| 00:19.45 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 00:19.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.1 (2018/06/11), Standard: 15.4.1 (2018/06/11); 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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| 10:01.41 | mehale | How can I create a context rule or script do define which number it will dial based on wether its online or not? |
| 10:02.12 | mehale | I ask this because if I dont have the softphone running on my mobile phone, I must route a call to it via PSTN |
| 10:11.53 | robl^ | Samot, according to the new carrier, they state it was denied due to discrepencies in account details and I was told to reqest a CSR from the old/existing carrier. The existing carrier is trying to delay providing the CSR. I had to open a "ticket" with the existing carrier's support website. After a couple days they responded with a question, which I answered (within 10 minutes), and a few more days go by, they asked another question, |
| 10:11.53 | robl^ | <PROTECTED> |
| 10:14.11 | robl^ | Samot: We're talking about 50 local DIDs, 6 toll-free, and 1 international |
| 10:17.27 | robl^ | mehale: you can do several options. You can Dial both in the same line both will ring (if softphone is online). Dial(sip/softphone&sip/deskphone) |
| 10:17.59 | mehale | robl^: I tried both, but then I have zoiper and the pstn call at the same time, huge mess⦠|
| 10:18.47 | mehale | sip call is cheaper than pstn, but the phone gives precedence to the pstn call⦠so, it keeps ringing if I answer zoiper. |
| 10:24.20 | robl^ | mehale: try sequential. exten => 222, 1, Dial(sip/softphone) and then exten => n, Dial(sip/deskphone) |
| 10:25.08 | mehale | would there be a way to do it based on registration latency? |
| 10:25.11 | robl^ | so if dialing the softphone doesn't answer or is offline, it fails over to the desphone |
| 10:25.19 | mehale | ok |
| 10:29.48 | mehale | eventually I have sip registered, but with poor internet connection quality. |
| 10:29.57 | robl^ | you can make it a bit more complex. after dialing the softphone, the next line can check the DIALSTATUS variable. One of the returned values is CHANUNAVAILABLE. With a GOTOIF, you can add some logic |
| 10:30.12 | mehale | ok |
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| 10:31.14 | robl^ | mehale are you using chan_sip or pjsip? |
| 10:31.22 | mehale | chan-sip |
| 10:33.07 | robl^ | in the sip.conf, add a "qualify=yes" to you softphone details. Asterisk will regulary check that the softphone is online and responds and then update its status |
| 10:33.37 | robl^ | It will check every 2 seconds. |
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| 10:48.33 | mehale | ok, thanks |
| 10:54.46 | Samot | 6:14:12 AM <robl^> Samot: We're talking about 50 local DIDs, 6 toll-free, and 1 international <-- Are you trying to do all this in one LOA? |
| 10:56.24 | Samot | robl^: Because if you are, that's probably part of the problem. |
| 10:57.01 | robl^ | Samot, no, I am breaking it up into groups. The local DIDs are from 2 rate centers, so I am submitting 2 for them |
| 10:57.40 | Samot | So you submitted four LOAs? |
| 10:58.35 | robl^ | at the moment, only 1 submitted. The idea was to wait until 1 successfully ports, then work with the next group. |
| 10:58.48 | Samot | Yeah, that's not how it works. |
| 10:59.08 | Samot | Just one bad number in the whole LOA can have the entire LOA rejected. |
| 10:59.43 | Samot | The 50 local DIDs technically can't be denied. |
| 10:59.54 | Samot | The 6 Toll Frees and the International, can be. |
| 11:00.54 | Samot | It also means that all the local DIDs in the same Rate Center have to have the same CSR details. |
| 11:01.42 | Samot | robl^: Are you in a contract or do you have an outstanding balance? |
| 11:01.49 | robl^ | hrmm. they are consecutive numbers. XXX-XXX-6720 to XXX-XXX-6749 in this group, hence grouping them into one request |
| 11:02.31 | Samot | I get that and that's the right way |
| 11:03.08 | robl^ | It's a prepaid service. I add funds and it deducts as services are used are charged. I have plenty of credit as I am still using them and will continue to use them as a backup for outbound calls. |
| 11:03.24 | Samot | But if one group is 25 and the other is 25 and one of the 25 in Group A has mis-matching CSR data all can be denied. |
| 11:03.40 | Samot | In Group A |
| 11:06.21 | Samot | Were these 50 DIDs purchased from this provider or did you port them into the provider? |
| 11:07.29 | robl^ | all of the local DIDs were purchased from the them. 1 of the toll-frees was a port. |
| 11:08.05 | Samot | Well the Toll Free are a different story. |
| 11:08.19 | Samot | If you owe even a $1 in Toll Free usages they can deny them. |
| 11:09.17 | Samot | I'd break up the LOAs. |
| 11:09.28 | Samot | Two for the local DIDs in each Rate Center |
| 11:09.37 | Samot | One for the Toll Frees and one for the International. |
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| 11:11.16 | robl^ | Samot: That is what I am doing. I have only submitted one..so far. I submit for one rate center. I am waiting to the port completed before submitting the LOA for the next rate center. |
| 11:11.46 | Samot | And is this a real carrier? |
| 11:14.42 | robl^ | Samot: It is one of the CLECs purchased by Onvoy. |
| 11:15.01 | Samot | 100% sure it's a CLEC? |
| 11:15.13 | Samot | Vitelity is owned by Onvy/Intelliquent |
| 11:15.18 | Samot | They are not a CLEC. |
| 11:17.03 | robl^ | Vitelity isn't? I know they claim to be. |
| 11:17.20 | Samot | They claimed to be now because Onvoy owns them. |
| 11:17.30 | Samot | But Vitelity is an aggregate. |
| 11:17.42 | robl^ | hrmm. |
| 11:17.43 | Samot | Just because you get a DID from Vitelity doesn't mean it's an Onvoy DID. |
| 11:18.10 | Samot | While Vitelity is owned by Onovy, Vitelity has been left as it's own entity pretty much. |
| 11:18.19 | Samot | I have contracts with both. |
| 11:18.42 | Samot | The contract with Viteltiy doesn't mean I have the services I get from the Onvy contract on my Vitelity services. |
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| 11:19.02 | Samot | And Viteity offers services that Onvoy does not. |
| 11:19.47 | Samot | If Vitelity is the losing carrier, then it is possible those DIDs you have are from Level 3 or Bandwidth or ATT or CenturyLink... |
| 11:20.05 | robl^ | right. I know they are separate but Vitelity is owned by Onvoy. I was misunderstood that they were not a CLEC |
| 11:20.26 | Samot | So if the winning carrier is submitting a port to DID owned by Level 3 |
| 11:20.31 | Samot | It's not going to Vitelity |
| 11:20.35 | Samot | It's going to LEvel 3 |
| 11:21.59 | robl^ | so that could be adding to the confusion |
| 11:22.07 | Samot | Correct. |
| 11:22.21 | Samot | See when I get numbers from my carriers.... |
| 11:22.47 | Samot | They are in my name. The carrier has no idea if I've given them to someone... |
| 11:23.23 | Samot | Also, I've had to fight with Vitelity over ports in the past. |
| 11:23.30 | Samot | For all the reasons I'm saying.... |
| 11:24.03 | Samot | It's probably not that they are "fighting" with you, they just have a process to follow |
| 11:24.19 | Samot | And really, I'm usually unimpressed with their LNP team. |
| 11:25.59 | Samot | LNP Requests go from the Winning Carrier to the Losing Carrier... |
| 11:26.29 | Samot | The actual Winning and Losing Carriers are determined by what SPIDs the number is going to and is current on. |
| 11:26.41 | Samot | Vitelity does not have their own SPIDs. |
| 11:27.19 | robl^ | so they are having to relay messages to the owner of the SPIDs.. |
| 11:27.36 | Samot | They don't relay anything. |
| 11:27.45 | Samot | If the winning carrier is ATT |
| 11:28.10 | Samot | ATT is going to look up what SPID those numbers are on. They are going to send the LNP to the owner of the SPID |
| 11:28.23 | Samot | Because *they* are the ones that have to make the changes |
| 11:29.08 | Samot | The owner of the SPID could alert Vitelity that the port is happening... |
| 11:29.19 | Samot | It could ask Vitelity to confirm. |
| 11:29.30 | Samot | Or it might not. |
| 11:32.07 | Samot | And even then, when the losing carrier sends the confirmation to Vitelity it generally is "If you don't respond, it happens" |
| 11:33.04 | robl^ | Samot: thanks for the information |
| 11:34.23 | Samot | Np. I just ported a bunch from Vitelity the other week |
| 11:34.44 | Samot | 60% of the numbers were "on-net" ports for the winning carrier. |
| 11:35.15 | Samot | In other words, 60% of the numbers were already on *their* SPIDs but under Vitelity's account with them. |
| 11:38.24 | robl^ | Ahh, so maybe some of these will be easier than others. ;-) |
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| 12:23.02 | maurice2k | hi there |
| 12:23.55 | maurice2k | anyone in here who can have a look at my "license agreement" and gerrit access? |
| 12:24.38 | file | only legal can look at and accept license agreements. |
| 12:24.54 | maurice2k | how long do they need? |
| 12:25.09 | file | it can take a few days depending on things |
| 12:25.29 | maurice2k | okaaay.. |
| 12:26.21 | maurice2k | i hope they'll hurry up a bit with checking an address :) |
| 12:29.01 | file | it's also early morning, so they're just getting into the office |
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| 12:35.22 | maurice2k | ok, rejected. thanks |
| 12:35.37 | maurice2k | funny guys |
| 12:36.11 | seanbright | you got rejected? sucks, bro |
| 12:36.23 | seanbright | did they say why? |
| 12:36.32 | maurice2k | i got rejected because they don't like my email address... |
| 12:36.38 | maurice2k | well, i can use another one |
| 12:36.56 | seanbright | nice |
| 12:37.11 | maurice2k | honestly, this feels just ... well... awkward |
| 12:39.11 | seanbright | how's that? |
| 12:39.14 | maurice2k | re-applied. |
| 12:40.09 | maurice2k | it would be nice if you like my new email (to whom it may concern)... i just want to get a bugfix into asterisk |
| 12:41.57 | file | legal isn't on here. |
| 12:43.06 | maurice2k | ok then it was just a coincidence that they rejected it right now after waiting a week =) |
| 12:43.29 | seanbright | yes |
| 12:43.49 | seanbright | which issue are you fixing? |
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| 12:45.04 | maurice2k | ASTERISK-26094, ASTERISK-27816 and there are some other bugs i'm currently trying to fix involing stasis |
| 12:45.40 | maurice2k | no one seems to be using stasis apps |
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| 12:46.07 | file | there are people using stasis apps. |
| 12:46.27 | file | they may not be using what you are referring to or in a way you think, but it is being used. |
| 12:46.36 | maurice2k | of course but it's rather unmaintained code |
| 12:46.41 | maurice2k | for years |
| 12:47.13 | file | in what way? |
| 12:47.24 | file | that specific bugs in it aren't getting resolved? |
| 12:48.12 | maurice2k | it was introduced in 2014 but there's not much development going on regarding stasis, no new features and only few code changes |
| 12:48.47 | file | it's doing what many people need |
| 12:48.48 | maurice2k | and the node client (i'm using that) is completely out of date with the swagger api |
| 12:48.52 | maurice2k | ok |
| 12:49.13 | maurice2k | i guess i'm a little bit of a heavy user |
| 12:49.35 | file | as for your license agreement - I reached out to legal |
| 12:49.42 | maurice2k | perfect thanks :) |
| 12:49.48 | file | if you are contributing on behalf of a company then your email address should be for the company |
| 12:49.53 | file | if it's personal, then leave the job title blank |
| 12:50.20 | maurice2k | ok yes, i've fixed that |
| 12:50.52 | file | the email address you used on the pending license is the same as the first one, and the job title is still there |
| 12:51.11 | maurice2k | ? hmm |
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| 12:51.48 | maurice2k | ok then reject and once again? |
| 12:53.17 | file | we pull the email from your JIRA account I think, so if that needs to change it would probably be done in the JIRA profile |
| 12:55.37 | maurice2k | updated |
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| 12:58.11 | offline | i put the reboot in coreboot |
| 12:58.45 | offline | oh, wrong window though :) |
| 12:59.39 | file | maurice2k: you'll need to resubmit... |
| 13:03.49 | maurice2k | file: lovely :) thanks |
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| 14:18.30 | Samot | file: Refresher, both Playback and Background support URIs? |
| 14:18.39 | file | afaik |
| 14:18.45 | Samot | OK. |
| 14:20.16 | Samot | I got three new servers that have to be able to playback/background the same files. |
| 14:20.29 | Samot | So central media server. |
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| 16:05.35 | *** join/#asterisk anonymouz666 (aafe510a@gateway/web/freenode/ip.170.254.81.10) |
| 16:06.46 | anonymouz666 | hi. while dialing outbound (to a trunk), is it possible to change the DOMAIN of contact header? |
| 16:06.57 | anonymouz666 | already did this in the from with from_domain parameter |
| 16:07.30 | anonymouz666 | the outbound invite is set "contact:user@IP.Address" |
| 16:07.41 | anonymouz666 | but I need to use the same domain as is From header |
| 16:07.58 | anonymouz666 | contact:user@FromDomain |
| 16:08.06 | anonymouz666 | I am using PJSIP with asterisk 13.21. |
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| 16:14.21 | file | no, it's not possible |
| 16:26.17 | anonymouz666 | @file: why? any special reason for this? |
| 16:26.31 | file | because the Contact header is supposed to be an address that can be used to reach Asterisk |
| 16:26.49 | anonymouz666 | yes, but while interconnecting some providers are expecting the full FQDN |
| 16:27.07 | anonymouz666 | so they reject the INVITE |
| 16:27.55 | file | I've never heard or seen that in practice, and it's not something chan_sip ever supported |
| 16:28.29 | anonymouz666 | yes, that's why my bet was to use PJSIP |
| 16:28.50 | anonymouz666 | @file: from the code point of view, do you think that will be a trivial change? |
| 16:29.18 | file | I haven't looked at the Contact generation in ~4 years so I don't know |
| 16:31.12 | anonymouz666 | bad luck for me then |
| 16:36.19 | anonymouz666 | @file: maybe if I look into contact_user logic that changes the URI part, I can replicate changing the right var the domain name |
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| 17:32.48 | Intektor | hey, i am new to asterisk and i want to use it for my app, so that every user can call the other one, my question is, how can I dynamically add users/peers to asterisk without haveing to restart the server? |
| 17:34.16 | Intektor | every time a user would register my application, a new peer should be created for that user, do I have to change the file by program or is there a way to tell asterisk with CLI? |
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| 17:35.08 | craigify | Intektor, check out https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 17:35.16 | craigify | specifically for PJSIP, https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
| 17:35.54 | Samot | Keep in mind there is more than just generating a peer. |
| 17:36.06 | Samot | You need to generate the dialplan/voicemail/etc that would go with it. |
| 17:37.33 | Intektor | sure sure, btw when I have compiled asterisk by source, do I have to install PJSIP or is that already included, because I didn't do anything with pjsip but everything worked anyway |
| 17:45.44 | malcolmd | pjproject gets compiled, by default, as part of the build process, for >=15.0.0 |
| 17:46.37 | malcolmd | for >= 13.8.0, you have to direct its installation: recommend ./configure --with-pjproject-bundled |
| 17:46.52 | malcolmd | there's a bit about all of this here - https://wiki.asterisk.org/wiki/display/AST/PJSIP-pjproject |
| 18:05.47 | Samot | Intektor: If you're going to use Asterisk as the telephony engine for your application. You should spend some time on the wiki learning what Asterisk can do and how it does it. |
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| 23:05.23 | Dovid | Does anyone have any ideas as to what would cause this? http://lists.digium.com/pipermail/asterisk-users/2018-June/292970.html |
| 23:09.24 | file | includes are only checked if nothing within the current context matches. |
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