, my PBX pretty much dies as it won't do any sip connections. Any way to prevent this? (I have a PSTN line as well, so it still needs to work with no internet access.) |
| 17:05.31 | *** join/#asterisk jonno11 (~Jon@108.61.155.148) |
| 17:07.32 | *** join/#asterisk airjump (~Thunderbi@p20030070CE3251D570DEACD98728407F.dip0.t-ipconnect.de) |
| 17:13.14 | [TK]D-Fender | lvlinux: get a caching DNS proxy |
| 17:13.47 | lvlinux | you mean so that my local phones will have their DNS still available to Asterisk? |
| 17:14.08 | lvlinux | Is there any way to do it with Asterisk config? Seems like a bug to me but maybe it's a "feature" :-) |
| 17:17.45 | WIMPy | I guess you need be be more specific. I know that VOIP "connections" sometimes don't recover after som DNS issue, but if you can't us a PSTN interface, either, you need to provide more information. |
| 17:19.02 | lvlinux | Ok well it's a Obihai device that I'm using as the PSTN connection. It connects to * as a SIP peer. I'm sure a DAHDI interface would work I guess---but I couldn't use my SIP handsets. Even dialing *97 wouldn't work. |
| 17:19.29 | *** join/#asterisk airjump (~Thunderbi@p20030070CE3251D570DEACD98728407F.dip0.t-ipconnect.de) |
| 17:19.37 | lvlinux | it looked to me like * was ignoring SIP registration attempts from handsets. |
| 17:19.59 | WIMPy | Do the hsndsets still find your Asterisk server? |
| 17:20.20 | lvlinux | what do you mean by "find"? |
| 17:20.26 | WIMPy | And, yes, cahn_sip does block on some occasions. Make sure you have dnsmgr enabled. |
| 17:20.27 | lvlinux | ping works in both directions |
| 17:20.41 | lvlinux | what will dnsmgr do? |
| 17:21.11 | WIMPy | dns cacheing |
| 17:22.03 | lvlinux | within asterisk? oh ok. |
| 17:22.15 | lvlinux | so i wouldn't really need to setup an external DNS server. |
| 17:22.29 | lvlinux | actually i already have one in the house, but it doesn't have records for all my phones and such. |
| 17:23.12 | lvlinux | I'm a bit confused though as to why * would even care about DNS for "friend" peers that register to a certain IP/user combo. |
| 17:23.57 | WIMPy | It won't. |
| 17:24.33 | WIMPy | But failing outbound registrations may prevent other communication as well, while still trying. |
| 17:24.57 | WIMPy | I don't know the exact details, but there are such issues. |
| 17:27.39 | lvlinux | ah ok. that makes sense |
| 17:28.42 | lvlinux | problem was I lost internet right at the same time as a power outage, so any caching would have been lost anyway... |
| 17:28.49 | lvlinux | does pjsip fix that issue? |
| 17:29.52 | file | chan_pjsip has asynchronous DNS, it won't block everything |
| 17:30.27 | malcolmd | yay, chan_pjsip |
| 17:30.29 | lvlinux | ok so it should continue to work with my LAN devices, even while being without DNS? |
| 17:30.46 | file | provided it doesn't have to resolve addresses to get to them, yes |
| 17:30.51 | lvlinux | I guess it's time to upgrade to v13... I'm on 11 now. |
| 17:32.36 | lvlinux | If I do turn off srvlookup, will the blocking problem go away, or will non-resolvable registration DNS addresses still cause a problem? |
| 17:32.47 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:37.52 | *** join/#asterisk TKOC (~kenneth@0x5e922d72.adsl.cybercity.dk) |
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| 17:40.31 | TKOC | hi i am looking to setup a asterisk server and have it connect to a danish ISDN30 connection but i can't finde out how tis is done |
| 17:42.46 | [TK]D-Fender | Get an E1 interface. connect it to your server. |
| 17:42.46 | *** join/#asterisk ketas- (~ketas@229-211-191-90.dyn.estpak.ee) |
| 17:42.51 | [TK]D-Fender | configure Asterisk. Done |
| 17:43.16 | tompaw | Collect Segfault traces. Repeat. |
| 17:44.12 | TKOC | [TK]D-Fender can you recon a E1 interface card pci |
| 17:44.26 | TKOC | pci-e |
| 17:44.39 | [TK]D-Fender | There are plenty from Digium, Sangoma, etc |
| 17:45.09 | TKOC | thanks i will look ind to it |
| 17:45.45 | WIMPy | TKOC: Or look for anything with an HFC-E1 chip. |
| 17:47.53 | *** part/#asterisk midsandhighs (~midsandhi@li296-126.members.linode.com) |
| 17:48.43 | WIMPy | TKOC: You should find out which driver suits your needs and then find the fitting hardware. |
| 17:51.28 | WIMPy | Oh, and PCI-e tends to cost a lot extra. Might make sense to get a board with PCI instead. |
| 17:58.40 | *** part/#asterisk MKEbrew (~MKEbrew@h69-129-178-254.nwblwi.dedicated.static.tds.net) |
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| 18:18.55 | tompaw | So, we've been using flowroute for the past 3 days and 30% of the US calls are stuttered as f*ck. |
| 18:19.24 | tompaw | Most of the affected numbers are business, I wonder if that means they're voip did and this somehow affects the quality. |
| 18:20.00 | tompaw | Anyway, can you recommend a good provider for US termination? Quality above anything else. |
| 18:21.39 | cyford33 | when u install asterisk from source does it include devel files automaticlly |
| 18:22.33 | WIMPy | What "devel files"? |
| 18:22.54 | lvlinux | tompaw: flowroute is a good provider---i would double check your connection and config before assuming that they are the problem. |
| 18:23.21 | lvlinux | cyford33: if you are referring to the xxxxx-devel packages for the libraries needed to compile Asterisk, then the answer is no. |
| 18:24.08 | lvlinux | when you install Asterisk from source, it includes (surprise!) the source of Asterisk, not other stuff. |
| 18:24.08 | tompaw | lvlinux: yeah we've been using them for ages and never had that before |
| 18:24.11 | nunopereira | tompaw: we use voicetrading, for many of the international termination |
| 18:24.33 | cyford33 | how can i install asterisk-devel by using source files |
| 18:24.41 | tompaw | by using source files |
| 18:24.59 | tompaw | asterisk-devel *is* source files |
| 18:25.20 | lvlinux | tompaw: flowroute doesn't proxy media, so your problem may be with a specific RTP termination point. |
| 18:25.21 | tompaw | header files, etc. |
| 18:25.33 | cyford33 | i am told to run this yum install asterisk-devel |
| 18:25.33 | tompaw | lvlinux: they don't? how come? |
| 18:26.29 | lvlinux | tompaw: that's the way SIP is supposed to work---it's just a connector between two endpoints. SIP says: get your audio from over there...and send it to over here... |
| 18:26.48 | tompaw | lvlinux: yes, in theory... |
| 18:27.29 | lvlinux | no, not in theory, that's exactly how it works. To proxy media you have to use software that intervenes and takes the RTP streams. |
| 18:27.39 | lvlinux | Few providers actually do that. |
| 18:28.29 | tompaw | well, my background is in wholesale platforms, where you buy and sells tens of thousands of channels. if anyone tried not to proxy audio, they'd be out of business in days, because of frauds, FAS, and all sorts of crap. |
| 18:28.31 | lvlinux | Unless the provider has a super large network, and tight control over every part of it, proxying media will only add latency and jitter, and increase their own server overhead and bandwidth usage. |
| 18:29.10 | lvlinux | what does proxying the audio have to do with fraud? |
| 18:29.30 | lvlinux | SIP is where all the accounting and connection information comes from. |
| 18:29.33 | tompaw | some providers demand billing on rtp |
| 18:30.40 | tompaw | lvlinux: don't get me wrong, I fully agree that's the way it SHOULD work, but in wholesale world it's exactly the opposite |
| 18:30.46 | tompaw | hence my surprice re: flowroute |
| 18:31.14 | lvlinux | well if that's the way it commonly works then that explains a lot of the junk call quality... |
| 18:32.49 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:33.00 | lvlinux | There's only one provider I know of that really successfully proxys RTP, and that's because they do make sure to keep a tight grip on every part of their network, use QoS, and they directly peer with most of the telcos that they terminate to. So everything works well. But you can't proxy media and do a good job with calls if you are just throwing them out over the internet when they leave your network... |
| 18:34.07 | tompaw | lvlinux: I don't disagree, I merely inform you of my experience. wholesale trading is quite different than on-premises asterisk setups. |
| 18:34.09 | cyford33 | to install espeak i am told i need its dependacies an to run yum install asterisk-devel - but it makes me download asterisk.1.8 as well which conflicts with my asterisk 11.16 . how can i get asterisk-devel depend from source |
| 18:34.30 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 18:34.31 | lvlinux | tompaw: true |
| 18:34.59 | lvlinux | cyford33: wait a minute---give some details about the system you are on---debian? |
| 18:35.18 | cyford33 | centos 6.5 |
| 18:35.19 | lvlinux | nevermind no you wouldn't have yum on debian |
| 18:35.21 | lvlinux | ok |
| 18:35.52 | lvlinux | you should be able to install espeak without asterisk-devel |
| 18:36.00 | lvlinux | i would think. |
| 18:36.17 | cyford33 | ok i installed it, but i can not load its app module in asterisk |
| 18:36.37 | lvlinux | The app module is part of Asterisk |
| 18:37.00 | cyford33 | register*CLI> module load app_espeak.so |
| 18:37.00 | cyford33 | Unable to load module app_espeak.so |
| 18:37.00 | cyford33 | Command 'module load app_espeak.so' failed. |
| 18:37.08 | lvlinux | you probably didn't compile it when you compiled Asterisk, becuase you didn't have the devel files for espeak at the time. |
| 18:37.44 | cyford33 | oh i downloaded espeak an compiled it |
| 18:37.56 | cyford33 | <PROTECTED> |
| 18:38.02 | lvlinux | is the file app_espeak.so in your modules folder? if not, that's why asterisk can't find it. |
| 18:38.28 | lvlinux | you compiled espeak before Asterisk? |
| 18:38.44 | cyford33 | after asterisk |
| 18:38.56 | cyford33 | i followed this instruction http://nerdvittles.com/?p=7448 |
| 18:39.30 | cyford33 | search for espeach |
| 18:39.33 | lvlinux | menuselect is a configure program for compiling asterisk---so if you run it now, it won't change your existing Asterisk program---you have to recompile it with the new menuselect options. |
| 18:40.18 | cyford33 | yes, i didnt look for espeak in menuselect, i was just askin if it is in there |
| 18:40.22 | cyford33 | ill check now |
| 18:40.42 | lvlinux | yes i bet it is. |
| 18:41.14 | *** join/#asterisk superscrat (asanders@nat/digium/x-klfizurykskivzjj) |
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| 18:41.24 | lvlinux | and i bet it wasn't checked when you compiled, because you didn't have the needed files for the Asterisk configure program to find it. |
| 18:42.30 | lvlinux | you may have to re-run the configure program too (probably). It may find it by itself or you may have to add "--with-espeak" or something like that to it. |
| 18:42.45 | cyford33 | i dont see any tts engines in menuselect, and i have all apps selected anyway |
| 18:43.28 | file | "app_espeak" is not distributed with Asterisk |
| 18:45.25 | lvlinux | yeah I just saw that---it's a 3rd party app. So it wouldn't be in menuselect at all. |
| 18:45.31 | lvlinux | thanks file |
| 18:46.01 | lvlinux | cyford33: did you get the espeak-asterisk thing from git? |
| 18:48.07 | cyford33 | yes, i followed these instructions http://pastebin.com/PQDxuW5i |
| 18:48.32 | *** part/#asterisk mjordan (mjordan@nat/digium/x-ekvizpkkwlorflwd) |
| 18:48.57 | cyford33 | but cant load the .so |
| 18:49.01 | lvlinux | cyford33: when you did make samples did it create the espeak.conf file in your /etc/asterisk folder as it should? |
| 18:49.24 | lvlinux | and did you check the modules directory to see if the .so file is actually there? |
| 18:50.00 | lvlinux | It could be that the Asterisk-eSpeak makefile put the files in a different place than your Asterisk installation. |
| 18:50.21 | cyford33 | yes both files are there with and asterisk has ownership |
| 18:52.33 | lvlinux | hmmm, then I'm not sure. If the module is there then it should be able to load it I would think. |
| 18:52.52 | cyford33 | yeah, all my other modules load |
| 18:56.57 | cyford33 | hm, i dont think my cli will load any module same error for everything lol |
| 18:59.17 | lvlinux | try without the .so |
| 18:59.40 | cyford33 | no |
| 18:59.48 | cyford33 | no luck |
| 19:00.15 | lvlinux | module load app_espeak works for me. |
| 19:00.19 | lvlinux | i just installed it |
| 19:00.58 | lvlinux | I'm on v13.4 |
| 19:01.02 | cyford33 | ok |
| 19:01.04 | lvlinux | on this machine |
| 19:01.40 | *** join/#asterisk theron_ (~theron@199.201.64.131) |
| 19:01.45 | cyford33 | all u did was the directions i sent you? |
| 19:01.45 | lvlinux | i just did the git clone command, make, make install, make samples, then went into my console and ran that and it worked. |
| 19:02.05 | lvlinux | are you sure it compiled correctly with no errors? |
| 19:02.30 | cyford33 | when i download the 4.0 i get errors |
| 19:02.49 | cyford33 | <PROTECTED> |
| 19:03.03 | lvlinux | i think the 4.0 is ancient... |
| 19:03.07 | lvlinux | so that makes sense |
| 19:03.17 | lvlinux | i'm running debian 7 |
| 19:03.56 | lvlinux | i had to install the dev files for libsndfile and libsamplerate and libespeak before compiling but after that it worked no probs |
| 19:04.24 | lvlinux | but if you can't load other moduesl it looks like it's something to do w your * install |
| 19:04.34 | lvlinux | is everything else working? what moduels are currently loaded? |
| 19:05.03 | lvlinux | run "module show" --- does it spit out a bunch? |
| 19:05.46 | cyford33 | i installed it again without doing the last step an it worked |
| 19:06.32 | lvlinux | which last command? |
| 19:06.50 | cyford33 | <PROTECTED> |
| 19:06.50 | cyford33 | <PROTECTED> |
| 19:06.50 | cyford33 | <PROTECTED> |
| 19:07.12 | cyford33 | guess it didnt like the new config file |
| 19:07.22 | lvlinux | ah, so there is something in that config making it fail |
| 19:07.23 | lvlinux | yes |
| 19:07.45 | lvlinux | probably something left over from an earlier version |
| 19:08.24 | cyford33 | wow this engine sucks |
| 19:08.34 | lvlinux | that tutorial is from 2013, and the nerdvittles guy uses his "incredible PBX" magic wand scripts and I have no clue what all mangling is done to asterisk lol. |
| 19:08.38 | lvlinux | how so? |
| 19:09.00 | cyford33 | voice quality is Horrible |
| 19:09.08 | lvlinux | ah |
| 19:09.11 | cyford33 | not like google voice |
| 19:09.22 | cyford33 | i mean ggle tts |
| 19:09.54 | lvlinux | maybe try flite? |
| 19:10.25 | lvlinux | if you are comparing it to that then i think you may be dissapointed with whatever you try. |
| 19:10.34 | lvlinux | i suspect that google's voice engine is pretty big... |
| 19:10.51 | cyford33 | yes, but they starting blocking my server |
| 19:11.07 | cyford33 | <PROTECTED> |
| 19:11.30 | lvlinux | you were giving too many requests i guess |
| 19:13.31 | cyford33 | yes, but i also tryed it from my laptop and got it on my first try |
| 19:13.45 | cyford33 | they say they want make sure i am not a robot |
| 19:13.57 | *** join/#asterisk c|oneman (cloneman@2605:6400:2:fed5:22:0:3b06:3913) |
| 19:14.06 | cyford33 | <PROTECTED> |
| 19:14.13 | cyford33 | not ivr's |
| 19:15.44 | lvlinux | yep |
| 19:16.32 | lvlinux | have u tried festival or flite? |
| 19:25.30 | lvlinux | wow it is horrible... |
| 19:26.22 | cyford33 | lol |
| 19:31.24 | lvlinux | maybe it can b better with some settings though |
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| 19:34.58 | cyford33 | trying now |
| 19:35.08 | cyford33 | but not hearing much differnce lol |
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| 20:18.29 | TazzNZ | hey guys - does asterisk "support" sip publish messages ? the info i found seems to say no |
| 20:19.04 | TazzNZ | PUBLISH sip:vq@10.199.176.168:5060 SIP/2.0 <-- that is the SIP header that is sent |
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