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| 07:44.17 | bhavikpatel6842 | Hi All |
| 07:44.48 | bhavikpatel6842 | I want to enable http websocket in asterisk |
| 07:45.11 | bhavikpatel6842 | For that I made changes and but not working,Any one have idea how to enable |
| 07:46.32 | bhavikpatel6842 | I run command http show status and showing Server Disabled |
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| 08:57.02 | puzzled | morning |
| 09:13.49 | WIMPy | wonders how a morning can be good when you're awake |
| 09:15.41 | bhavikpatel6842 | :-D |
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| 10:18.37 | kafal1 | Hello! I am trying to use Authenticate application in 12.3. But I am getting the following error message: Invalid/unknown command: Authenticate . I have loaded app_agent_pool and app_authenticate modules also. |
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| 11:39.14 | kafal1 | Hello! I am trying to use Authenticate application in 12.3. But I am getting the following error message: Invalid/unknown command: Authenticate . I have loaded app_agent_pool and app_authenticate modules also. |
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| 16:35.15 | _0x5eb_ | hi, is it possible with Asterisk 1.8 to override in sip.conf "host = sip.domain" with a specific IP/hostname (not to be used in any URI)? |
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| 16:54.57 | davlefou | hi, something have no sens for me. I have an realtime asterisk pbx in postgresql. I use ovh line. when i launch it, i can get call from ovh. But if i made an 'sip set debug peer ovh-number', il woks well |
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| 17:00.10 | qakhan | can we setup asterisk as a SMS gateway? |
| 17:00.30 | [TK]D-Fender | qakhan: Depends what it is talking to. |
| 17:01.09 | [TK]D-Fender | qakhan: chan_donge can work, or SIP MESSAGE with a compatible provider |
| 17:01.34 | [TK]D-Fender | davlefou: You are unclear as to what doesn't work. |
| 17:08.08 | qakhan | [TK]D-Fender i want to send SMS to customers phones using asterisk |
| 17:08.34 | [TK]D-Fender | qakhan: And you have not describe what you intend to send it OVER. |
| 17:09.42 | qakhan | verification codes |
| 17:09.55 | qakhan | generated by DB |
| 17:12.20 | [TK]D-Fender | [13:08][TK]D-Fenderqakhan: And you have not describe what you intend to send it OVER. |
| 17:12.40 | davlefou | [TK]D-Fender, when i put an debug one peers ovh, i start to work |
| 17:13.15 | [TK]D-Fender | davlefou: enabling SIP debug doesn't make things start to work. |
| 17:13.39 | davlefou | [TK]D-Fender, if a stop it go on to work but i a made en sip reload or relaunch it, i breakdown |
| 17:14.15 | davlefou | [TK]D-Fender, i am agree with you but i made several teste and i do! |
| 17:14.20 | [TK]D-Fender | davlefou: reload shouldn't change anything unless you made config changes. |
| 17:14.49 | [TK]D-Fender | davlefou: And you are not showing anything. |
| 17:17.49 | davlefou | http://pastebin.com/eEQU9APC |
| 17:18.33 | davlefou | [TK]D-Fender, i made en sample, what i do wrong? |
| 17:18.50 | [TK]D-Fender | [Jun 28 19:15:05] NOTICE[14519][C-00000019]: chan_sip.c:25533 handle_request_invite: Failed to authenticate device "Sabrine El Hammami" <sip:0422130140@sip.ovh.fr;user=phone>;tag=10874-RZ-03952261-199dee883 |
| 17:18.56 | [TK]D-Fender | I see clear auth errors |
| 17:19.38 | davlefou | [TK]D-Fender, yes, juste after my reload, look forward |
| 17:20.14 | *** join/#asterisk theron (~theron@173-29-74-236.client.mchsi.com) |
| 17:20.41 | [TK]D-Fender | I see one call coming in, dialing 3 things, actually dialing one and choosing to end the call... |
| 17:23.18 | [TK]D-Fender | [Jun 28 19:16:30] WARNING[14519]: chan_sip.c:4176 retrans_pkt: Retransmission timeout reached on transmission 3eeb6d4e6aa39d643a55e76c309483fe@91.121.136.189:5060 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 17:23.27 | [TK]D-Fender | You ar also not getting responses from this channel... |
| 17:23.34 | [TK]D-Fender | And you are NOT looking at that call. |
| 17:25.37 | *** join/#asterisk syadnom (~syadnom@host-184-166-238-96.bln-mt.client.bresnan.net) |
| 17:25.57 | syadnom | hi all. anyone know of a good CDR analytics software? |
| 17:35.00 | qakhan | [TK]D-Fender we have mobile app which takes user info like phone#, address, etc.. and send it to one of our DB. DB get the info and generates a code. we want to send that code to user |
| 17:35.27 | [TK]D-Fender | qakhan: you keep saying where you are getting to data FROM, not how you intend to send it OUT. |
| 17:35.57 | qakhan | through Asteisk using GSM card |
| 17:36.08 | qakhan | can we do that |
| 17:36.13 | [TK]D-Fender | qakhan: And how does * talk to that card? |
| 17:36.29 | qakhan | that is my question |
| 17:38.23 | [TK]D-Fender | qakhan: read the instrucstions for the card. You seem to have a probelm being specific with the model you're looking at. |
| 17:39.00 | qakhan | check this http://shop.atcom.cn/index.php?main_page=product_info&cPath=4&products_id=27 |
| 17:39.48 | [TK]D-Fender | qakhan: Go download a manual |
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| 19:00.02 | _0x5eb_ | hi, in sip.conf is it possible to associate a peer handling a certain sip domain with a specific hostname or IP address? |
| 19:00.38 | _0x5eb_ | I found that host=mysipdomain is actually defining the SIP domain, not the SIP server to connect to |
| 19:00.51 | gusto | ??? |
| 19:01.21 | gusto | one can restrict registration and communication of a peer through ACLs |
| 19:01.25 | _0x5eb_ | if my SIP domain is: mysipdomain.com and if I want to connect to a server named sipsrv1.mysipdomain.com |
| 19:01.46 | [TK]D-Fender | _0x5eb_: it is the server. the actual IP you send to as well unless you specify an outboundproxy |
| 19:01.59 | WIMPy | Still only possible with pjsip. |
| 19:03.28 | _0x5eb_ | for now I put a fake entry "my.sip.server.ip mysipdomain.com" into my /etc/hosts, I try (again) with outboundproxy (no success last time) |
| 19:04.23 | gusto | does asterisk look into /etc/hosts ? |
| 19:04.41 | [TK]D-Fender | As part of its DNS lookup, yes |
| 19:04.47 | gusto | hm |
| 19:05.33 | gusto | maybe he has not refreshed yet |
| 19:05.48 | gusto | and using dnsmgr is not a good idea too |
| 19:06.05 | _0x5eb_ | I confirm that outboundproxy does not work for this case, I had to revert using the ugly /etc/hosts trick |
| 19:06.33 | _0x5eb_ | yes, I'd like to override any DNS lookup for this peer but don't find a way to do that |
| 19:06.48 | gusto | why dont you use the IP address? |
| 19:06.54 | gusto | I am doing that |
| 19:07.25 | _0x5eb_ | OK, but how do I then specify the SIP domain? |
| 19:09.21 | _0x5eb_ | if I put the IP address in host=sip.server.ip.address, then it sends INVITE to user@sip.server.ip.address ("domain=sipdomain" does not help) |
| 19:11.57 | gusto | so, and where is the problem, then? |
| 19:13.20 | _0x5eb_ | the SIP server is handling several domains; the correct SIP domain must be given in the INVITE |
| 19:14.32 | _0x5eb_ | I tried outboundproxy=sip.server.ip.address , does not work as well: asterisk is still looking up the "host" SIP domain for DNS A records instead |
| 19:14.40 | [TK]D-Fender | What SIP server? |
| 19:14.40 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
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| 19:15.16 | _0x5eb_ | the remote (peer) SIP server |
| 19:15.31 | *** join/#asterisk kafal1 (~neo@110.225.2.121) |
| 19:16.27 | _0x5eb_ | so to summarize I want to connect to a specific SIP server (ideally using its hostname, but IP address is still OK) |
| 19:16.54 | _0x5eb_ | handling some SIP domain (handling SIP URI like: users@SIPdomain) |
| 19:17.21 | WIMPy | You can't do it with chan_sip unless the dns name and the sip domain are identical. |
| 19:17.35 | WIMPy | Because for chan_sip they are. |
| 19:18.10 | _0x5eb_ | OK, so as long as I use chan_sip, I have to keep using my fake /etc/hosts entry |
| 19:18.12 | gusto | what i have found looking at sip debug is that he does not care about what he is transporting as hostname or whatever, mostly there is false information anyway |
| 19:18.19 | kafal1 | Hello! I am working on Asterisk 12.3. I am able to use Authenticate application and authenticate the agent. After that I use AgentLogin(AgentID) applciation. The agent is logged in successfully but it is not firing AgentLogin event. Any ideas why? |
| 19:20.31 | [TK]D-Fender | kafal1: Show us |
| 19:20.33 | [TK]D-Fender | ~pb |
| 19:20.34 | 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. |
| 19:20.35 | [TK]D-Fender | ^^^ |
| 19:20.42 | _0x5eb_ | WIMPy, gusto: so that would explain why outboundproxy is ignored for a peer? |
| 19:21.09 | [TK]D-Fender | _0x5eb_: show us your peer & the call. |
| 19:21.49 | WIMPy | I'm not sure what it explains or what outboundproxy is supposed to so. AFAIK there's no way to do it. |
| 19:23.08 | _0x5eb_ | OK, found the solution: outboundproxy is ignored... as long as we do not add ",force" after the hostname |
| 19:23.37 | WIMPy | huh? |
| 19:23.42 | _0x5eb_ | so: "outboundproxy=my.sip.server,force" in sip.conf does the trick :) |
| 19:23.58 | _0x5eb_ | if I don't put the ",force", it is apparently ignored |
| 19:24.30 | WIMPy | has never seen a ",force" before. |
| 19:24.54 | _0x5eb_ | I just found it by looking carefully at http://www.voip-info.org/wiki/view/Asterisk+sip+outboundproxy |
| 19:25.41 | gusto | why do you need a outboundproxy anyway? |
| 19:25.46 | gusto | what is that account? |
| 19:26.53 | WIMPy | needs to check that ,force thing later on. |
| 19:27.36 | _0x5eb_ | outboundproxy: I need it to override the DNS lookup of the domain specified by "host=", in order to specify a very specific server |
| 19:28.02 | _0x5eb_ | for instance: I have a SIP server accessible both with public IP address and private IP address |
| 19:28.05 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 19:28.40 | _0x5eb_ | I want to use the private IP address when I have a VPN to the SIP server, not doing a lookup on the (public) domain |
| 19:29.42 | gusto | oh |
| 19:30.15 | kafal1 | <PROTECTED> |
| 19:31.06 | [TK]D-Fender | kafal1: What's the point of the confbridge after agentlogin? |
| 19:32.03 | gusto | _0x5eb_: i would do it over routing |
| 19:32.04 | kafal1 | I was testing how to put an agent in a conference. |
| 19:32.22 | [TK]D-Fender | You don't |
| 19:32.34 | [TK]D-Fender | kafal1: you don't seem to understand what agentlog in. |
| 19:32.40 | [TK]D-Fender | is* |
| 19:33.01 | [TK]D-Fender | kafal1: they SIT there in that app waiting for a queue to get a call to distribute |
| 19:34.00 | kafal1 | ok |
| 19:34.53 | _0x5eb_ | gusto: is it possible with Asterisk 1.8? |
| 19:35.13 | [TK]D-Fender | [15:01]WIMPyStill only possible with pjsip. |
| 19:35.42 | gusto | i meant over ip routing in the operating system |
| 19:36.00 | gusto | but i am not sure even that would solve that problem |
| 19:36.10 | gusto | and sip proxy does not seem to be an idea as well |
| 19:36.25 | gusto | it should not matter what IP address he comes from |
| 19:36.26 | _0x5eb_ | gusto: OK, well except (e.g.) if the public IP address is (also) the address of the VPN concentrator |
| 19:36.44 | WIMPy | [TK]D-Fender: that's what I found so far, and also what I've been told so far. |
| 19:36.53 | gusto | asterisk does not care if it is a VPN or whatever |
| 19:37.43 | _0x5eb_ | gusto: no but altering the route (at the OS level) wouldn't work in this case |
| 19:38.03 | gusto | i know |
| 19:38.25 | gusto | and your proxying does not make any sense as well |
| 19:38.29 | _0x5eb_ | gusto: btw, this also solves the issue described here: http://lists.digium.com/pipermail/asterisk-users/2013-April/278434.html |
| 19:38.43 | gusto | when you want to make calls from there you will also have to take care of RTP |
| 19:38.52 | _0x5eb_ | no it's just a trick to work around the confusion between host and domain |
| 19:40.03 | _0x5eb_ | well RTP will be handled correctly since it will not go through the NAT processing being already a RFC1918 address |
| 19:40.05 | WIMPy | Wow. Haven't seen that ! thing, either. |
| 19:40.26 | gusto | that looks like a lot of crap to me |
| 19:40.28 | _0x5eb_ | WIMPy: yes, I've been looking for doc about that without success |
| 19:40.49 | WIMPy | _0x5eb_: Congratulations. You told me more interesting things about Asterisk than anyone else this year so far. |
| 19:44.50 | _0x5eb_ | WIMPy: that "!" stuff is really weird, it actually puts your from address in the INVITE |
| 19:45.45 | _0x5eb_ | (instead of the to) |
| 19:45.58 | gusto | well, i am not against experimenting with asterisk, but then one can not expect help, with something no one else tried before |
| 19:46.18 | gusto | and i am still puzzled by the use-case of yours |
| 19:48.49 | _0x5eb_ | anyway if anyone is using ippi, I confirm that their SIP-to-Skype gateway (described in the previous link) is working using the "outboundproxy=sip.ippi.fr,force" trick |
| 19:51.41 | gusto | aha, so it actually is that skype gw |
| 19:51.42 | gusto | hm |
| 19:51.56 | WIMPy | _0x5eb_: That ,force thing is genious. I wonder why nobody told me before. |
| 19:52.25 | WIMPy | Now next time there is some internet outage, Asterisk will hopefully recover when connectivity is re-established. |
| 19:52.28 | *** join/#asterisk slackie (~x@unaffiliated/slackie) |
| 19:52.35 | gusto | i am asking myself why the force option is not the default since the reason of having proxy is to send all messages over it in the first place |
| 19:53.20 | _0x5eb_ | [TK]D-Fender: thanks for pointing me out outboundproxy, that solved two issues at once (as long as I use the ",force") |
| 20:09.33 | *** join/#asterisk slackie (~x@unaffiliated/slackie) |
| 20:14.43 | kafal1 | please point me to some sample code of getting agent info from db in asterisk 12.3 . In Asterisk 1.4, I was fetching the info from ast_config table. |
| 20:16.38 | [TK]D-Fender | What "agent info"? What db & table? |
| 20:20.33 | *** join/#asterisk Draecos (~Draecos@94.205.252.227) |
| 20:22.10 | kafal1 | We were fetching agents.conf data from db using Static Realtime. As the agents.conf schema has changed in 12.x, I just wanted to know how should the data be stord in db now. Any code sample or documentation link will help a lot. |
| 20:27.36 | [TK]D-Fender | I don't see it anywhere... |
| 20:33.49 | *** join/#asterisk evilman_home (~evilman_h@89-179-77-66.broadband.corbina.ru) |
| 20:47.02 | kafal1 | [TK]D-Fender can you please guide me on how to store/fetch agents.conf info from db in asterisk 12.x? |
| 20:47.13 | [TK]D-Fender | [16:27][TK]D-FenderI don't see it anywhere... |
| 20:48.37 | kafal1 | ok. i will try to search. |
| 20:55.49 | davlefou | [TK]D-Fender, it seems an dns problem, |
| 20:56.08 | davlefou | the sql should use ip but not ip, |
| 20:57.23 | [TK]D-Fender | you aren't making any sense |
| 20:58.19 | davlefou | the sql should use ip but not dns. In my case, i put sip.ovh.fr in host but it look for the ip. |
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| 21:02.24 | davlefou | these request seems strange : instruction : SELECT * FROM sip_conf WHERE name = 'sip.ovh.fr' AND host = 'dynamic' |
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| 21:05.44 | [TK]D-Fender | davlefou: And you are showing a single query with NO sense of context |
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| 21:11.26 | [TK]D-Fender | kafal1: https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 21:11.35 | [TK]D-Fender | kafal1: Doesn't seem to list agents.conf at all for support |
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| 23:00.33 | davlefou | [TK]D-Fender, i work now. |
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| 23:33.40 | ChannelZ | damn my C is rusty |
| 23:36.11 | mjordan | are you looking at Asterisk? |
| 23:41.37 | *** join/#asterisk PHunter (~Pghpunkid@ip68-3-220-113.ph.ph.cox.net) |
| 23:42.08 | PHunter | im being plagued by 1 site of mine flip flopping on registration or something.. |
| 23:42.12 | PHunter | NOTICE[28772]: chan_sip.c:29427 sip_poke_noanswer: Peer 'management.101' is now UNREACHABLE! Last qualify: 86 |
| 23:42.19 | PHunter | Any ideas why this would go berserk? |
| 23:42.38 | PHunter | I get this every like 10 minutes for all of the enpoints |
| 23:42.43 | PHunter | off an on. |
| 23:42.46 | PHunter | and* |
| 23:43.12 | PHunter | Im constantly monitoring the connection status, and have plenty of bandwidth. |
| 23:44.00 | PHunter | [Jun 28 16:42:43] NOTICE[28772]: chan_sip.c:23522 handle_response_peerpoke: Peer 'roy.301' is now Reachable. (63ms / 2000ms) |
| 23:44.06 | PHunter | [Jun 28 16:43:47] NOTICE[28772]: chan_sip.c:29427 sip_poke_noanswer: Peer 'roy.301' is now UNREACHABLE! Last qualify: 63 |
| 23:44.16 | PHunter | 1minute 4 seconds and it dropped. |
| 23:45.11 | WIMPy | Are these peers behind a NATting router? |
| 23:45.41 | PHunter | Yes. |
| 23:46.18 | WIMPy | Looks like it forgot the "connection". |
| 23:46.31 | PHunter | Router sucks at NAT'ing? |
| 23:46.54 | WIMPy | Some do |
| 23:47.24 | PHunter | Would SIP ALG be of any use in this case? |
| 23:47.40 | WIMPy | It will probably cause some extra headaches. |
| 23:47.41 | PHunter | its a Cisco E4500. |
| 23:47.56 | PHunter | Headaches are not what I want/need. This is already causing me a big one. |
| 23:48.25 | PHunter | So ditch the router and try again? |
| 23:49.04 | WIMPy | That's soemtimes the only thing that helps. |
| 23:49.24 | PHunter | Okay. ill find a way to drop it out of the mix. |
| 23:49.26 | WIMPy | Although I'd hope that yours would be configurabe enough to make it work. |
| 23:49.47 | PHunter | Its a store bought Cisco one. i hate them with a passion. This makes me happy to chuck it. |
| 23:52.16 | PHunter | Any recommendations as far as a router goes? |
| 23:52.53 | WIMPy | prefers to just use a linux box. |
| 23:53.50 | PHunter | Unfortunately our monkeys are smart enough to leave a linux box or any other computer for that matter alone. |
| 23:55.27 | PHunter | I suppose most routers shouldnt forget NAT stuff. But i read that Cisco E4500's have NAT timeout issues. |
| 23:55.46 | WIMPy | They all have timeouts. |
| 23:56.08 | PHunter | True, but low thresholds. |