| 01:02.01 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:02.01 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 03:30.55 | cleron | im trying to get my astrisk based dailer to connect to my pbx but i get the following error. "Spawn extension (ib_dial, failed, 3) exited non-zero on 'OutgoingSpoolFailed'" anyone know why? |
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| 06:56.09 | cleron | im trying to get my astrisk based dailer to connect to my pbx but i get the following error. "Spawn extension (ib_dial, failed, 3) exited non-zero on 'OutgoingSpoolFailed'" anyone know why? |
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| 11:43.41 | Samot | cleron: you need to show a whole call |
| 11:44.41 | wyoung | Samot: hey bro |
| 11:54.14 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 11:55.24 | Samot | ?? |
| 11:55.43 | wyoung | Samot: sup? |
| 11:55.50 | wyoung | I just joined |
| 11:55.55 | wyoung | I am seeing how you are going |
| 11:57.32 | file | moo |
| 12:15.57 | *** join/#asterisk avb (~avb@45.55.203.176) |
| 12:17.21 | wyoung | file: is your first name peter? |
| 12:17.27 | file | no. |
| 12:17.37 | wyoung | Good to know |
| 12:19.56 | file | https://blogs.asterisk.org/author/jcolp/ |
| 12:19.59 | file | that's me. |
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| 12:45.30 | avb | file: omg, you are the guy behind webrtc :) |
| 12:45.45 | file | I work on it in Asterisk, yes. |
| 12:47.34 | avb | file: so finally high resolution video will be possible in asterisk15? |
| 12:47.43 | avb | its a pleasure to meet you :) |
| 12:47.53 | file | it's been possible for awhile, it's all in how the attributes are negotiated between each side |
| 12:48.39 | avb | i spent a year fighting with high res video on asterisk13 and its been a huge pain |
| 12:49.10 | avb | freezing video were my daily nighmares :) |
| 12:49.22 | file | that's due to dropped packets |
| 12:49.27 | avb | and the lack of nacks |
| 12:49.31 | avb | yeh |
| 12:49.55 | file | NACK support is partially done on one side, it's being worked on for ingress traffic |
| 12:50.09 | avb | were ought to move to webrtc but im not losing a hope to go back to asterisk |
| 12:50.22 | file | REMB is also done, so bandwidth conditions will alter encoding quality |
| 12:50.34 | avb | is that in asterisk15? |
| 12:50.39 | file | yes. |
| 12:50.44 | avb | nice |
| 12:50.45 | file | requires PJSIP. |
| 12:51.14 | avb | thats sounds very promision |
| 12:51.16 | avb | thats sounds very promising |
| 12:51.45 | avb | ill play with it on a test server again |
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| 12:52.07 | avb | we just finished redoing all mobile apps and web to webrtc :( |
| 12:52.22 | avb | but my clients hates to have audio and video separated |
| 12:52.29 | avb | client* |
| 13:26.13 | igcewieling | "Thank you for your assistance, the phone lines are bouncing perfectly." |
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| 14:31.58 | WizJin | I am getting response "Temporary Unavailable" from my side |
| 14:32.02 | WizJin | from my server |
| 14:32.11 | WizJin | in Sip DEBUG On log |
| 14:32.46 | Samot | For? |
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| 14:35.18 | [TK]D-Fender | How can you get a response from YOUR side? |
| 14:36.39 | [TK]D-Fender | Show us the call. I don't think I've seen * generate that kind of response on its own before. |
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| 14:38.36 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:38.56 | Samot | It doesn't. |
| 14:39.21 | Samot | Outside of when it dials to Dial() a peer that doesn't show up. |
| 14:39.25 | Samot | Or is UNREACHABLE |
| 14:40.43 | Samot | WizJin: You need to state your issue more clearly. |
| 14:41.11 | WizJin | wait |
| 14:43.13 | [TK]D-Fender | Better yet, show us. |
| 14:43.28 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:43.36 | [TK]D-Fender | Actual debug is better than a poor description of something we can't judge for ourselves |
| 14:49.14 | WizJin | ok |
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| 14:55.52 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:56.03 | WizJin | https://paste.pics/36IE1 |
| 14:57.19 | WizJin | The setup basically is like, i have a softswitch (Vos3000), i have created a phone in it, i can successfully call from my eyebeam but i have configured the same phone in dialer, and it is giving this error. When i do "Sip show registry" the trunk is registered.. the trunk name is trunkme1 as you can see in the error logs |
| 15:04.25 | Samot | No. |
| 15:04.30 | Samot | Do not give us screen shots. |
| 15:04.40 | Samot | Give us actual logs in a pastebin |
| 15:18.41 | *** join/#asterisk WizJin (~WizJin@202.91.72.44) |
| 15:18.50 | WizJin | samot, its from the putty, so its hard for me to copy |
| 15:18.53 | WizJin | whats the command :? |
| 15:19.10 | Samot | Highlighting it copies it |
| 15:19.21 | Samot | Or clicking on the top bar and select all |
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| 15:23.44 | WizJin | https://pastebin.com/pX8iHNXA |
| 15:23.46 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:23.50 | WizJin | paste |
| 15:24.11 | WizJin | trunkme1 is the carrier i have configured in sip.conf. 6762 is the prefix |
| 15:27.35 | Samot | [May 30 11:04:20] -- Executing [676219022030996@default:1] Dial("Local/676219022030996@default-af7d,2", "SIP/trunkme1/19022030996||To") in new stack |
| 15:27.46 | Samot | Why do you have |'s in that? |
| 15:27.56 | Samot | That's completely wrong. |
| 15:28.23 | Samot | If you're trying to set the TO and FROM headers it's ! |
| 15:28.53 | Samot | An exclamation point. Further it's !touser@todomain.com!fromuser@fromdomain |
| 15:29.09 | Samot | It needs to be a VALID SIP header. |
| 15:29.14 | Samot | And SIP URI |
| 15:29.20 | Samot | "To" is not going to cut it. |
| 15:30.02 | Samot | [May 30 11:04:20] -- Got SIP response 480 "Temporarily Unavailable" back from 209.126.79.141 |
| 15:30.27 | Samot | ^^ So where ever you are Dial()ing, they are returning that probably because they can't find a valid user or something else. |
| 15:30.37 | Samot | So go look on that system as to why a 480 was returned. |
| 15:31.51 | *** join/#asterisk WizJin (~WizJin@123.63.213.217) |
| 15:45.29 | [TK]D-Fender | I'd be betting it's because you're flooding them with calls |
| 15:46.02 | [TK]D-Fender | more or faster than they allow |
| 15:47.18 | Samot | Well the Dial() string is wrong. |
| 15:47.36 | Samot | There are no pipes ( | ) in the Dial() string. |
| 15:48.01 | [TK]D-Fender | [May 30 11:04:20] -- Called trunkme1/19022030987 |
| 15:48.05 | Samot | OK |
| 15:48.08 | Samot | It called the line |
| 15:48.14 | [TK]D-Fender | and * didn't complain |
| 15:48.15 | WizJin | hmm |
| 15:48.17 | Samot | It doesn't set the TO or FROM header |
| 15:48.25 | [TK]D-Fender | I wonder if we're looking an an ancient version of * here |
| 15:48.27 | Samot | Because it doesn't see a !! |
| 15:49.51 | WizJin | hmm |
| 15:50.01 | [TK]D-Fender | we shold also be loking at SIP debug |
| 15:50.23 | [TK]D-Fender | just for a clear picture of the actual full formatting |
| 15:50.52 | [TK]D-Fender | because these are originated somehow (betting on call files) and I'm betting there is plenty more not being set in a good way |
| 15:51.09 | [TK]D-Fender | But for the rejection itself, my money is still on it rejecting your flooding |
| 15:52.31 | Samot | Well either it's hitting the desired destination and it's being rejected. |
| 15:52.49 | Samot | Or someone in the path is blocking and sending that back |
| 15:53.09 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 15:54.49 | [TK]D-Fender | WizJin, So who is "trunkme1" exactly and what service is it supposed to provide you? |
| 15:55.13 | WizJin | [TK]D-Fender, wait, i am making a detailed post with sip debug |
| 15:56.09 | WizJin | trunkme1 is a outbound calling plan i have purchased from a provider that specializes in call center termination.. it was mentioned in my communication that i need high cps and concurrent calls |
| 15:57.02 | WizJin | after that, i configured a softswitch (Vos3000) so i can use it for billing purposes, i created phone there, called 1001, which logins from eyebeam perfectly and calls are also going through to canada |
| 15:57.33 | WizJin | now same username password i am using to configure [trunkme1] in sip.conf, so i can channel my dialer calls through the extension 1001 in vos |
| 15:57.39 | WizJin | but somehow it doesn't go through |
| 15:58.11 | Samot | Show the SIP Debug |
| 15:58.55 | WizJin | sip set debug on is the command right ? |
| 15:59.38 | Samot | Yes. |
| 15:59.41 | WizJin | i did it and core set verbose 10 too, but somehow the CLI doesn't capture the debug info, i am entering asterisk with -rn flag, i don't know may be if i need to use some other flag to see sip debug |
| 16:00.04 | Samot | asterisk -rvvvv |
| 16:00.08 | Samot | sip set debug on |
| 16:00.11 | Samot | That's it. |
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| 16:06.42 | WizJin | whats the command to copy putty shell output of debug info |
| 16:08.24 | [TK]D-Fender | <Samot> Highlighting it copies it |
| 16:08.24 | [TK]D-Fender | <Samot> Or clicking on the top bar and select all |
| 16:08.33 | [TK]D-Fender | pay attention when given instructions |
| 16:13.24 | *** join/#asterisk znoteer_ (~znoteer_@23.91.158.45) |
| 16:13.43 | WizJin | https://pastebin.com/0Kt3hm93 |
| 16:14.29 | [TK]D-Fender | clearly no SIP debug there |
| 16:18.21 | WizJin | https://pastebin.com/rpUpfrCZ |
| 16:18.57 | WizJin | the thing is, that i tried sip set debug on, but somehow even though i make calls, info doesn't come through on CLI, so instead i tried formt sip set debug on peer <peername> |
| 16:19.06 | WizJin | and that showed some SIP traffic on CLi |
| 16:20.28 | [TK]D-Fender | no |
| 16:20.40 | [TK]D-Fender | show us the attempt to use the command we gave you |
| 16:21.09 | WizJin | ok |
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| 16:25.56 | WizJin | https://pastebin.com/LjSipHyL |
| 16:25.58 | WizJin | this is the output i get |
| 16:27.08 | [TK]D-Fender | show us your connecting to * cli from OS CLI |
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| 16:30.16 | [TK]D-Fender | brb |
| 16:31.17 | WizJin | https://pastebin.com/HMXjELFW |
| 16:31.25 | WizJin | ok |
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| 16:34.11 | Samot | Oh man |
| 16:34.16 | Samot | This is an old version of Asterisk |
| 16:34.47 | Samot | Connected to Asterisk 1.4.39.1-vici <-- So this is some 10 year old version of Vicidial/Asterisk |
| 16:34.58 | Samot | 10 yrs is being kind. |
| 16:35.23 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 16:35.27 | rpifan | a job i applied for said they used vici dail had never heard of it |
| 16:35.50 | Samot | SIP/trunkme1/19022030996||To <-- Even if that was correct it would work as expected. |
| 16:35.58 | WizJin | the client i am using is called |
| 16:35.59 | WizJin | Xdial |
| 16:36.03 | Samot | Since the !! options for the TO/FROM where an Asterisk 14 add-on. |
| 16:36.16 | Samot | It's Asterisk 1.4 |
| 16:36.26 | Samot | It's so dead it's not even a zombie. |
| 16:36.33 | WizJin | haha! |
| 16:36.41 | Samot | There is ZERO support for this. |
| 16:36.58 | [TK]D-Fender | 10 year old shit. |
| 16:37.04 | Samot | You're trying to use _current_ Dial() features/functions that do not exist in 1.4 |
| 16:37.21 | Samot | And whatever else you've might of done that doesn't work in that version. |
| 16:37.26 | [TK]D-Fender | which option exactly? |
| 16:37.30 | Samot | You cannot use 2018 samples or docs |
| 16:37.39 | Samot | The set the TO/FROM in Dial() |
| 16:37.58 | [TK]D-Fender | What to? |
| 16:38.05 | [TK]D-Fender | that isn't what's being used there |
| 16:38.06 | Samot | "SIP/trunkme1/19022030996||To" |
| 16:38.11 | Samot | That's the ||'s |
| 16:38.17 | Samot | Which is wrong, it should be !! |
| 16:38.20 | [TK]D-Fender | that's a pipe.... |
| 16:38.23 | [TK]D-Fender | so he's passing options |
| 16:38.34 | Samot | SIP/trunkme1/19022030996!TO!FROM |
| 16:38.37 | [TK]D-Fender | no |
| 16:38.43 | Samot | That's the new DIAL |
| 16:38.45 | Samot | For Asterisk 14 |
| 16:38.47 | Samot | JFC |
| 16:38.50 | Samot | I do it ALL THE TIME |
| 16:38.52 | [TK]D-Fender | it's in places of COMMAS as a a fuckingDELIMITER |
| 16:38.58 | [TK]D-Fender | because he's on ANCIENT SHIT |
| 16:39.22 | [TK]D-Fender | <PROTECTED> |
| 16:39.49 | Samot | Those are now options in the DIAL string for SIP |
| 16:39.50 | [TK]D-Fender | Dial Options : T + o |
| 16:40.06 | [TK]D-Fender | this has nothing to do with the dial-string |
| 16:40.15 | Samot | OK |
| 16:40.15 | Samot | Dude |
| 16:40.16 | [TK]D-Fender | He's just dialing a stupid number. |
| 16:40.21 | Samot | My point is the same |
| 16:40.25 | Samot | It's not || It's !! |
| 16:40.31 | Samot | It's also something new |
| 16:40.35 | Samot | So it won't work in 1.4 |
| 16:40.48 | [TK]D-Fender | and || is something old and those ARE valid Dial() options |
| 16:40.54 | Samot | OK |
| 16:41.00 | Samot | So old I forgot. |
| 16:41.17 | [TK]D-Fender | I don't care if !! is something that does exist with a purpose NOW, what is there is standard notation for completely valid dial options |
| 16:41.17 | Samot | Another problem.. |
| 16:41.24 | Samot | I wasn't correct BY HIM |
| 16:41.28 | Samot | I wasn't corrected BY HIM |
| 16:41.31 | [TK]D-Fender | he doesn't have a clue |
| 16:41.41 | Samot | OK. |
| 16:41.56 | [TK]D-Fender | and you should know that | was a delimiter and that'd make it the 3rd parameter. You correct tons of people on that fact already |
| 16:42.05 | Samot | I forgot. |
| 16:42.06 | WizJin | right now, i dont have |
| 16:42.17 | Samot | How many CPS does this provider allow you? |
| 16:42.24 | [TK]D-Fender | no idea why you'd jump to a different conclusion especially when we know we're on a decrepit version |
| 16:42.33 | [TK]D-Fender | "oops" ,_ I'm good with that |
| 16:42.34 | Samot | I didn't know it was that old. |
| 16:42.42 | Samot | Until I saw that last debug. |
| 16:42.45 | Samot | Chill. |
| 16:42.45 | [TK]D-Fender | WizJin, that version of * was POOR at handling a lot of calls |
| 16:43.24 | WizJin | my sip.conf https://pastebin.com/DFG6ZFHB |
| 16:43.58 | [TK]D-Fender | WizJin, your provider is rejecting you. Just go talk to them |
| 16:44.12 | Samot | 12:42:15 PM <Samot> How many CPS does this provider allow you? |
| 16:44.17 | WizJin | my extensions.conf |
| 16:44.17 | WizJin | https://pastebin.com/ZUDXZSxb |
| 16:44.20 | Samot | STOP |
| 16:44.24 | WizJin | ok |
| 16:44.26 | Samot | ANSWER THE QUESTION |
| 16:45.21 | *** join/#asterisk visip (~visix@gateway/tor-sasl/visip) |
| 16:45.30 | WizJin | 30 cps |
| 16:45.37 | Samot | What?! |
| 16:45.40 | WizJin | yes |
| 16:45.45 | Samot | That's rather impressive. |
| 16:45.55 | [TK]D-Fender | <[TK]D-Fender> WizJin, your provider is rejecting you. Just go talk to them |
| 16:45.56 | WizJin | because my provider specializes in call center termination, as i told you earlier |
| 16:46.14 | Samot | 30 CPS is a lot of outbound calls. |
| 16:46.22 | [TK]D-Fender | the calls are PERMITTED, but declined either because they don't like how much you're requesting, or they themselves are overloaded, etc |
| 16:46.22 | Samot | Like a way lot. |
| 16:46.25 | [TK]D-Fender | So go talk to them |
| 16:47.38 | WizJin | [TK]D-Fender, see.. the problem is i am the provider here, as my provider --> mysoftswitch(vos300) -> my dialer.. i am trying to configure this dialer for a clien, i have created account for him in vos3000 for billing purposes and the same account i am trying to use here in this autodialer xdial.. (the account works fine in eyebeam) |
| 16:47.57 | [TK]D-Fender | mytrunk2 <------ |
| 16:48.08 | [TK]D-Fender | go figure out why that is responding this way |
| 16:48.12 | [TK]D-Fender | We don't care who/what it it |
| 16:48.20 | [TK]D-Fender | It isn't Asterisk and that is what we support here |
| 16:48.26 | WizJin | ok |
| 16:48.27 | WizJin | chill |
| 16:48.51 | [TK]D-Fender | I am chill |
| 16:48.59 | [TK]D-Fender | there is nothing more for us to debug on this side |
| 16:49.05 | [TK]D-Fender | #solved |
| 16:50.41 | Samot | 30 CPS with a 1 minute duration and a 80% answer rate is like 1440 concurrent calls. |
| 16:52.29 | *** join/#asterisk Typhon (~Typhon@dslb-084-056-163-015.084.056.pools.vodafone-ip.de) |
| 16:53.27 | WizJin | i am permitted 300 concurrent calls |
| 16:53.39 | Samot | Then 30 CPS makes no sense. |
| 16:54.43 | Samot | It's like 6x less roughly to have your CPS match 300 concurrent calls. |
| 16:57.32 | Samot | The only way 30 CPS can match a 300 concurrent call limit with short duration is if 80% of your calls never answer. |
| 16:58.02 | [TK]D-Fender | Go look at how many calls you have going on and look into the other end |
| 17:03.35 | Samot | Well I think drmessano called this weeks ago when these questions started flowing in. All the maths on this equal SPAM calling. |
| 17:06.38 | [TK]D-Fender | Fair suspicion. I like having a nice length of rope in hand before I give it a solid tug personally though.... |
| 17:07.11 | [TK]D-Fender | We're pretty much there now of course |
| 17:07.54 | Samot | Like I said if they are given him 30 CPS with 300 concurrent calls. They are expecting a high failure rate of those calls. |
| 17:08.09 | *** join/#asterisk lilDangerz (~lildanger@fixed-187-189-68-170.totalplay.net) |
| 17:09.03 | [TK]D-Fender | and that is under ideal circumstances. I haven't heard of what is actually controlling the flow at the start. and we don't see how many calls are in progress. |
| 17:09.17 | [TK]D-Fender | But all of that isn't really an issue for us here, |
| 17:09.55 | Samot | Well I do have issue with helping spammers/scammers setup their networks/voice systems. |
| 17:10.27 | [TK]D-Fender | With you there... |
| 17:11.09 | Samot | It's an outbound call center in India. |
| 17:11.24 | Samot | Unless India has radically changed in the last 3.5 years, SIP is highly restricted. |
| 17:12.17 | [TK]D-Fender | IP: 123.63.213.217 Location: |
| 17:12.17 | [TK]D-Fender | <PROTECTED> |
| 17:12.23 | [TK]D-Fender | if it's to be believed |
| 17:12.41 | [TK]D-Fender | or an impact on where the actual system is being deployed |
| 17:12.52 | [TK]D-Fender | In my books "close enough" |
| 17:12.53 | Samot | Oh that just makes it better. |
| 17:13.02 | Samot | Because China does so much legit business when calling the US |
| 17:17.49 | WizJin | well, its for process of data entry |
| 17:18.15 | WizJin | where people are called automatically to fill forms, its predictive dialing and it has 1000s of other uses in business other than spamming |
| 17:18.24 | WizJin | as you gentlemen might very well know |
| 17:18.36 | [TK]D-Fender | plus the spamming |
| 17:18.41 | [TK]D-Fender | you know .. because |
| 17:18.43 | [TK]D-Fender | but whatever |
| 17:18.59 | [TK]D-Fender | maybe you'll let us know when you've figured out the bottleneck |
| 17:19.06 | WizJin | i ll do |
| 17:19.12 | igcewieling | Telemarketers should be shunned by all good people. |
| 17:23.22 | Samot | Torrents can be used legit too. |
| 17:23.48 | Samot | There is a difference between what has legit use and if it is being used legit. |
| 17:24.08 | [TK]D-Fender | You know it's 99.99% of torrents that give the rest of us users a bad name.... |
| 17:24.10 | Samot | I use outbound dialers but I have places doing it for reminder confirmations. |
| 17:24.47 | Samot | Or the auto shop that sends "Your car is ready for pickup" |
| 17:25.48 | Samot | And prisons that use the service for calling back guards to duty |
| 17:25.59 | Samot | Or announcing missing head counts. |
| 17:27.34 | Samot | However, an outbound call center sending 100's of calls a minute to the USA from China. A country we have sanctions on. I highly doubt they are calling to fill out forms from US companies or any companies that do legit US business. |
| 17:29.25 | WizJin | its a survey thing, i will provide you more details of it later |
| 17:30.00 | Samot | That's OK. |
| 17:30.06 | Samot | I don't need to be shown. |
| 17:31.03 | Samot | No one actively signs up to have their numbers called with random surveys. |
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| 17:31.40 | [TK]D-Fender | #optoutfailure |
| 17:32.07 | Samot | Well you know who doesn't have to honor the US "Do Not Call" list |
| 17:32.17 | igcewieling | Anyone! |
| 17:32.20 | Samot | Or the "Robocalling Rules" |
| 17:32.25 | Samot | Well call centers not in the US |
| 17:32.41 | Samot | Or controlled by a US company. |
| 17:33.25 | Samot | Mobile numbers are only allowed to receive unsolicated auto/robo dialed calls from specific sectors. |
| 17:33.28 | igcewieling | in the US they can close down the "company" and re-open under a new name |
| 17:33.31 | Samot | Like emergency or medical. |
| 17:34.39 | Samot | Sure they can. |
| 17:35.23 | Samot | But again, claiming the software being used can be used for more than illegal activities is a poor argument. Especially when all the data provide says otherwise. |
| 17:37.33 | Samot | Like I pointed out a few weeks ago with this. Almost all of the US businesses on Vicidial's website are either no longer active or have dropped Vicidial from their offerings. |
| 17:37.56 | WizJin | well you guys have taken the 30 cps capacity seriously, thats just a thing in my dashboard, |
| 17:38.06 | Samot | Well |
| 17:38.15 | Samot | It's a serious factor. |
| 17:38.16 | *** join/#asterisk jkroon (~jkroon@165.16.204.170) |
| 17:39.29 | Samot | Generally though when the CPS limit is hit a 503 should be returned. |
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| 17:40.01 | Samot | The fact that you're getting a 480 makes me think the call went through, it was in someones switch and it routes to a busy/OOS desitnation. |
| 17:46.40 | [TK]D-Fender | Odd response vs busy/congestion |
| 17:46.52 | [TK]D-Fender | but either way ... still not our concern thankfully |
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| 18:04.16 | igcewieling | a 480 response is not all that uncommon. |
| 18:04.38 | igcewieling | as far as I know it means "something messed up at destination" |
| 18:11.54 | igcewieling | file: do you recall if/when asterisk started supporting tel: in the Diversion header? |
| 18:22.45 | file | no clue, tel stuff comes from contributions |
| 18:22.51 | file | currently unsupported in PJSIP |
| 18:31.26 | igcewieling | file: thanks. My AGI manually parses out the required info from the header since Asterisk can't parse tel: parts. |
| 18:31.43 | igcewieling | not too ugly of a workaround. |
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| 20:24.09 | vader- | hello |
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| 20:52.29 | DanQuinney | hello |
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| 21:21.20 | baby_seal | DanQuinney: vader-: hellp |
| 21:21.26 | baby_seal | hello etc. |
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| 23:32.50 | vader- | Have any of you guys recently done a VoIP system comparison? I have a company who wants to line up a few systems to compare for a small call center. About 35 agents... They have an older system (analog/old digital avaya) now I guess they are getting dinged on monthly or something and also it requires they have a special trained staff member because there is no GUI for it. Everything is command line or something.. |
| 23:33.32 | vader- | so i told them to check out everything from FreePBX, 3CX, Avaya, Mitel, Cisco... but i didn't know if you guys had any comparison sheets already you seen out there |
| 23:33.40 | vader- | they are even open to a cloud based system. |