| 00:19.02 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:19.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:35.35 | *** join/#asterisk Asterisco (~catan@host230-43-dynamic.181-80-r.retail.telecomitalia.it) |
| 08:35.37 | Asterisco | hi |
| 08:35.55 | Asterisco | I've a new problem! :'( |
| 08:36.27 | Asterisco | i've two asterisk server 1 with static ip and 1 with dynamic ip |
| 08:37.31 | Asterisco | Now, i could connect an internal ip of server with dynamic ip that call an internal of server with static ip |
| 08:38.00 | Asterisco | But not the other way round |
| 08:38.45 | Asterisco | Goto (interni,9915,1) |
| 08:38.45 | Asterisco | <PROTECTED> |
| 08:38.45 | Asterisco | [Jul 17 10:18:51] WARNING[18262][C-00000001]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Subscriber absent) |
| 08:38.45 | Asterisco | <PROTECTED> |
| 08:38.45 | Asterisco | <PROTECTED> |
| 08:38.53 | Asterisco | some ideas? |
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| 10:12.42 | FarhaadN | i install asterisk 14.5.0 and after 2 core reload, asterisk crash |
| 10:30.00 | FarhaadN | anyone can help me? |
| 11:07.05 | *** join/#asterisk Asterisco (~catan@host230-43-dynamic.181-80-r.retail.telecomitalia.it) |
| 11:15.13 | Asterisco | Unable to create channel of type 'IAX2' (cause 20 - Subscriber absent) |
| 11:15.24 | Asterisco | how can I resolve it? |
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| 11:25.19 | Samot | Asterico: You need to register the IAX trunk to the other server |
| 11:26.09 | Asterisco | Samot... in server with dynamic id... i've put in [general] of iax.conf this: register => europtical:linux@188.15.111.173 |
| 11:26.37 | *** join/#asterisk FarhaadN (~Farhad@82.99.206.194) |
| 11:26.40 | FarhaadN | i install asterisk 14.5.0 and after 2 core reload, asterisk crash |
| 11:26.40 | Samot | OK |
| 11:26.45 | Samot | Is itregistered? |
| 11:26.47 | Asterisco | but in the other ??? |
| 11:27.24 | Asterisco | yes |
| 11:27.33 | Asterisco | iax2 show peers |
| 11:27.43 | Asterisco | europtical/cerz 188.15.111.173 (S) 255.255.255.255 4569 Unmonitored |
| 11:28.15 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 11:28.33 | Asterisco | in the main server (with static ip) how can I register the dynamic ip server? |
| 11:29.11 | Samot | You have to set the trunk up properly so it can be registered to |
| 11:29.58 | Asterisco | in iax.conf of (dynamic ip server) ... |
| 11:30.06 | Samot | No |
| 11:30.14 | Samot | In the OTHER one |
| 11:30.29 | Samot | The one it is supposed to register to. |
| 11:30.30 | dnit | Hi lets say I bridged two different channels and one of the channels hungup. Both of them dies. How can I unbridge I channel before the other hangup |
| 11:30.39 | Asterisco | [europtical] |
| 11:30.40 | Asterisco | host=188.15.111.173 |
| 11:30.45 | Asterisco | in the other... |
| 11:30.53 | Asterisco | [cerza] |
| 11:30.57 | Asterisco | host=dynamic |
| 11:31.05 | Samot | OK |
| 11:31.13 | Asterisco | so where is the error? |
| 11:31.15 | Samot | Is the user and password configured properly? |
| 11:31.21 | Asterisco | yes |
| 11:31.58 | Samot | Well it's not showing the other server registered. |
| 11:32.17 | Samot | The other server can't call out that trunk because it's not registered. |
| 11:32.19 | Samot | That's your issue. |
| 11:32.27 | Samot | You need to fix your IAX configs and truns. |
| 11:37.34 | dnit | Is there a way to remove a channel from bridge before the other channel hangsup ? |
| 11:37.54 | Asterisco | Samot can you watch my 2 iax.conf? |
| 11:38.29 | Samot | dnit: The hangup of one channels ends the bridge between two channels. |
| 11:38.35 | Samot | dnit: What are you trying to do? |
| 11:38.56 | Samot | Asterisco: I don't use IAX. |
| 11:39.01 | Samot | I've already stated that. |
| 11:39.10 | Samot | Stop PMing me. |
| 11:39.14 | Samot | I didn't ask for it |
| 11:39.19 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 11:39.36 | Samot | And I don't read them. |
| 11:40.10 | Samot | STOP PMING ME |
| 11:40.15 | Asterisco | Samot excuse me |
| 11:40.24 | dnit | I have a scenario where I want my agent to hear the early media. ( Now the agent is assigned to the call after it is dialled ). |
| 11:40.31 | Samot | I didn't not ask you to do it. I don't read them. |
| 11:40.34 | Asterisco | https://pastebin.com/1nT8VFQj |
| 11:40.51 | Asterisco | this is my iax.conf of server with static ip |
| 11:41.00 | Samot | dnit: Is your agent making the call? |
| 11:41.02 | Asterisco | ..... |
| 11:41.14 | dnit | So inorder to make the agent hear the early media I bridged the outgoing call and agent |
| 11:41.19 | Asterisco | ...and this is iax.conf of server with dynamic ip: https://pastebin.com/L8jRniDL |
| 11:41.31 | dnit | Samot: No agent is not making the call . It is attached afterwards |
| 11:41.53 | Samot | dnit: Then how do you expect the agent to get early media? |
| 11:42.09 | Samot | How will they hear early media? |
| 11:42.15 | dnit | For that reason only I had to use bridge |
| 11:42.16 | Samot | It's not for the callee, it is for the caller. |
| 11:42.29 | Samot | Early media is not for the callee. |
| 11:42.33 | Asterisco | Samot could you see my config files? |
| 11:43.39 | Samot | Asterisco: cerza trunk has not host= setting |
| 11:43.45 | Samot | That is required |
| 11:44.00 | Samot | dnit: Your agent receiving the call is going to have their phone ring. |
| 11:44.02 | Samot | That's it. |
| 11:44.08 | Samot | There is no early media for that agent. |
| 11:44.21 | Asterisco | ?!?!?!?!?!?!? |
| 11:44.45 | Samot | https://www.irccloud.com/pastebin/imhtSL04/ |
| 11:45.02 | Samot | There is no host= setting in this. |
| 11:45.07 | Samot | It is needed. |
| 11:45.33 | Asterisco | in ?!?!? |
| 11:45.46 | Asterisco | i need put host= dynamic? |
| 11:46.46 | dnit | Asterico: Yes put hots=dynamic in [general] |
| 11:46.48 | Asterisco | i put host setting in [cerza] and [europtical] |
| 11:47.05 | Asterisco | tnx |
| 11:47.47 | Asterisco | i need put host=188.15.111.173 in the [generic] of the other iax.conf file? |
| 11:47.48 | Samot | Asterisco: NO |
| 11:47.54 | dnit | Samot: I am not calling my agent. My agent are already there on one bridge. I am making an outgoing call and bridging the agent and the outgoing call. |
| 11:48.03 | Asterisco | Samot??? |
| 11:48.03 | Samot | In the PBX that is REGISTERING TO the other PBX |
| 11:48.07 | Asterisco | ok |
| 11:48.08 | Samot | It needs a host= setting |
| 11:48.18 | Samot | Which would be the IP of the other server. |
| 11:48.28 | Asterisco | so... |
| 11:48.35 | Samot | dnit: That's not what you said when I asked |
| 11:48.39 | Asterisco | [generic]; cerza hots=dynamic |
| 11:48.42 | Asterisco | and stop |
| 11:48.56 | Asterisco | host |
| 11:48.58 | Asterisco | host=dynamic |
| 11:49.01 | Samot | There is nothing wrong on the static IP server. |
| 11:49.35 | Samot | The server with the dynamic IP has a misconfigured trunk. |
| 11:49.40 | Samot | It is missing the host= setting |
| 11:49.42 | Samot | Period. |
| 11:49.44 | Samot | Fix it. |
| 11:50.07 | dnit | Samot: You asked if my agent is making the call but agent is realy not making the call, I am attaching agent on the call ( at the time of Dail I don't know which agent is going to assigned to this call) |
| 11:50.14 | Samot | dnit: How is the agent on the bridge? |
| 11:50.24 | Samot | dnit: You need to explain how this is actually happening. |
| 11:50.29 | dnit | Ok |
| 11:50.30 | Asterisco | samot.... is the same problem |
| 11:51.32 | Asterisco | this is the iax.conf of my server with dynamic ip: https://pastebin.com/Qak3H6Dn |
| 11:52.31 | Asterisco | and this is my iax.conf of my server with static IP |
| 11:52.33 | Asterisco | https://pastebin.com/mTz6CsG4 |
| 11:52.35 | Samot | host=dynamic <-- NOT WHAT I SAID AT ALL |
| 11:52.48 | Asterisco | ?!?!?!?!? |
| 11:52.54 | Samot | 7:48:09 AM <Samot> It needs a host= setting |
| 11:52.54 | Samot | 7:48:19 AM <Samot> Which would be the IP of the other server. |
| 11:52.56 | Asterisco | what i need to put? |
| 11:53.09 | Samot | Where is it REGISTERING to? |
| 11:53.11 | Asterisco | I've a Dynamic IP |
| 11:53.14 | Samot | FFS |
| 11:53.17 | Samot | The OTHER SERVER |
| 11:53.25 | Asterisco | ? |
| 11:53.30 | Samot | 188.15.111.173 <-- THIS |
| 11:53.37 | Samot | The OTHER server |
| 11:53.39 | Asterisco | host=188.15.111.173 |
| 11:53.41 | Asterisco | so^ |
| 11:53.43 | Asterisco | ??? |
| 11:53.44 | Samot | YES |
| 11:53.44 | Asterisco | ok |
| 11:53.46 | Asterisco | :D |
| 11:54.59 | Asterisco | Samot... is the same!!! |
| 11:55.24 | Asterisco | i've got the same error |
| 11:56.16 | Samot | OK |
| 11:56.26 | Samot | I will say this for a third and final time... |
| 11:56.33 | Samot | I do not use IAX. |
| 12:08.30 | post-factum | hey. looks like circular dep is present in 14.6. res_pjsip wants ast_sip_session_register_supplement, which resides in res_pjsip_session, but res_pjsip_session wants ast_sip_get_distributor_serializer, which resides in res_pjsip. i've described it in 27132. may someone help me with this? |
| 12:08.36 | post-factum | file: ^^ |
| 12:09.10 | dnit | Samot: First I intiate a call lets say X via AMI originate command. After I get the callstarted event ( custom userevent). I picks up one agent from my available agents and bridge my agent channel with outgoing channel. |
| 12:09.33 | dnit | Now this bridging is also initiated via AMI |
| 12:09.52 | Samot | OK. |
| 12:10.11 | dnit | But as soon as my outgoign channel hangus up my agent channel hangusup which I want to avoid. |
| 12:10.12 | Samot | So you want your agent to hear early media of the outbound call to the PSTN? |
| 12:10.20 | dnit | Yes |
| 12:10.24 | Samot | You can't. |
| 12:10.54 | Samot | Early Media is a response from the called side. |
| 12:10.54 | file | post-factum: there's already an issue for that |
| 12:10.56 | dnit | I can't help it from hanging up or I can't make him hear the early media ? |
| 12:11.10 | Samot | Well I'm not sure why your originate is hanging up. |
| 12:11.16 | Samot | But you cannot hear early media |
| 12:11.30 | Samot | That is up to the other side answering the call. |
| 12:11.34 | dnit | But I achieved in making him hear the ealy media by bridging. |
| 12:11.47 | dnit | I can hear the early media just fine. |
| 12:12.25 | Samot | Show this happening.. |
| 12:12.28 | dnit | Yes ofcourse if the other side is sending the earl;y media |
| 12:12.31 | post-factum | file: 26518? |
| 12:12.35 | file | yes |
| 12:12.37 | Samot | So a call getting hungup and dropping both channels from the originate... |
| 12:12.50 | Samot | Why do you need early media? |
| 12:13.01 | post-factum | file: okay. thanks for checking |
| 12:13.21 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d1f:ffcb:a1e4:a2b:2d5f:119b) |
| 12:13.47 | dnit | In early media we receive Out of service messages. So that our agent can hear those message and the mark the numbers are unvailable / Not in use any more |
| 12:14.08 | Samot | Uhm. |
| 12:14.20 | Samot | OK. |
| 12:14.32 | Samot | Show a call originating |
| 12:14.46 | dnit | ok will paste the bin |
| 12:14.51 | Samot | First, what tech are you using? |
| 12:15.01 | dnit | PJSIP |
| 12:15.02 | Samot | In the Dial() |
| 12:15.06 | dnit | yes |
| 12:15.06 | Samot | And that's your problem. |
| 12:15.33 | dnit | why ? |
| 12:15.42 | Samot | You are originating a call from Asterisk via PJSIP to X |
| 12:15.53 | Samot | Via PJSIP |
| 12:16.22 | Samot | you are making another Dial()? |
| 12:16.37 | dnit | No Just one dial. |
| 12:16.44 | Samot | How do you call both locations? |
| 12:17.05 | Samot | How is the AMI triggered command calling the agent and the end user? |
| 12:18.18 | Samot | You are calling two destinations, that's one channel each. |
| 12:18.28 | Samot | Then you are bridging those two channels... |
| 12:18.33 | Samot | How are you doing this? |
| 12:19.35 | dnit | I have a small dial plan for bridging that. where I set the targetchannel via SetVar and then use "redirect" action to call the dialplan which bridges the targetchannel (pstn call) |
| 12:20.09 | Samot | Show your originate command. |
| 12:20.16 | Samot | Or whatever diaplan is doing this. |
| 12:20.29 | Samot | I need to see the code/logic |
| 12:20.37 | dnit | bridging dialplan or the one which is makeing the call to pstn ? |
| 12:20.43 | Samot | Both |
| 12:20.46 | dnit | Ok |
| 12:20.50 | Samot | Everyting that starts the call |
| 12:20.56 | Samot | To the agent |
| 12:20.58 | Samot | then to the end user |
| 12:21.00 | Samot | then to the brige |
| 12:28.47 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:30.16 | *** part/#asterisk post-factum (~post-fact@vulcan.natalenko.name) |
| 12:31.10 | FarhaadN | i install asterisk 14.5.0 and after 2 core reload, asterisk crash |
| 12:32.00 | Samot | OK |
| 12:32.03 | file | FarhaadN: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 12:32.05 | Samot | What do the logs show? |
| 12:34.17 | FarhaadN | Samot: nothing |
| 12:34.46 | Samot | Show the logs before the crash |
| 12:34.51 | Samot | ~pb |
| 12:34.52 | infobot | pastebin is, like, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:35.32 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:39.15 | dnit | Samot: here are the dialplans , https://pastebin.com/icLmyt3y. I am calling aMI originate action with context: inbound, and channel : number@outbound. |
| 12:42.30 | dnit | Then bridging the channels, using bridge-channel dial plan where I set the TargetChannel first via SetVar and then use redirect action with context: bridge-channel |
| 12:48.40 | Samot | OK, so you've bridged two channels together.. |
| 12:48.50 | Samot | When one of them hangs up, that's it. |
| 12:49.04 | Samot | You want to keep the other channel going? |
| 12:49.16 | Samot | Or one of them? |
| 12:50.02 | dnit | Yes I dont want my agent channel to hangup when the other hangs up |
| 12:51.28 | dnit | I can't use confbridge because the agent is available after the dial and if I use confbridge my outgoing channel answers and hangsup. |
| 12:51.59 | Samot | You know that Originate has an EarlyMedia setting? |
| 12:52.20 | Samot | But that's not really going to do much with the agent. |
| 12:52.21 | Samot | See. |
| 12:52.34 | Samot | Originate is sending the call to the agent. |
| 12:52.37 | Samot | Directly |
| 12:53.14 | Samot | Then once that agent answers, you are redirecting to the outbound context. |
| 12:53.18 | Samot | Which dials. |
| 12:53.25 | Samot | That is where the early media is going to happen |
| 12:53.27 | Samot | On that channel |
| 12:53.56 | Samot | So yes, it works when you bridge them because now they are bridge and the media on one channel will flow to the bridged channel. |
| 12:54.41 | Samot | Why aren't you just using a Local channel? |
| 12:55.14 | *** join/#asterisk jgowdy (uid239878@gateway/web/irccloud.com/x-fvamdkeqkhqrgdhq) |
| 12:55.27 | dnit | local channel for agent? |
| 12:55.33 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:55.39 | Samot | Well.. |
| 12:55.47 | Samot | I mean the one issue with AMI, regardless.. |
| 12:56.09 | Samot | Is that you're going to send the Originate and nothing is going to happen until that originate is done. |
| 12:56.29 | Samot | When the hangup happens, the originate is done. |
| 12:56.34 | Samot | call is over. |
| 12:57.01 | *** part/#asterisk FarhaadN (~Farhad@82.99.206.194) |
| 12:58.36 | Samot | There is nothing, that I know of, in Originate that lets you issue a command to continue in the dialplan when the Originate is completed. |
| 12:59.24 | Samot | So yes, I would use Local. |
| 12:59.36 | Samot | A Local channel is a 2 leg channel. |
| 13:00.14 | Samot | It will go to the "inbound" context, you can issue a Dial() to the Agent, which will allow you to send the needed commands to continue in the dialplan or jump to another context, etc... |
| 13:00.29 | Samot | When the Agent answers, it will then fire off the Outbound context.. |
| 13:00.36 | Samot | Which will call the end user. |
| 13:00.47 | Samot | When the end user answers, two calls are bridged. |
| 13:01.30 | Samot | If the end user hangs up, then the Dial() from the "inbound" context will complete and continue on in the dialplan or jump contexts as you have told it to. |
| 13:03.07 | dnit | But thats the problem. We are making the call the first, then assigning the agent so untill the dial has actually happened we are not sure of which agent is going to be assigned to this call. |
| 13:03.27 | Samot | OK |
| 13:03.33 | Samot | So you're calling the end user first? |
| 13:03.50 | dnit | Yes |
| 13:03.57 | Samot | Oh, one of those. |
| 13:04.08 | Samot | The type I hang up on immediately. :) |
| 13:04.29 | Samot | Well then you do the same thing |
| 13:04.34 | Samot | Just the opposite. |
| 13:04.42 | Samot | Dial the user first |
| 13:04.52 | Samot | Then when they answer, dial the agent. |
| 13:05.06 | Samot | Or agents |
| 13:06.08 | dnit | Yes this is what is exactly happening right now. But this way agents dont hear the early media for bad calls ( OOS nessages in early media) and the system can't mark these calls as Out of Service messages. |
| 13:06.14 | Samot | And if all you want to hear early media for is if the number is Out of Service.... |
| 13:06.28 | Samot | You shouldn't be counting on hearing error recordings for this |
| 13:06.31 | dnit | Yes. |
| 13:06.34 | Samot | You should be looking at the HANGUPCAUSE |
| 13:06.44 | Samot | Which will tell you why the call didn't happen |
| 13:07.00 | Samot | Bad Number, Busy, No Answer, Timeout, etc, etc |
| 13:07.20 | dnit | HANGUPCAUSE doesnt help much because there is a timoeout of 45 seconds so HANGUPCAUSE always gives NOANSWER |
| 13:07.27 | Samot | Depending on early media to detect "bad numbers" is wrong. |
| 13:07.33 | Samot | No. |
| 13:07.38 | Samot | If the number is BUSY it happens right away |
| 13:07.47 | Samot | If the number is dead it can happen right away |
| 13:07.51 | Samot | I do this all the time. |
| 13:08.03 | dnit | No there is no SIP signalling received for such numbers |
| 13:08.12 | Samot | Uhm |
| 13:08.13 | Samot | Yes. |
| 13:08.14 | dnit | It doesn't even send a 180 ringing. |
| 13:08.17 | Samot | Well there should be. |
| 13:08.20 | Samot | OK. |
| 13:08.23 | Samot | So use AMD |
| 13:08.23 | [TK]D-Fender | THERE ISN'T |
| 13:08.30 | [TK]D-Fender | WEER ON WEEK 2 OF THIS AT THE VERY LEAST |
| 13:08.33 | Samot | No, you're right.. |
| 13:08.36 | [TK]D-Fender | oops... too early for caps |
| 13:08.47 | [TK]D-Fender | suspects it'll come soon |
| 13:09.01 | Samot | I already had them with Asterico. |
| 13:09.07 | Samot | So feel free. |
| 13:09.16 | [TK]D-Fender | same here yesterday |
| 13:10.57 | Samot | 180 Ringing is to show that the "user" is being contacted |
| 13:10.59 | Samot | Or tried |
| 13:11.05 | Samot | Not that there is actual "ringing |
| 13:11.07 | Samot | Not that there is actual "ringing" |
| 13:11.17 | Samot | Early Media is going to return a 183 Session in Progress |
| 13:11.34 | Samot | Followed by either the 200 OK to show the call was setup |
| 13:11.45 | Samot | Or the proper error code for why it wasn't setup. |
| 13:11.54 | [TK]D-Fender | If that's the case you're never getting early media in that originate |
| 13:12.07 | [TK]D-Fender | because * doesn't pass anything until the channel is ANSWERED |
| 13:12.07 | Samot | Well early media is possible in Originate |
| 13:12.14 | Samot | There is a setting for it. |
| 13:12.18 | Samot | To use early media. |
| 13:12.18 | [TK]D-Fender | Is there? |
| 13:12.21 | Samot | Yup. |
| 13:12.26 | [TK]D-Fender | that's news |
| 13:12.29 | [TK]D-Fender | And.... is it working? |
| 13:12.33 | Samot | I don't know |
| 13:12.42 | Samot | I just saw it in the Wiki today |
| 13:13.09 | Samot | Nope, it's been there for a while. |
| 13:13.17 | Samot | This is the v11 page |
| 13:13.23 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate |
| 13:13.33 | dnit | Yes I dont get 200 for these call I just get 183 and then early media. |
| 13:13.34 | Samot | EarlyMedia - Set to true to force call bridge on early media.. |
| 13:13.43 | Samot | 183 = Early Media |
| 13:13.46 | Samot | INVITE |
| 13:13.51 | Samot | 100 Trying as the response |
| 13:14.02 | Samot | Followed by either a 180 Ringing or a 183 Session in Progress. |
| 13:14.12 | Samot | Or even a 180 Ringing followed by a 183 Session in Progress |
| 13:14.35 | Samot | Followed by, hopefully, 200 OK for a connected/answered call |
| 13:14.42 | dnit | Yes I meant I only receive 183 singalling and any 4xx or 5xx reponse. |
| 13:14.48 | Samot | Right |
| 13:14.55 | dnit | not $xx 5xx |
| 13:14.56 | Samot | 4XX and 5XX responses are the CALL BEING BAD |
| 13:15.09 | Samot | User and Server responses |
| 13:15.42 | Samot | 488, 415, 480, 408 those are all failure responses for the user being called |
| 13:15.54 | dnit | This is not present just the early meaid saying the number is out of service . So lack of 200 makes it unanswered. |
| 13:16.06 | Samot | 200 OK means it was answered |
| 13:16.11 | [TK]D-Fender | Are you using the option he told you to use? |
| 13:16.14 | [TK]D-Fender | Where's the debug? |
| 13:16.17 | Samot | Anything else is an error and unanswered. |
| 13:16.18 | [TK]D-Fender | Where's the AMI dump? |
| 13:16.42 | Samot | dnit: The response code will follow after the 183 |
| 13:16.46 | Samot | Once the media is done |
| 13:17.03 | Samot | "Out of Service Message" then response code |
| 13:17.17 | Samot | There has to be a response following a 180 or 183 |
| 13:17.18 | dnit | Ideally it should follow but before that 45 seconds are up and * teminates the call |
| 13:17.20 | Samot | or the call will die. |
| 13:17.31 | Samot | It depends on the RESPONSE |
| 13:17.40 | Samot | There is no 45 seconds of dialing when the number is bad. |
| 13:17.57 | Samot | Or busy |
| 13:18.24 | Samot | A SIP response code is REQUIRED |
| 13:18.30 | Samot | Early media or not. |
| 13:18.43 | Samot | It is required to show the actual state of the call setup |
| 13:18.50 | Samot | If it was setup or not |
| 13:19.01 | Samot | Early Media has nothing, _nothing_, to do with it |
| 13:19.42 | Samot | It is just them playing some media before the call is setup. |
| 13:20.06 | Samot | Early Media happens during the initial INVITE |
| 13:20.14 | Samot | The reason you are seeing a 180 and a 183 |
| 13:20.20 | Samot | Is because your carrier is making the calll |
| 13:20.43 | Samot | The Early Media is setup on *their* INVITE/dial to the destination. |
| 13:22.37 | dnit | Yes I understand that. Lets say I called a number which plays OOS in early for 1min and then sends the 4xx 5xx response. |
| 13:22.46 | Samot | OK |
| 13:22.51 | Samot | That could happen |
| 13:22.56 | Samot | I highly doubt it will. |
| 13:23.06 | dnit | But at my end I dial with a timeout of 45 seconds. |
| 13:23.06 | Samot | They may play it back a couple times... |
| 13:23.15 | Samot | But those messages are 5 seconds. |
| 13:23.32 | Samot | "The number you are dialing is out of service" is like under 10 seconds. |
| 13:23.45 | Samot | Even if they replay it 3 times, it will never go over 30 seconds. |
| 13:23.55 | Samot | This is where you have to start to compromise |
| 13:24.23 | Samot | And if your outbound list is filled with a lot of dead numbers or numbers that just never answer.. |
| 13:24.25 | Samot | Get a better list. |
| 13:24.50 | dnit | In my cases most of the calls keeps repeating the OOS message for more than 45 seonds. |
| 13:25.08 | Samot | How often do you have this issue? |
| 13:25.11 | Asterisco | Samot... i've found the error! |
| 13:25.46 | Samot | How many calls out of X calls get an OOS message that lasts almost a minute? |
| 13:26.09 | dnit | Quite often. The users are dialing are dialing a lot of numbers. and I don't have control over what they are dialing . I just want to tag such numbers as BAD numbers so that they wont attemp them, |
| 13:26.27 | Samot | How are the users dialing |
| 13:26.32 | Samot | What users? |
| 13:26.41 | Samot | you just said you originate these calls via AMI |
| 13:28.04 | Samot | Is there some GUI they are using for this? |
| 13:30.45 | Samot | Where is the Originate command? I asked to see that with the dialplan. Anything that starts the call until it ends. |
| 13:33.14 | dnit | Yes the users are on the web, |
| 13:33.24 | dnit | the system dials using AMI |
| 13:33.25 | Samot | Who are the users? |
| 13:33.34 | Samot | The Agents or the people being called? |
| 13:33.39 | dnit | endusers who are using the product. |
| 13:33.44 | Samot | OK |
| 13:33.51 | Samot | Show the Originate |
| 13:41.35 | dnit | context: inbound, channel : number@outbound; extension: s |
| 13:42.02 | dnit | action: Originate |
| 13:42.18 | dnit | or do you want to see the raw paket? |
| 13:42.24 | Samot | No. |
| 13:42.50 | [TK]D-Fender | <[TK]D-Fender> Are you using the option he told you to use? |
| 13:42.50 | [TK]D-Fender | <[TK]D-Fender> Where's the debug? |
| 13:42.51 | Samot | You should just try the EarlyMedia setting |
| 13:42.54 | [TK]D-Fender | <[TK]D-Fender> Where's the AMI dump? |
| 13:43.02 | [TK]D-Fender | welcome to HALF AN HOUR AGO |
| 13:43.17 | dnit | Ill paste the ami dump |
| 13:44.42 | dnit | manager set debug on |
| 13:44.46 | [TK]D-Fender | The FULL call <-------------- |
| 13:44.50 | dnit | after this right ? |
| 13:45.18 | dnit | Yes Ill put the dump for the whole call |
| 13:49.57 | *** part/#asterisk jgowdy (uid239878@gateway/web/irccloud.com/x-fvamdkeqkhqrgdhq) |
| 13:57.07 | *** join/#asterisk r1ppa (~Rippa@64.26.133.67) |
| 13:58.59 | r1ppa | Can someone nudge me in the right direction, I am trying to control the display on our Polycom phones, I need to try and create a "staff directory" button in the front instead of what it is doing now, reading 00000000-directory.xml and only putting the first name |
| 13:59.52 | Samot | That would be the "global" directory |
| 14:00.11 | Samot | It will replace 00000000 with the MAC |
| 14:00.19 | Samot | If one doesn't exist with the MAC |
| 14:00.21 | igcewieling | r1ppa: r1ppa: what version of firmware? Which model? |
| 14:00.31 | Samot | But it's an XML file. Is it formatted right? |
| 14:00.45 | [TK]D-Fender | correct |
| 14:01.08 | [TK]D-Fender | That directory was not meant for use as a "corporate directory" |
| 14:01.20 | [TK]D-Fender | it is supposed to be user-specific |
| 14:01.30 | r1ppa | the file itself is not the problem, lets say Amber is at the top of the xml directory file, that will be the only thing you will see on the staff direcotry button, Amber....I want that button to delve into the directory |
| 14:01.32 | [TK]D-Fender | so they can control their speed-dials on other ilne keys, etc |
| 14:01.46 | [TK]D-Fender | Corporate should be ANOTHER directory of some sort |
| 14:01.57 | [TK]D-Fender | LDAP / MicroBrowser, etc |
| 14:02.42 | [TK]D-Fender | All of which are in the ADMIN GUIDE. |
| 14:02.42 | [TK]D-Fender | Go get it and look up the terms |
| 14:02.42 | igcewieling | Unless you are running VVXs with 5.2(?) modifying the contact directory files outside of the phone can be a problem. |
| 14:02.57 | igcewieling | leaving you with [TK]D-Fender' |
| 14:03.00 | igcewieling | s suggestion |
| 14:03.19 | r1ppa | yes, READ READ READ, which is fine I am willing to read but cannot find what it is I am looking for |
| 14:03.40 | [TK]D-Fender | <[TK]D-Fender> All of which are in the ADMIN GUIDE. <------------- |
| 14:03.41 | r1ppa | it is not the file itself, it is the display, how it is displayed on the Polycom |
| 14:04.03 | igcewieling | r1ppa: I doubt we understand what you mean. What do you mean by only showing one contact? |
| 14:04.23 | r1ppa | igcewieling, the physical phones display screen buttons |
| 14:04.23 | igcewieling | r1ppa: it is displayed when you press the. Directory button. |
| 14:05.02 | r1ppa | igcewieling, I want to make a directory button instead of it actually opening the directory to 1 button, which will always only show the top staff member |
| 14:05.37 | igcewieling | that should be a simple key assignment |
| 14:05.41 | [TK]D-Fender | yup |
| 14:06.13 | r1ppa | well, thats where I am stuck |
| 14:06.13 | [TK]D-Fender | EFK <---- |
| 14:06.13 | igcewieling | You would get any help with that from me. I HATE polycom custom key setup. |
| 14:06.22 | igcewieling | You would NOT get any help from me. |
| 14:06.23 | r1ppa | indeed |
| 14:06.27 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 14:06.27 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:06.33 | [TK]D-Fender | "which will always only show the top staff member" <- what is a "top staff member"? How is that determined? |
| 14:06.47 | [TK]D-Fender | Clarify these vague concepts you're introducing |
| 14:07.21 | r1ppa | directory.xml file is being READ to the display, and only shows the first user/member, whatever you want to call them. So there is a button on the phone that shows AMBER, Aida, etc, and it calls them |
| 14:07.22 | igcewieling | [TK]D-Fender: he means unassigned line keys showing the first x directoy entries. |
| 14:07.31 | *** join/#asterisk kharwell (kharwell@nat/digium/x-dubwulrlndjpirds) |
| 14:07.31 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:07.57 | r1ppa | I want DIRECTORY, then it delves into the actual directory, or change the soft key from reading directory to showing a "Voicemail" shortcut |
| 14:08.03 | igcewieling | r1ppa: if you assign something to that key it won'y default to showing the directory entry. |
| 14:08.05 | [TK]D-Fender | Directory DOES delve in |
| 14:08.14 | [TK]D-Fender | You only SEE one at a time maybe, that's what scrolling is for... |
| 14:08.35 | r1ppa | nah this is not something we can scroll, sorry not explaining this well |
| 14:08.40 | [TK]D-Fender | Indeed |
| 14:08.43 | r1ppa | I want to manually control my soft keys |
| 14:08.49 | [TK]D-Fender | Still vauge |
| 14:08.53 | [TK]D-Fender | vague* |
| 14:08.57 | r1ppa | ZZzz |
| 14:09.14 | igcewieling | r1ppa: you want a key open the contact directory. |
| 14:09.18 | [TK]D-Fender | What you'll need to look at is a FULL MicroBrowser application |
| 14:09.41 | [TK]D-Fender | You're WAY past just EFK to shove a button to start the process |
| 14:09.46 | r1ppa | igcewieling, I just want to start controlling the soft keys as well |
| 14:10.05 | igcewieling | r1ppa: stop complicating things. |
| 14:10.21 | igcewieling | "start controlling the softkeys" is useless. |
| 14:10.21 | [TK]D-Fender | There is no "I just". MicroBrowser. That's what you've got. |
| 14:10.25 | r1ppa | igcewieling, I am trying to ask questions, lol |
| 14:10.32 | [TK]D-Fender | There is no "I get full control of every keypress outside of that |
| 14:10.47 | [TK]D-Fender | Go look it up and start there |
| 14:11.03 | r1ppa | I did, not finding what I need because prolly using bad terms |
| 14:11.09 | r1ppa | anyway thanks guys! I will keep trying |
| 14:11.19 | [TK]D-Fender | And you're not confirming what you found.... which doesn't tell us much |
| 14:11.27 | r1ppa | NADA |
| 14:11.32 | r1ppa | thats why I am here |
| 14:11.37 | [TK]D-Fender | All we have are "probably" and "didn't find". |
| 14:11.42 | [TK]D-Fender | We don't know what you tried |
| 14:11.46 | [TK]D-Fender | we don't know where you looked |
| 14:12.14 | [TK]D-Fender | Don't be vague about what you actually TRIED. |
| 14:14.29 | igcewieling | Maybe I missed answer to my question about models and firmware versions. |
| 14:14.42 | [TK]D-Fender | We have no information of any kind |
| 14:15.16 | [TK]D-Fender | No point in drilling further |
| 14:16.03 | igcewieling | We could let him try stuff which won't work on the specific firmwre or phone he is working with. |
| 14:18.16 | igcewieling | poor customer "The client says a dump truck hit the telephone pole for a 2nd week in a row knocking out their phone lines," |
| 14:18.26 | [TK]D-Fender | Well we did ask. We never got an answer. This is consistent at least... |
| 14:19.01 | Samot | Hey now. |
| 14:19.14 | Samot | Just help. |
| 14:19.16 | Samot | Don't be whining about needed stuff to help, just do it. |
| 14:19.58 | [TK]D-Fender | This isn't Nike :p |
| 14:20.39 | Samot | True. |
| 14:20.44 | [TK]D-Fender | Can't tell the doctor where it hurts? Well I guess we'll just start AMPUTATING till they grow a clue. |
| 14:20.56 | Samot | It's more like Jordan in minor baseball. |
| 14:21.17 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:21.17 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:21.51 | [TK]D-Fender | This is a telecommunications support channel ... you'd better come in capable of clearly communicating. |
| 14:21.51 | Samot | Silliness. |
| 14:24.19 | dnit | Samot , [TK]D-Fender I have got a 20k lines dump for one call :( |
| 14:25.04 | dnit | ami dump |
| 14:25.22 | Samot | How is that even possible? |
| 14:29.56 | dnit | I dont know how, but I see all the events for the same call. |
| 14:30.24 | dnit | Same channel is mentioned in almost every event. |
| 14:31.37 | Samot | Well... |
| 14:31.51 | Samot | If the end user is going to a website.. |
| 14:32.33 | Samot | Putting their number in for a callback AND you are getting a _high_ amount of OOS or bad numbers.... |
| 14:33.19 | Samot | I'd look at your method of allowing people to randomly input numbers. |
| 14:33.19 | Samot | I find it hard to believe a user wanting a callback would be in a bad number. |
| 14:33.41 | dnit | Users are not calling themselves they are calling other prospects. |
| 14:33.45 | dnit | for sales. |
| 14:35.38 | Samot | So Party A inputs a number.. |
| 14:35.38 | Samot | That calls Party B |
| 14:35.38 | Samot | But then agents get involved? |
| 14:35.38 | dnit | Yes kind of . list of numbers |
| 14:35.38 | Samot | Or is Party A the agent? |
| 14:35.38 | dnit | No agents are different |
| 14:35.38 | Samot | OK |
| 14:35.38 | Samot | so now there are THREE parties involved in this? |
| 14:35.38 | dnit | Actually in terms of * users and agents are both agents. |
| 14:35.38 | dnit | Yes 3 parties |
| 14:35.38 | Samot | So you have people entering numbers from a call ist |
| 14:35.38 | Samot | So you have people entering numbers from a call list |
| 14:35.49 | Samot | That call people and then route them to an agent... |
| 14:35.50 | dnit | agents facilitate the process of calling peple. They make sure all the gatekeepers, sercretariesa re eliminated and end user is connected to end user, |
| 14:36.24 | Samot | OK,I said it before. |
| 14:36.29 | Samot | Get better lists. |
| 14:37.06 | Samot | You're trying to put a solution in place because your source data is highly incorrect. |
| 14:37.06 | dnit | Its not upto me to make the lists. |
| 14:37.06 | dnit | Its not my data. Users have teir own data which they load in our syste, |
| 14:37.06 | Samot | Well then.. |
| 14:37.06 | dnit | Its not my data. Users have teir own data which they load in our system |
| 14:38.43 | Samot | I would employ the user of a CLID lookup service. |
| 14:38.43 | Samot | s/user/use/ |
| 14:38.43 | Samot | This sounds more like an issue with a lack of validation processes. |
| 14:38.43 | *** join/#asterisk TandyUK2 (~admin@2a02:13a0:a006:1:5dda:f105:5272:2a5a) |
| 14:38.43 | Samot | There are services out there that you can make API calls to and get the details of the number. |
| 14:38.56 | *** join/#asterisk bl3nto (~bl3nto@dh207-88-219.xnet.hr) |
| 14:39.03 | Samot | If it's mobile, landline and some will tell you if it is in service. |
| 14:40.33 | dnit | Their data could be wrong too. Right. Lets a number just went out of service. |
| 14:40.33 | Samot | JFC |
| 14:40.33 | Samot | Come on. |
| 14:40.33 | dnit | It would take some time to update it. |
| 14:40.33 | Samot | You have bad data. |
| 14:40.33 | Samot | OK |
| 14:40.33 | Samot | So? |
| 14:40.33 | Samot | The error margin is much lower. |
| 14:41.50 | Samot | Instead of having a list of 60% bad numbers |
| 14:41.50 | Samot | You may only have 10% |
| 14:41.50 | Samot | BIG difference |
| 14:41.50 | Samot | And you can tag that number as BAD immediately |
| 14:41.50 | Samot | Also, there are rules about making unsolicited calls to mobile numbers via autodialers. |
| 14:41.50 | dnit | Yes it is. But agagin I can't tag those 10%. They will keep rotating. It is one of the alternatives, but not what I am looking for as of now. |
| 14:41.55 | Samot | No.. |
| 14:41.59 | Samot | You catch those AFTER the call |
| 14:42.20 | Samot | When it gives you a status code of the call being bad. |
| 14:42.43 | Samot | You stop them before the call happens by knowing if they are valid numbers or not |
| 14:42.43 | Samot | And those that get through, you have to catch when the call returns the error. |
| 14:43.41 | Samot | If I dump a list of 500 numbers to you |
| 14:43.41 | Samot | And 300 of them are bad |
| 14:43.47 | Samot | You could probably catch all 300 with the CLID lookup or the majority |
| 14:44.04 | Samot | Those that get through, get dialed but when they return the bad status, you tag them THEN. |
| 14:44.24 | Samot | So the next time I dump that list |
| 14:44.32 | Samot | Only 200 or so calls are made |
| 14:44.52 | Samot | Because you've flagged the rest of the numbers alread. |
| 14:44.52 | Samot | Because you've flagged the rest of the numbers already. |
| 14:44.54 | *** join/#asterisk TandyUK (~admin@2a02:13a0:a006:1:5dda:f105:5272:2a5a) |
| 14:47.23 | dnit | I am already filtering the bad numbers which are not callable. Its just the numbers which gets through and doesnt return bad status( for which you are suggesting to use CLID lookup service). |
| 14:47.43 | Samot | Are these numbers that timeout? |
| 14:47.47 | Samot | No Answer? |
| 14:47.47 | dnit | Yes |
| 14:48.00 | Samot | Not playing back an OOS message. |
| 14:48.15 | dnit | Time out. after playing OOS message |
| 14:48.18 | Samot | No. |
| 14:48.22 | Samot | Those are bad numbers. |
| 14:48.56 | dnit | Yes I know that. But the system doesn't. |
| 14:48.56 | Samot | Listen |
| 14:49.03 | Samot | If they are playing back OOS messages longer than 45 seconds... |
| 14:49.13 | Samot | And your Dial() timeout is 45 seconds... |
| 14:49.22 | Samot | Then Asterisk counts those as timeouts |
| 14:49.33 | Samot | Because the DIAL timed out |
| 14:49.44 | Samot | So you are getting a false reading |
| 14:49.45 | dnit | Yes. |
| 14:51.03 | dnit | For this purpose only I want the agents to hear to such calls' early media and tag them as Out of service. |
| 14:51.04 | dnit | Is something wrong with this approach ? |
| 14:51.23 | Samot | Yeah but Originate doesn't send the call to the context value unless the call is ANSWERED |
| 14:51.48 | dnit | Exactly. That is why I used Bridge to connect agent to the outgoing call. |
| 14:51.48 | Samot | If I dial 1NXXNXXXXXX@outbound and the context is "inbound" |
| 14:51.55 | Samot | The Outbound has to answer. |
| 14:52.14 | dnit | But I can bridge it before it answered right . |
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| 14:52.36 | Samot | No. |
| 14:52.36 | dnit | But I can bridge it before the channel is answered right . |
| 14:52.36 | Samot | It doesn't go to the inbound context unless the call is answered |
| 14:52.38 | Samot | That's how Originate works. |
| 14:53.03 | dnit | Yes I understand it won't fall to the context. |
| 14:53.03 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 14:54.12 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 14:54.13 | Samot | OK |
| 14:54.42 | Samot | so unless there is a 200 OK from the PSTN on the call... |
| 14:54.42 | Samot | The Dial() in the outbound context is going to fail... |
| 14:54.42 | dnit | But I can bridge two channels. right. So I bridged the outgoing channel ( on which the early media is playing right now) and the agents channel ( on which the agent is waiting for a call). |
| 14:55.09 | Samot | So you have to dial them both |
| 14:55.10 | Samot | At the same time |
| 14:55.10 | dnit | So this way agent can hear the early media of the outbound channel |
| 14:55.13 | Samot | Or you can use the EarlyMedia setting |
| 14:55.21 | Samot | That I have mentioned numerous times. |
| 14:55.40 | Samot | But since you're calling the END USER first.. |
| 14:55.46 | Samot | It's really pointless... |
| 14:56.13 | Samot | Because you would need to dial the end user and the agent at the same time and then bridge them |
| 14:57.34 | dnit | Yes exactly. |
| 14:57.34 | Samot | Really, it sounds like you need a voip switch/sip router. |
| 14:58.34 | lorsungcu | Samot: I'm disappointed. |
| 14:58.46 | lorsungcu | That only took three hours to get to |
| 14:58.46 | dnit | I have a sip router in front of my asterisk. |
| 14:58.56 | Samot | .... |
| 14:59.06 | dnit | I can't see how it can help me |
| 14:59.06 | Samot | Well.. |
| 14:59.26 | Samot | How many times does a 183 Session in Progress reply result in an OOS message? |
| 15:00.08 | dnit | Not always. But there are considerable amount of such calls. |
| 15:00.08 | Samot | 10 out of 10? |
| 15:00.14 | Samot | How many? |
| 15:00.17 | Samot | What is the ratio? |
| 15:00.22 | Samot | Is it high or low? |
| 15:00.22 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 15:00.29 | dnit | I had 800 such calls for 1 week |
| 15:00.38 | Samot | Out of? |
| 15:00.40 | dnit | 5 working days |
| 15:00.47 | Samot | Out of how many calls? |
| 15:01.33 | Samot | 2000? |
| 15:01.33 | Samot | 5000? |
| 15:01.33 | dnit | Out of 2k calls that a user dialled. |
| 15:01.33 | dnit | I havent even analyzed it for other users. |
| 15:01.50 | Samot | So roughly 40% |
| 15:02.03 | Samot | You mean you had 800 OOS calls? |
| 15:02.05 | dnit | Yes for one user. |
| 15:02.09 | dnit | Yes |
| 15:02.21 | Samot | K |
| 15:02.53 | Samot | How many other calls respond with a 183 and go through? |
| 15:02.53 | Samot | That's what I'm asking. |
| 15:02.56 | Samot | When you get a 183 in a call... |
| 15:03.04 | dnit | You mean they had proper signalling ? |
| 15:03.15 | Samot | OK |
| 15:03.32 | Samot | You send an INVITE to your carrier |
| 15:03.41 | Samot | You're going to get back 100 Trying |
| 15:03.41 | dnit | Every call gets a 183. |
| 15:03.41 | dnit | Yes |
| 15:03.41 | Samot | Which should be followed by a 180 Ringing |
| 15:03.42 | Samot | OK |
| 15:03.47 | dnit | I know the flow. |
| 15:04.33 | Samot | You know what... |
| 15:04.37 | Samot | You're right. |
| 15:05.03 | Samot | Every call will get a 183 Session in Progress. |
| 15:05.04 | Samot | Because your carrier is sending back Early Media so you can hear the call progress. |
| 15:05.09 | Samot | Like Ringing |
| 15:05.19 | dnit | Only after SDP we can et early media |
| 15:05.21 | dnit | Yes |
| 15:05.24 | Samot | OK |
| 15:05.47 | Samot | 100 Trying can be followed by a 180 or 183 |
| 15:05.47 | Samot | It has to be one of the other |
| 15:06.32 | Samot | But 180 cannot follow 183 |
| 15:06.32 | Samot | However... |
| 15:06.32 | dnit | Yes and later a 200 OK which indicates the phone has been picked up. |
| 15:06.32 | Samot | Right |
| 15:06.32 | Samot | So you have to wait for that response. |
| 15:06.32 | Samot | If it is 45 seconds, it's 45 seconds. |
| 15:08.06 | dnit | Yes if its 45 seconds. * teminates the call by sending CANCEL request. |
| 15:08.06 | Samot | So it is always a timeout |
| 15:08.25 | dnit | Which is followed by a 487 RT response |
| 15:08.39 | Samot | Yes |
| 15:08.39 | dnit | Yes for the system it is a timeout. |
| 15:08.39 | Samot | Because that's the proper response to a CANCEL |
| 15:08.40 | Samot | OK |
| 15:09.05 | Samot | The SIP Router... |
| 15:09.26 | Samot | Can actually watch for those replies... |
| 15:09.26 | Samot | And act one them.. |
| 15:09.26 | Samot | And act on them.. |
| 15:09.51 | Samot | But setting the timeout to 45 seconds means everything always timesout |
| 15:10.10 | Samot | At this point they could be valid numbers with no answering machine |
| 15:10.30 | Samot | So they will ring until someone higher up in the chain ends the call. |
| 15:11.15 | dnit | Yes. This is true for valid numbers whose control is being passed to the end users. |
| 15:11.24 | dnit | by the carriers. |
| 15:11.43 | Samot | Right but if they don't have something connected or in place to answer the call in X time |
| 15:11.46 | Samot | Like voicemail |
| 15:11.54 | Samot | The call will continue to ring. |
| 15:12.01 | Samot | Therefore if you are timing out at 45 seconds. |
| 15:12.11 | Samot | Asterisk is reporting a timeout for the DIAL on a valid number. |
| 15:12.34 | Samot | If you are taking that timeout status as gospel, you've flagged a valid number as bad. |
| 15:13.27 | Samot | Numbers that can't be routed, that are busy, or return some other status.. |
| 15:13.34 | Samot | Will do it in under 30 seconds. |
| 15:13.41 | Samot | Mainly in under 10 or so. |
| 15:14.00 | Samot | Those that are routed will make it to the other carrier.. |
| 15:14.09 | Samot | And have to wait for that carrier to respond... |
| 15:15.03 | Samot | Also.. |
| 15:15.04 | Samot | What do the CDRs show for calls that have OOS messages? |
| 15:15.09 | Samot | Do they show as timeouts? |
| 15:15.21 | Samot | Or do some show as answered? |
| 15:15.38 | Samot | Because some providers do not use early media for this |
| 15:15.50 | Samot | And will drop you on a line with a recorded message. |
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| 15:16.20 | dnit | No they show don't show answered. |
| 15:16.45 | Samot | OK |
| 15:16.45 | Samot | So timeout? |
| 15:16.49 | dnit | Most of the carriers are just passign the early media unless * hangups after 45 seonds. |
| 15:16.58 | dnit | terminates* |
| 15:17.09 | Samot | They are passing the early media so you can hear the other side |
| 15:17.17 | Samot | Like ringing |
| 15:17.38 | dnit | Yes. |
| 15:17.38 | Samot | Plain ole ringing |
| 15:17.38 | Samot | OK |
| 15:17.38 | Samot | My point was... |
| 15:17.54 | Samot | That a bad number could be answered by the final route before the end user |
| 15:18.05 | Samot | And could playback an OOS message. |
| 15:18.12 | Samot | You know how many small voip providers do that? |
| 15:18.14 | Samot | A LOT |
| 15:18.24 | Samot | So for you, the call could look answered. |
| 15:19.16 | dnit | Yes. Maybe thats why out of my 5 carriers I get this behaviour on 4. And the 5th one my CEO has decided to blacklist. |
| 15:19.31 | Samot | Your logic is counting on the destination carrier being the only stop |
| 15:19.41 | Samot | When there could be 3 more |
| 15:20.04 | Samot | The destination carrier could be Onvoy who has sold that DID to Vitelity |
| 15:20.21 | Samot | Who in turn as sold that DID to a voip provider who may sold it to an end user |
| 15:20.27 | Samot | Or another reseller/wholesaler |
| 15:20.45 | Samot | At any point during that chain someone can answer that call and playback a message. |
| 15:21.15 | dnit | Yes. But when this early media is played with OOS message it is evident that the end user is not going to pick the call right. |
| 15:21.24 | Samot | You're not understanding. |
| 15:21.31 | Samot | You always have early media. |
| 15:22.26 | Samot | You will always have a 183 followed by something |
| 15:22.30 | dnit | Yes I understand that we always have the early media. But its mostly ringing / null rtp. |
| 15:22.33 | Samot | 200 OK, 404, 503... |
| 15:22.37 | Samot | Right |
| 15:22.54 | Samot | I'm just pointing out where a false positive can happen |
| 15:23.02 | Samot | In both cases. |
| 15:23.04 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 15:23.15 | Samot | An OOS message played back could show as Answered. |
| 15:23.51 | Samot | A valid number with no voicemail will show as Timeout |
| 15:23.53 | dnit | Yes. |
| 15:24.08 | Samot | You should just dial for 120 and let things take their course |
| 15:24.18 | Samot | Specially if you have a SIP router in front of everything. |
| 15:24.31 | Samot | It will handle failures and look at the actual codes for them. |
| 15:24.47 | Samot | Real failures will happen almost immediately |
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| 16:04.46 | igcewieling | Vitelity was bought by Onvoy a while ago. |
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| 16:24.25 | Samot | The logic is still the same. |
| 16:24.30 | Samot | Replace Onvoy with Broadband or L3 |
| 16:25.42 | Samot | Take you... |
| 16:26.23 | Samot | Just because the number they dial belongs to Verizon or whoever else you use, means that Verizon is not the final stop on the routing of that number. |
| 16:26.59 | Samot | And if you resell that number to a wholesale or whatnot |
| 16:27.09 | Samot | Then you're probably not the last stop on that route... |
| 16:34.11 | igcewieling | signaling usually goes via the carrier. RTP could be from anywhere. |
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| 17:50.38 | elcontrastador | I have a compressed timeline to turn up a client Asterisk server to replace their Cisco CCM/Unity servers. They have hundreds of ancient Cisco 7940 and 7960 handsets that we're going to user chan-sccp-b with. Looking for someone with production experience with chan-sccp-b. I'd like to contract with someone to consult with me and provide some initial best practices and initial configurations. I will pay via paypal or venmo, |
| 17:50.38 | elcontrastador | <PROTECTED> |
| 17:52.42 | Samot | hears crickets |
| 17:55.11 | igcewieling | That sounds like a recipe for disaster. |
| 17:55.22 | elcontrastador | please dont say that :-) |
| 17:55.36 | igcewieling | You won't find a lot of SCCP users here (no sane people use them with Asterisk). |
| 17:56.05 | elcontrastador | I've read chan-sccp is pretty robust. |
| 17:56.35 | igcewieling | Glad to hear it. Still wouldn't touch SCCP without a programmer on staff. |
| 17:56.41 | Samot | In like 2007. |
| 17:56.46 | elcontrastador | lol |
| 17:56.51 | elcontrastador | lol |
| 17:57.44 | Samot | I wouldn't touch SCCP even with a programmer on staff. |
| 17:57.46 | elcontrastador | Samot have you deployed it before? |
| 17:58.21 | Samot | Outside of a lab? No. |
| 17:58.32 | Samot | I mainly replace those systems. |
| 17:59.01 | igcewieling | I tried it about 10 years ago. Gave up when I realized Cisco actively discourages people from using their 79xx phones with anything except CCM. |
| 17:59.13 | Samot | Funny how that is. |
| 17:59.45 | Samot | Proprietary PBX maker only wants their proprietary phones to work with their system. |
| 18:00.34 | elcontrastador | Yeah, I ran it many years ago as a small lab test...worked. Also, I re-imaged our whole office of cisco phones with their sip image and ran those for several years. |
| 18:00.39 | igcewieling | I *like* the idea of dumb phones like in SCCP or MGCP .vs. the idea of each phone being a mini-PBX like in SIP. The problem is SIP won, so it is rather silly to try and fight that. |
| 18:01.25 | Samot | I'm not sure how a phone is a mini pbx. |
| 18:01.25 | Samot | I mean there are devices that have those features but SIP phones generally don't |
| 18:01.36 | elcontrastador | not that silly as interim solution when there's nearly 1000 handsets to replace...give them an option to migrate away from sccp to sip with new phones |
| 18:01.56 | Samot | That should have been part of the PBX replacement discussion |
| 18:02.41 | elcontrastador | yeah, the ccm's are failing... they want to buy other than cisco... |
| 18:03.04 | Samot | Along with "You can churn those phones on Ebay for the tune of about $7,500 USD" |
| 18:03.10 | elcontrastador | will take them a few years to sawp out their pones |
| 18:03.42 | Samot | Well enjoy a few years in hell. |
| 18:03.44 | elcontrastador | lol |
| 18:03.49 | Samot | I'm not kidding. |
| 18:03.54 | elcontrastador | joy |
| 18:04.26 | Samot | Like I said.. |
| 18:04.30 | elcontrastador | what, in particular, is the biggest pain point in your opinion |
| 18:05.01 | Samot | $7,500 for 1000 phones |
| 18:05.01 | Samot | If you get a good buyer. |
| 18:05.01 | Samot | That's how much those phones are worth these days |
| 18:05.01 | Samot | Less than $8 per |
| 18:05.01 | elcontrastador | doesnt matter how much theyre worth...it's the replacement cost of new sip phones |
| 18:05.06 | Samot | True. |
| 18:05.23 | Samot | But then again.. |
| 18:05.35 | Samot | The costs of making the current phones work and keeping them working... |
| 18:05.46 | Samot | Probably going to work out to the same. |
| 18:06.21 | Samot | Or more. |
| 18:06.21 | Samot | And you call any phone seller... |
| 18:06.23 | Samot | Tell them you want 1K phones today.. |
| 18:07.15 | Samot | You will get a discount. |
| 18:08.08 | Samot | Yes, there will be upfront costs... |
| 18:08.08 | elcontrastador | They dont have a 150k right now... |
| 18:08.08 | Samot | But compared to the long term expenses of supporting those phones... |
| 18:08.08 | Samot | Finance. |
| 18:08.08 | elcontrastador | wont be long term |
| 18:08.08 | Samot | A few years |
| 18:08.08 | Samot | Is long term. |
| 18:08.08 | elcontrastador | y |
| 18:08.17 | *** join/#asterisk [NC] (~NC@208.94.241.140) |
| 18:08.33 | Samot | And I'm not sure were you get 150K |
| 18:10.40 | Samot | Depending on the maker, the model and what kind of bulk deal you can cut, I doubt this would be over 6 figures. |
| 18:10.55 | Samot | ANYTHING is an improvement to those 79XX |
| 18:11.52 | Samot | Look at Grandstream, Yealink, Aastra |
| 18:11.52 | Samot | You can probably get two phones for $150 |
| 18:12.30 | Samot | So reduces your estimate already by 50% plus or minus. |
| 18:13.23 | Samot | Now you want to drop some Polycoms in there and more the mid to high level models, then you're looking at 6 figures |
| 18:14.21 | Samot | But 4 and 6 line phones...depending on the requirements under $100 per |
| 18:14.21 | Samot | And the maker. |
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| 18:24.15 | elcontrastador | yeah, i'm aware of the these phones. They want to get new high quality handsets, like Digium D80 and D65, when they get funds approved. |
| 18:24.45 | elcontrastador | prob do many phases not all at once |
| 18:26.01 | RedHatter | Hey all! I'm trying to originate a call from the cli but it's not working. When I run "channel originate SIP/xxxxxxxxxx@twilio-trunk application Playback echo-instruct" I get "No such command 'channel originate ... '". Any ideas what could be wrong?" |
| 18:26.21 | lorsungcu | elcontrastador: I've done similar |
| 18:26.34 | lorsungcu | With roughly 500 handsets |
| 18:26.38 | lorsungcu | Using sip |
| 18:27.06 | lorsungcu | All at once. Customer was sure they wanted it despite the shortcomings |
| 18:27.23 | lorsungcu | They came back in 2 years and bought handsets |
| 18:27.52 | lorsungcu | I'd really advise against going through with it |
| 18:28.40 | lorsungcu | You can easily get plenty workable yealink or something for about 30-35$ at that volume |
| 18:28.51 | elcontrastador | lorsungcu you flashed them all to SIP? |
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| 18:28.54 | lorsungcu | Yes |
| 18:29.04 | elcontrastador | ugh...that alone is a big ordeal |
| 18:29.07 | lorsungcu | That part was the easy bit. |
| 18:29.29 | lorsungcu | Seriously, sell them all in bulk and buy cheap Chinese handsets |
| 18:29.41 | wabbits | I wonder if a large number of these old phones can be replaced with a softphone on the pc that is likely already in front of the user? |
| 18:30.03 | elcontrastador | wabbits...yeah, that softphone aint happening |
| 18:30.34 | lorsungcu | elcontrastador: might even find digium or sangona handets for around 50. I haven't sold the digium stuffing a while so I can't say for sure |
| 18:30.40 | wabbits | elcontrastador why not? |
| 18:30.45 | lorsungcu | But you COULD do this for far less |
| 18:31.06 | elcontrastador | I guess I'm struggling with what the exact pain point of using sccp is...i'm hearing you all that it's a bitch...just wondering exactly the biggest pitfall you've all found |
| 18:31.06 | lorsungcu | wabbits: because softphones are miserable |
| 18:31.33 | lorsungcu | elcontrastador: it isn't supported or developed. So as igcewieling said, hire someone that can maintain and troubleshoot the driver |
| 18:31.59 | elcontrastador | softphones on teachers laptops...support nightmare...teachers and their 4GB windows user profiles and 4GB of ram laptops, etc....screw that |
| 18:32.41 | lorsungcu | Yes softphones are no good. |
| 18:39.03 | igcewieling | elcontrastador: try it and see. |
| 18:39.29 | igcewieling | I'm sure you can find a few phones to pull off the current system to test with Asterisk. |
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| 18:52.40 | elcontrastador | I have a lab of about 12 phones in my office right now...getting it all set up. |
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| 19:54.33 | [TK]D-Fender | RedHatter, syntax for CLI changes. hit TAB and start looking at the other keywords |
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| 23:52.18 | Katty | I came to wish qwell a happy bday but he's not here )= |