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| 00:55.42 | pts0 | If this is a trunk to flowroute, what is the client_uri here represent? https://pastebin.com/Sp8jb8r3 |
| 00:56.48 | file | it configures the URI that the outbound registration states it would like to register to |
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| 01:10.24 | pts0 | hmm...I don't see anything like that in sip.conf |
| 01:10.40 | pts0 | for my trunk to flowroute |
| 01:10.47 | pts0 | is it needed? |
| 01:12.07 | file | I don't know your Flowroute setup. If it doesn't need outbound registration, then you don't need outbound registration. |
| 01:12.15 | file | it would be a register line in chan_sip |
| 01:13.46 | pts0 | If I'm just doing ip auth is it needed? |
| 01:14.07 | file | that doesn't control how Flowroute directs calls to you |
| 01:14.37 | file | if it's configured in their system to send calls to your IP address all the time, then registration wouldn't be needed |
| 01:15.49 | pts0 | This is my current flowroute config in sip.conf. https://pastebin.com/Ej2JVXFr |
| 01:20.56 | pts0 | Does this look comparable for pjsip? https://pastebin.com/VrSLrvD9 |
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| 01:34.55 | pts0 | file ^ |
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| 02:15.54 | Samot | There's no transport defined. |
| 02:16.01 | Samot | And nothing assigned to the endpoint. |
| 02:16.18 | personx | Where can one ask questions about carrier billing? |
| 02:16.30 | Samot | What questions? |
| 02:17.02 | personx | Question like how one can access it, use it and such. |
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| 02:17.20 | Samot | What do you mean? |
| 02:17.35 | Samot | "carrier billing" means nothing out of context. |
| 02:17.39 | Samot | Who is the carrier? |
| 02:18.04 | Samot | What are you trying to access? |
| 02:20.22 | personx | Like how can a programmer create an app, a service, or whatever is needed to access carrier billing of any given telecom provider so one can make monetary transactions for a given cellphone number. |
| 02:25.25 | personx | [carrier billing == for example, like if you need to make an in-app purchase without going thru the google pay channel [via carrier billing, when available of any telecom provider] or pay someone directly using only his phone number [if his telecom provider also supports carrier billing]] |
| 02:33.30 | Samot | Well one..\ |
| 02:33.41 | Samot | You need to secure an agreement with said providers. |
| 02:33.54 | Samot | No one's going to let you just dip into their swimming pool. |
| 02:34.24 | Samot | But I'm guessing this is going to be away beyond anything you're able to do. |
| 02:40.42 | pts0 | I have the transport defined already |
| 02:40.45 | pts0 | Samot |
| 02:41.34 | pts0 | Not sure about the endpoint, what do I need to do there, everything else in the example seemed to be related to auth |
| 02:42.43 | pts0 | https://pastebin.com/BUCw5yvV |
| 02:45.38 | Samot | In the endpoint section you are missing the transport= |
| 02:45.49 | Samot | To tell the endpoint which transport to use. |
| 02:47.25 | pts0 | ok thanks! |
| 02:48.42 | pts0 | if there's only one does it use that? |
| 02:48.50 | Samot | Of course. |
| 02:48.54 | Samot | It needs a transport. |
| 02:49.00 | pts0 | the few phones i have working don't have a transport= |
| 02:49.01 | Samot | udp, tcp, something. |
| 02:49.24 | pts0 | so if you only have one transport, you don't need that? |
| 02:50.01 | Samot | I honestly don't know. I've always used it because it was documented that way in the wiki for examples. |
| 02:55.33 | pts0 | Samot...thanks. Yeah I think I got lucky. I'll use it to. |
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| 03:28.09 | pts0 | If you guys had a small server on digital ocean with a few phones, would you restrict your open sip port by ip |
| 03:28.17 | pts0 | if you're connected to flowroute |
| 03:29.00 | pts0 | not sure if anyone knows all the ips flow route uses to connect to my sip port |
| 04:06.53 | life_of_e | pts0: https://support.flowroute.com/customer/portal/articles/1852958-firewall-policies-for-flowroute-s-direct-audio |
| 04:09.40 | life_of_e | There's also a link to their management tool which lists all the POPs |
| 04:10.39 | life_of_e | https://manage.flowroute.com/accounts/interconnection/setup/ |
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| 06:07.37 | pts0 | hey thanks! |
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| 09:50.36 | Posterdati | hi |
| 09:54.16 | Posterdati | I'm trying to place a call from a sip phone throught asterisk toward telecom italia provider, this is what I've got: https://paste.debian.net/1069914/ |
| 09:54.22 | Posterdati | please help |
| 09:55.03 | Posterdati | the net is telecom vdsl <--> cisco 887VA <--> OpenBSD Box <--> home lan |
| 09:55.19 | Posterdati | the net is telecom vdsl <--> cisco 887VA <--> OpenBSD Box, IP phone <--> home lan |
| 09:55.24 | Posterdati | thanks |
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| 11:31.58 | Posterdati | fixed! |
| 11:32.00 | Posterdati | thanks! |
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| 14:09.31 | simplydrew_ | Hi all - trying to put together a quick autp-dial solution this weekend for testing a phone deployment of ~600 phones. We had to do some carrier call forwarding to do 1:1 pointing from the old provider to dummy numbers in the new environment. Iâd like to make a batch test call and record the result to a test file. |
| 14:10.19 | simplydrew_ | Text file would either let me know if the number rang busy or fast busy, went straight to voicemail, or rang through successfully to voicemail. Looks like Iâll need a .call file, but not sure how to accomplish the output function Iâm looking for. Any thoughts? |
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| 14:43.05 | sibiria | the "output function"? |
| 14:44.34 | simplydrew_ | sibiria: dumping a text file that has the âresultâ - rang through successfully to a voicemail answer after a couple of rings, went straight to voicemail, fast busy/regular busy response, etc |
| 14:45.07 | sibiria | you can just concatenate the call's disposition etc. to a common file. that works well |
| 14:45.56 | sibiria | read up on the CDR and System functions |
| 14:46.06 | Samot | So you have a bunch of DIDs that are there to be CF'd to? |
| 14:46.13 | sibiria | ie. let system simply echo "blah" >> somewhere.txt |
| 14:46.23 | Samot | And what are you trying to test? If the CF works or that the DIDs are setup right on your end? |
| 14:47.47 | simplydrew_ | Samot: yes, bunch of DIDs that are carrier call forwarded to dummy DIDs on the new phone system. just want to verify 1) the carrier didnât fat finger any forwarding, making sure the DID routes correctly |
| 14:48.02 | sibiria | also, for this to work you will need a common dial plan context to do the dialing. you have limited programmatic functionality in just the call file itself |
| 14:48.12 | sibiria | a dialing context with a hangup handler <- basic robocall setup |
| 14:48.35 | Samot | And where are you going to do this testing from? |
| 14:49.12 | simplydrew_ | and 2) making sure the programming on our end on the new system is right (a successful test would be ringing to voicemail after a couple of rings - as no one is in the office for the weekend) a failed test would be busy or fast busy, immedate routing to voicemail without ringing |
| 14:49.34 | Samot | So the devices are registered? |
| 14:49.56 | simplydrew_ | doing the testing from an Asterisk 13 system with a SIP trunk of 50 channels |
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| 14:50.44 | simplydrew_ | yes, the devices are registered that Iâm testing. Theyâre registered to CUCM. Iâm just using Asterisk as the PSTN to shoot calls out from (inbound to CUCM) to verify and test |
| 14:51.03 | Samot | And the Asterisk to CUCM has been tested? |
| 14:51.38 | Samot | And Asterisk can't be the "PSTN" to shoot calls to the CUCM. |
| 14:51.40 | simplydrew_ | Itâs not integrated (like an Asterisk to CUCM trunk). Iâm just trying to not do manual call testing from my cell phone to 600ish numbers, essentially. Wanting Asterisk to do that work |
| 14:51.49 | Samot | Unless they are going Asterisk --> CUCM |
| 14:52.00 | Samot | Otherwise it's Asterisk -> PSTN -> CUCM |
| 14:52.19 | simplydrew_ | Right - Asterisk > PSTN > CUCM |
| 14:52.26 | Samot | when do these 600 DIDs port? |
| 14:52.52 | simplydrew_ | Not for a few more weeks |
| 14:52.59 | Samot | Wow. |
| 14:53.04 | Samot | That is quite a long time. |
| 14:53.17 | simplydrew_ | Yeah, thereâs some backstory to that. Out of my hands. |
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| 14:53.51 | Samot | Cuz you'll need to do this again. |
| 14:54.29 | Samot | More importantly, you'll need to check from the losing carriers side. |
| 14:54.29 | simplydrew_ | Yeah, Iâll run the test again to make sure the translations on the CUCM side are all correct after the port |
| 14:54.44 | simplydrew_ | Agreed. I understand those variables. |
| 14:54.54 | simplydrew_ | Just trying to accomplish this need for now |
| 14:54.55 | Samot | I mean their network. |
| 14:55.42 | Samot | 600+ DIDs could mean some don't get removed from their switch. Which means their customers could still end up never leaving the network to be routed to the DID and it would fail. |
| 14:55.52 | simplydrew_ | Yup, understood |
| 14:56.07 | simplydrew_ | Not my first rodeo with that element of it. Just with the call testing in this way. |
| 14:56.38 | Samot | I'm not sure you need to use call files for this. |
| 14:56.54 | Samot | You could just write something that parses the list of DIDs and then initiates an Originate. |
| 14:57.04 | Samot | Otherwise you're making 600+ callfiles. |
| 14:57.35 | simplydrew_ | Fair enough. Didnât know what the end state would look like with the callfiles - either one .call file that would kick off all of them or not. |
| 14:57.56 | Samot | How would you do that with one callfile? |
| 14:58.00 | simplydrew_ | Would need some help with what that looks like |
| 14:58.07 | Samot | It going to send you to an extension/context. |
| 14:58.16 | Samot | Then you're going to do a Dial() |
| 14:58.20 | simplydrew_ | I donât know. Wasnât sure. Was just what I found in my research |
| 14:58.26 | Samot | So you would need 600+ Dial()'s. |
| 14:58.34 | Samot | The would be handled 1 by 1 by 1 by 1 |
| 14:58.42 | simplydrew_ | Ah, yeah. Not what I want. |
| 14:59.01 | sibiria | or in parallel per your preference, really |
| 14:59.20 | sibiria | the simplest technical approach to this is exactly a robocaller |
| 14:59.21 | Samot | How would you do it in parallel? |
| 14:59.48 | sibiria | by just feeding multiple call files to asterisk's spool at a time... |
| 15:00.01 | Samot | Like i just said? |
| 15:00.02 | sibiria | unless limited to 1 outgoing channel at a time, that is |
| 15:00.06 | Samot | 600+ call files? |
| 15:00.10 | sibiria | yes? that's nothing |
| 15:00.13 | sibiria | peanuts |
| 15:00.20 | Samot | Or just parse a speadsheet and do Originate? |
| 15:00.22 | Samot | Same thing. |
| 15:00.57 | sibiria | sure, many methods to a robocaller. it's the most straight-forward approach to his problem |
| 15:01.06 | Samot | There is also a 50 channel limit. |
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| 15:01.21 | Samot | There would be a CPS |
| 15:01.34 | Samot | Dumping 600+ callfiles can cause issues. |
| 15:02.12 | sibiria | no it's fine. i do it all the time. asterisk manages. 600 calls at 50 a time, surely under 1 minute each, won't take long |
| 15:02.29 | simplydrew_ | Yeah, call wonât stay nailed up for long. |
| 15:02.30 | Samot | Do you have the same provider as him? |
| 15:02.37 | simplydrew_ | Or, rather, doesnât need to be |
| 15:02.38 | Samot | CPS isn't how many calls are IP |
| 15:02.43 | sibiria | no my point was about asterisk, not his provider |
| 15:02.44 | Samot | CPS isn't how many calls are UP |
| 15:02.50 | Samot | My point was about his provider. |
| 15:02.54 | Samot | The 50 channel limit |
| 15:02.56 | Samot | CPS limits. |
| 15:03.07 | sibiria | yes it's fine, not a problem to portion that |
| 15:03.23 | sibiria | really, this is a 5 line Perl script |
| 15:03.32 | sibiria | it's not more complicated than that |
| 15:03.40 | Samot | I know |
| 15:03.43 | Samot | I do this a lot. |
| 15:04.15 | Samot | I have a provider that is an emergency alerts provider. |
| 15:04.36 | Samot | We're tied into the DHS, NWS and other state/federal alert systems. |
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| 15:05.47 | simplydrew_ | What would this 5 line perl script look like, given all this? |
| 15:06.20 | sibiria | read CPS number of DIDs from text file, replace details in call file template, output, wait a second or two, rinse repeat |
| 15:07.41 | simplydrew_ | And that will also dump a text file with the disposition result? |
| 15:07.53 | sibiria | no, that's where the dial plan you call from the call file come into play |
| 15:08.18 | Samot | So every DID goes straight to an extension? |
| 15:08.20 | sibiria | it would simply dial and you have a hangup/failure handler attached to that dumping the call dispostion and other stuff into a text file |
| 15:08.28 | Samot | All 600+ are 1:1 DID to Extension? |
| 15:08.32 | simplydrew_ | Correct |
| 15:08.42 | simplydrew_ | All user extensions/phones |
| 15:08.54 | Samot | Not a single ring group or IVR for this company? |
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| 15:09.11 | simplydrew_ | Not the ones that are being tested as part of this, no |
| 15:09.17 | Samot | OK. |
| 15:11.11 | simplydrew_ | If either of you want Sunday morning coffee + breakfast money to help, happy to pay. Not the best at the scripting side of this. |
| 15:11.30 | Samot | No one at the company can? |
| 15:12.42 | simplydrew_ | Iâm just doing this out of concern/double check |
| 15:13.20 | simplydrew_ | Would rather do it now and reduce the day 1 cutover issues |
| 15:13.44 | Samot | Aren't they cut over? |
| 15:14.12 | simplydrew_ | They are. But itâs the weekend... |
| 15:14.42 | Samot | OK, so your DIDs work? Right? |
| 15:15.25 | Samot | They have your DIDs programmed into their CUCM? |
| 15:15.30 | simplydrew_ | The temporary, dummy DIDs work, yes. I know that. Need to make sure the 1:1 carrier call forwarding was done correctly |
| 15:15.49 | simplydrew_ | I just want to validate the carrier call forwarding with this script |
| 15:15.56 | Samot | I understand what you want to do. |
| 15:16.02 | Samot | Hence my questions. |
| 15:16.08 | Samot | They have your DIDs programmed into their CUCM? |
| 15:16.27 | simplydrew_ | Everything on the backend is done. Yes. Phones registered, DIDs programmed, etc. Yes. Itâs all done. |
| 15:16.40 | simplydrew_ | Donât need any help on that side of it |
| 15:17.20 | Samot | And how is the provider sending these calls? |
| 15:17.32 | simplydrew_ | I donât see how any of this is relevant |
| 15:17.47 | Samot | You will when shit breaks during the week through normal call flows. |
| 15:18.37 | simplydrew_ | Itâs carrier call forwarding. Theyâre sending them out of their switch, with as many channels that are needed, into our SIP trunk and then to CUCM |
| 15:18.48 | Samot | Not what I was asking. |
| 15:18.58 | Samot | Are they keeping the original DID in the call? |
| 15:19.06 | simplydrew_ | Yes |
| 15:19.21 | simplydrew_ | Again, all of this is covered. I just want to test the DIDs via a script. |
| 15:19.32 | Samot | Then you need to write the script. |
| 15:19.35 | Samot | How are you going to do it? |
| 15:19.46 | Samot | Are you going to use a speadsheet or pull the DIDs from a database? |
| 15:19.56 | simplydrew_ | I have all the DIDs in a text file |
| 15:20.10 | Samot | And what programming languages do you understand? |
| 15:20.26 | simplydrew_ | Good lord⦠|
| 15:20.33 | Samot | That is a valid question. |
| 15:20.36 | simplydrew_ | Nevermind guys. Iâll just go to someone else on this. |
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| 15:20.44 | simplydrew_ | Appreciate the guidance |
| 15:20.49 | Samot | Well one suggestion was PERL |
| 15:20.52 | Samot | Do you know PERL? |
| 15:20.57 | Samot | Would PHP be better? |
| 15:21.00 | Samot | Python? |
| 15:22.50 | Samot | This is going to be the same thing. |
| 15:23.01 | Samot | "Hey, I want to write a script any thoughts or help that can do X" |
| 15:23.14 | Samot | And the people are going to want to know what scripting language this is going to be in for the help. |
| 15:24.25 | Samot | Also how are you going to track the answering of the call? |
| 15:24.53 | simplydrew_ | Iâll just go give someone the requirements and pay them to do it at this point |
| 15:25.10 | Samot | Because you have to consider false positives. |
| 15:25.10 | simplydrew_ | Figured there might have been something already out there that could be edited up to do it |
| 15:25.18 | sibiria | do you know how to write dial plans, simplydrew_? |
| 15:25.33 | sibiria | you need not just something to portion the call files out |
| 15:25.44 | sibiria | but also something for asterisk to do with the call files and their result |
| 15:26.26 | Samot | Don't forget about false positives. |
| 15:28.00 | sibiria | here https://pastebin.com/raw/Zbre6cDN |
| 15:28.07 | sibiria | perl robocaller. have fun with it |
| 15:28.34 | sibiria | don't wardial!!1 |
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| 16:03.49 | genewitch | dang kids wardialing our modems and breaking into the network |
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| 18:04.07 | minipudding | hi, i'm playing around a bit with linking two asterisks (16.2) |
| 18:04.24 | minipudding | and when i look at the pjsip debug the sip headers have a field called "Accept" |
| 18:04.31 | minipudding | which lists a lot of things |
| 18:05.01 | minipudding | application/xpidf+xml, application/cpim-pidf+xml etc. and several of them are listed twice even |
| 18:05.20 | minipudding | what are these? is it other "services" asterisk is listening for besides normal sip + rtp? |
| 18:06.31 | file | no, it's defined in SIP as the content types that the implementation understands/produces |
| 18:07.31 | minipudding | ok so it's nothing i need to worry about and know how to disable |
| 18:07.43 | file | nope, just the way SIP works |
| 18:08.19 | minipudding | ok thanks |
| 18:08.46 | minipudding | and another question i have. i use pjsip and the "wizard" and i make endpoints that are just type=wizard |
| 18:08.57 | minipudding | and i use remote_hosts to point to the opposite virtual machine |
| 18:09.15 | minipudding | is this the correct way to make sure that only the ips in remote_hosts can connect? |
| 18:09.56 | minipudding | [box1] type=wizard remote_hosts=ip.of.box2 so to speak |
| 18:10.24 | minipudding | [box2]* |
| 18:10.41 | minipudding | and box2 defines the same but for box1 |
| 18:12.27 | Samot | Endpoints can't be type=wizard. |
| 18:12.35 | Samot | The PJSIP Wizard is to create the configs. |
| 18:12.45 | Samot | Endpoints are type=endpoint |
| 18:13.44 | minipudding | so what i have done is more like linking two phones? |
| 18:14.11 | Samot | Asterisk doesn't care what the actual endpoint is. |
| 18:14.18 | Samot | It could be to your PSTN provider |
| 18:14.20 | Samot | It could be a phone |
| 18:14.31 | Samot | It could be another Asterisk box. |
| 18:15.01 | Samot | The PJSIP Wizard will create the needed sections with the information you pumped in it to pjsip.conf |
| 18:15.10 | Samot | It will create the endpoint, aor, auth, etc sections. |
| 18:15.30 | Samot | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard |
| 18:16.13 | minipudding | yes that's where i came from, it's how i made the two objects or whatever the parlance is |
| 18:16.42 | minipudding | but ok so this is an acceptable way to link two asterisks so they can dial |
| 18:16.47 | minipudding | (eachother) |
| 18:17.02 | Samot | But the actual endpoints for those phones are type=endpoint |
| 18:17.07 | Samot | When you look at pjsip.conf |
| 18:17.18 | Samot | Each one has an endpoint, aor, auth section for it. |
| 18:18.00 | minipudding | i remove all auth options or rather i set sends/accepts for auth and registrations to no |
| 18:18.06 | Samot | OK\ |
| 18:18.08 | minipudding | this means effectivel that "IP auth" is in place? |
| 18:18.15 | Samot | Then you still have an endpoint and aor section |
| 18:18.40 | minipudding | what i'm trying to get is what i had with chan_sip when setting allowguest=no |
| 18:18.48 | Samot | Why? |
| 18:19.01 | Samot | By default PJSIP has no concept of anonymous. |
| 18:19.04 | minipudding | so that only the remote_hosts IPs are allowed to contact this or that pjsip object |
| 18:19.13 | Samot | That's default. |
| 18:19.21 | minipudding | ok then all is good |
| 18:19.25 | Samot | There is no "anonymous" in PJSIP |
| 18:19.36 | Samot | You would have to create an endpoint called anonymous for it to work. |
| 18:19.37 | minipudding | yes i understand you have to make an explicit object for that |
| 18:19.55 | minipudding | i'm just not sure how "remote_hosts" work when i set no auth/registration for the object |
| 18:19.55 | Samot | It only cares about the object is has. Anything else is not found/denied. |
| 18:20.02 | minipudding | ok |
| 18:20.09 | Samot | It means IP Auth. |
| 18:20.13 | minipudding | ok thanks |
| 18:20.23 | Samot | In PBX A you set the server_uri to PBX B |
| 18:20.27 | Samot | Vice versa. |
| 18:21.02 | Samot | You add an identify section with the match= setting to match the IP of PBX B. |
| 18:21.24 | Samot | That is what allows the endpoint to accept calls from the other PBX. Vice versa in PBX B. |
| 18:22.49 | minipudding | right now they accept calls from eachother, without registration/identify set. but i must add that as well? |
| 18:22.50 | Samot | Sorry, not server URI |
| 18:23.03 | Samot | You're going to have an AOR with the a static contact. |
| 18:23.24 | Samot | Show your pjsip.conf's. |
| 18:23.57 | minipudding | ok one moment |
| 18:30.37 | minipudding | https://paste.debian.net/hidden/241c438e |
| 18:36.06 | Samot | One sec, I had a call due to ATT/Hilton screwing things up. |
| 18:36.41 | Samot | minipudding: Did you look at the link I sent you? |
| 18:37.14 | minipudding | yes i've been there before but what specifically do you mean? |
| 18:37.18 | Samot | There's no endpoint here, there's no aor... |
| 18:37.22 | Samot | You didn't run the wizard. |
| 18:37.42 | minipudding | there should be an endpoint if i do endpoint/stuff no? if i understand that wizard correctly |
| 18:37.43 | Samot | You created the conf file but didn't generate the actual stuff for pjsip.conf |
| 18:38.28 | minipudding | that is if via the wizard say anything/xyz=123 it goes into an object of its own with type=anything xyz=123 |
| 18:38.38 | minipudding | or have i misunderstood the wizard |
| 18:39.04 | *** join/#asterisk Janos__ (~Janos@201.204.94.76) |
| 18:39.11 | minipudding | i mean the two endpoints show up on each box |
| 18:39.19 | minipudding | or "contacts" as asterisk refers to them as now |
| 18:39.26 | minipudding | and i can dial from one to the other |
| 18:39.44 | Samot | Ahh NM. |
| 18:39.50 | minipudding | "Contact box2/sip:10.0.28 is now Reachable" |
| 18:40.07 | Samot | No, I misunderstood what the wizard can fully do. |
| 18:40.15 | minipudding | ah ok |
| 18:40.23 | Samot | So you're fine. |
| 18:40.36 | minipudding | but maybe this is the wrong setup still? or are they really "contacts" in the eyes of pjsip? |
| 18:40.53 | minipudding | i don't fully know the correct lingo here |
| 18:41.10 | Samot | What is the output of: pjsip show contacts |
| 18:41.56 | minipudding | Contact: box2/sip:10.0.2.8 <some hash> Avail <rtt value> |
| 18:42.00 | Samot | There you go |
| 18:42.03 | Samot | It created things |
| 18:42.16 | Samot | pjsip show endpoint <-- That will kick out the endpoint details |
| 18:42.22 | Samot | pjsip show endpoints <-- That will kick out the endpoint details |
| 18:42.51 | minipudding | ah yes it shows box2 as endpoint with same aor and one contact, and match etc. |
| 18:42.58 | Samot | Yup. |
| 18:43.00 | Samot | You're fine. |
| 18:43.01 | minipudding | ok great |
| 18:43.03 | minipudding | thanks a bunch |
| 18:43.08 | minipudding | and thanks to file also |
| 18:43.22 | minipudding | and i will retreat to play more with this, have a good evening |
| 19:48.05 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:19.42 | *** join/#asterisk AndChat|620489 (~AndChat62@101.sub-97-44-1.myvzw.com) |
| 20:20.37 | *** join/#asterisk AndChat|620489 (~AndChat62@101.sub-97-44-1.myvzw.com) |
| 20:21.51 | *** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b021:ef25:d9d3:cf4f:140c:9a0c) |
| 20:30.37 | *** join/#asterisk rpifan (~rpifan@ipb218f026.dynamic.kabel-deutschland.de) |
| 20:36.41 | *** join/#asterisk as3ty (uid332392@gateway/web/irccloud.com/x-yntbvyspyuwugqkn) |
| 20:38.29 | as3ty | Just curious, how do you guys handle client tickets opened for call failures because the upstream provider is taking its time connecting the call? |
| 20:39.18 | as3ty | Because there is no ring from the time an outgoing call is matched with a dial pattern to the actual ring, the client thinks the whole system is down. |
| 20:39.51 | Samot | I provide ringback unless earlier media is detected. |
| 20:39.57 | Samot | So there is no silence. |
| 20:40.15 | Samot | And connecting the call how? |
| 20:40.55 | as3ty | Samot: Just through a good ol SIP trunk. |
| 20:41.17 | Samot | You're saying they are taking their time... |
| 20:41.23 | Samot | How are they taking their time to connect a call? |
| 20:41.27 | as3ty | The problem is with some of the international phone numbers, it can take up to 20 seconds to connect. And that's across multiple carriers. |
| 20:41.34 | Samot | You mean getting a 100 Trying or 180/183 back? |
| 20:41.56 | Samot | Then you just provide ringback |
| 20:41.59 | as3ty | Yes, we're getting a 100 Trying... |
| 20:42.02 | Samot | rR options. |
| 20:42.22 | as3ty | Ok, I will look into that. Thanks. |
| 20:43.08 | Samot | Users are going to look at you for Post Dial Delay issues if you are who they are paying for service. |
| 20:54.53 | *** join/#asterisk rpifan_ (~rpifan@ipb218f026.dynamic.kabel-deutschland.de) |
| 20:54.59 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 20:55.11 | as3ty | Samot: That makes sense.. |
| 21:06.35 | *** join/#asterisk rpifan (~rpifan@ipb218f12c.dynamic.kabel-deutschland.de) |
| 21:25.47 | *** join/#asterisk rpifan_ (~rpifan@ipb218f017.dynamic.kabel-deutschland.de) |
| 22:17.42 | *** join/#asterisk rpifan (~rpifan@ipb218f10a.dynamic.kabel-deutschland.de) |
| 22:18.51 | *** join/#asterisk rpifan (~rpifan@ipb218f10a.dynamic.kabel-deutschland.de) |
| 22:24.42 | *** join/#asterisk rpifan (~rpifan@ipb218f10a.dynamic.kabel-deutschland.de) |
| 22:41.42 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:6123:9107:fc29:f7d9) |
| 22:57.34 | *** join/#asterisk pts0 (~pts0@cpe-45-37-8-40.nc.res.rr.com) |
| 22:58.11 | pts0 | is this still a good way to send voicemail to email as mp3s or is there something builtin now? http://bernaerts.dyndns.org/linux/179-asterisk-voicemail-mp3 |
| 23:02.18 | [TK]D-Fender | nothing built-in |
| 23:02.25 | [TK]D-Fender | same solutions as always |
| 23:06.05 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 23:06.11 | pts0 | thanks |
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