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| 00:50.15 | kitsune | mvanbaak: Interesting. They could have named it a bit more obviously. |
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| 01:24.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.4 (2017/12/13), Standard: 15.1.4 (2017/12/13); 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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| 14:11.26 | bittis | hey guys, sort of repeating a question from yesterday, i have cdr records where disposition says busy, billsecthough is > 0, which doesn't make sense as billsec should be greater than 0 if the call was answered. Which of the 2 is getting things wrong here? |
| 14:30.58 | Samot | What's the duration of the call? |
| 14:31.13 | Samot | And what is the value of "billsec" |
| 14:34.51 | *** join/#asterisk EmleyMoor (~phil@topdeck.tinsleyviaduct.com) |
| 14:36.40 | EmleyMoor | I've got a bit of a problem... when my Asterisk setup receives an IAX registration attempt on its public IPv4 address from a private one on the local network, it responds from its private, rather that public, address, breaking the communication. Is there aty easy way I can fix this in, say, the configuration? |
| 14:38.04 | [TK]D-Fender | IAX always responds to the IP it comes from based on that that stack |
| 14:38.09 | [TK]D-Fender | IAX does not care about NAT |
| 14:38.26 | [TK]D-Fender | Stop having your devices use the public when it's local |
| 14:39.25 | EmleyMoor | [TK]D-Fender: Problem is, this is a smartphone with Zoiper on it, so sholt of doing some DNS trickery I really can't |
| 14:39.40 | EmleyMoor | short* |
| 14:40.06 | [TK]D-Fender | *'s rules aren't changing |
| 14:40.11 | [TK]D-Fender | pick your poison |
| 14:41.34 | EmleyMoor | I'm just looking for a poison that works... |
| 14:41.49 | Samot | SIP |
| 14:44.09 | EmleyMoor | Samot: I'm not convinced SIP wouldn't foul up in at least the same way |
| 14:44.24 | Samot | Foul up how? |
| 14:44.30 | baby_seal | grandstream wave is a nice softphone for andrioid |
| 14:44.32 | Samot | SIP does care about NAT. |
| 14:44.43 | Samot | And has both internal/external settings for that. |
| 14:44.50 | Samot | It will use the local for local devices |
| 14:45.00 | Samot | And external for external requests/devices |
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| 14:48.23 | bittis | Samot: for example i get duration to be 18 seconds, billsec to be 17, and disposition BUSY |
| 14:48.58 | bittis | this is on a call that was originated |
| 14:49.32 | Samot | You mean via Originate or a callfile? |
| 14:49.52 | [TK]D-Fender | EmleyMoor, either way you're destined for hairpin NAT failure |
| 14:49.56 | bittis | via Originate |
| 14:50.09 | bittis | using AMI |
| 14:51.26 | Samot | In your origination context, are you doing an Answer()? |
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| 15:21.13 | drmessano | grumbles about Zoiper |
| 15:22.36 | drmessano | I wonder if Zoiper will ever be fixed |
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| 15:30.27 | Samot | I use the desktop version for testing here and there but I'd never it use for day to day use. |
| 15:30.41 | Samot | But the mobile version is horrid. |
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| 15:41.05 | drmessano | Zoiper is fine for desktop testing |
| 15:41.22 | drmessano | Their mobile products are broken.. Especially on IOS |
| 15:42.06 | drmessano | They never updated for iOS 10.3, and 11.2.5 is nearly out.. the breaking point at 10.3 means they can't run in the background |
| 15:42.26 | Samot | Yeah. |
| 15:43.00 | drmessano | You know whats interesting |
| 15:43.07 | Samot | What is that? |
| 15:43.08 | drmessano | Im glad 10.3 happened |
| 15:43.32 | drmessano | Because it showed who REALLY was interested in continuing to sell and support a mobile softphone |
| 15:43.50 | drmessano | Some of them were version bumping with minor tweaks, for YEARS |
| 15:44.07 | drmessano | 10.3 fucked it up for all of them |
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| 16:12.02 | koss | anyonw know of a cloud service that can play a pre-recorded message then hangup on the caller, that i can setup extremely quick? |
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| 16:16.02 | mahafyi | drmessano : what is a good choice for a mobile iax2 / sip phone? (free of course, heh) |
| 16:16.28 | drmessano | Bria is really the only good choice |
| 16:17.24 | drmessano | There are no good free options at all |
| 16:18.24 | drmessano | Nobody is going to go through the trouble of writing and maintaining a mobile softphone for free anymore |
| 16:18.36 | drmessano | Not a quality one |
| 16:19.10 | drmessano | Bria is $10 a year. Lunch money. |
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| 16:36.03 | Samot | mahafyi: IAX2 is an Asterisk thing. |
| 16:36.32 | mahafyi | drmessanu , Samot : ty |
| 16:36.39 | mahafyi | pardon the typo |
| 16:36.45 | Samot | mahafyi: It's not really on the radar of softclients that are agnostic. It's really for clients that have Asterisk integration in their scope. |
| 16:37.08 | Samot | mahafyi: As well in the VoIP world, IAX isn't really a thing. |
| 16:37.40 | Samot | Much like Chan_SIP is now, it's been community driven for quite sometime. |
| 16:37.47 | mahafyi | Samot: i use with zoiper actually, sonce sip ports are blocked by ISP here |
| 16:38.11 | Samot | Well if you control the Asterisk box... |
| 16:38.20 | Samot | You don't have to make SIP use 5060 |
| 16:38.40 | Samot | You can make SIP use any ports you want for signaling and/or RTP |
| 16:39.58 | mahafyi | I have not faced any issue with the free android zoiper so far using iax2 actually. |
| 16:40.36 | drmessano | mahafyi: Does Zoiper run in the background without draining your battery? |
| 16:40.42 | drmessano | and do you get all your calls? |
| 16:40.52 | mahafyi | i have to charge my phone twice a day... |
| 16:41.08 | drmessano | and do you get all your calls? |
| 16:41.12 | Samot | That's because Zoiper is draining your battery.. |
| 16:41.24 | mahafyi | i do get the calls, i know because i have a failover to call international to my cell if the zoiper call does not come thru |
| 16:41.43 | Samot | How often does that happen? |
| 16:41.50 | mahafyi | yes, i am sure it is a drain on battery |
| 16:42.12 | Samot | How often are calls that Zoiper should answer get sent to your failover? |
| 16:42.27 | mahafyi | never, unless network is problem |
| 16:42.31 | drmessano | Neither Zoiper nor Linphone have updated appropriately to background properly on newer Android or IOS releases |
| 16:42.42 | Samot | Network problem? |
| 16:42.51 | drmessano | So on IOS and Android, Zoiper runs basically foreground |
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| 16:43.05 | drmessano | But isn't perfect at all.. |
| 16:43.12 | drmessano | Sucks the battery like crazy |
| 16:43.24 | drmessano | Linphone is even worse |
| 16:43.40 | mahafyi | it drains the battery alright |
| 16:45.24 | mahafyi | Samot: mobile data is what i meant by network. but as long as i am in good coverage area, it is fine. |
| 16:45.55 | Samot | Right because it's probably not going to the background. |
| 16:46.33 | Samot | Zoiper decided that instead of fixing the backgrounding issue, they just abuse your battery. |
| 16:46.54 | mahafyi | Samot : i never thought of it that way, it does show up always inthe notifications area |
| 16:47.03 | mahafyi | i will look at Bria also |
| 16:47.50 | mahafyi | maybe i should just make the hangouts my phone |
| 16:47.59 | Samot | What was that other crappy free softphone...sipsomething |
| 16:48.14 | Samot | That was designed to use the Asterisk Presence feature... |
| 16:48.23 | Samot | But only on Chan_SIP |
| 16:48.30 | Samot | When PJSIP was introduced... |
| 16:48.48 | file | backgrounding requires infrastructure to do it correctly, which is a recurring cost |
| 16:48.55 | Samot | People could still use it like normal but all the cool extra features that worked with Chan_SIP did not work with PJSIP. |
| 16:49.37 | Samot | Well Zoiper has a paid version. |
| 16:51.05 | mahafyi | the main thing is in paid there is no iax conference feature, and i just got used to this, since all i need it are for inbounds |
| 16:51.40 | mahafyi | i had a paid one prior, but it was a couple of years ago. |
| 16:53.55 | Samot | OK. |
| 16:53.58 | Samot | IAX = dead |
| 16:54.05 | Samot | IAX = not standard |
| 16:54.21 | Samot | The only providers that can support IAX are using Asterisk.. |
| 16:54.43 | Samot | Or some other rare system that added IAX to draw Asterisk users to them. |
| 16:54.52 | Samot | Like 10+ years ago |
| 16:56.12 | mahafyi | its is my own box, and used just for my international local lDIDs. Also, in India, if you have a public IP, then SIP ports are blocked. If you have a wireless last mile, SIP ports are blocked. You need to get a license to open it up... Hence, this setup works for me. |
| 16:56.57 | Samot | Right. |
| 16:57.06 | Samot | I've had wholesales in India. |
| 16:57.16 | Samot | We used non-standard ports for them on SIP |
| 16:57.33 | Samot | No need for IAX. |
| 16:58.35 | mahafyi | I have yet to change to pjsip :) and stop using iax2, i guess i have to get around to it soon. |
| 17:06.42 | thiagoc | hi all, what parameter in pjsip.conf is used to set the "From" header field? |
| 17:06.50 | thiagoc | from_user and from_domain? |
| 17:07.29 | file | yes. |
| 17:10.36 | thiagoc | in chan_sip if I use "fromdomain=179.124.44.234" the From header is like this: |
| 17:10.39 | thiagoc | From: "asterisk" <sip:2754375@131.161.42.186:5062>;tag=as30bd6a51 |
| 17:10.52 | thiagoc | 131.161.42.186 is the asterisk IP address |
| 17:11.07 | [TK]D-Fender | Show us |
| 17:11.09 | [TK]D-Fender | don't tell us |
| 17:11.12 | thiagoc | if I change to PJSIP the From is like this: |
| 17:11.16 | [TK]D-Fender | we should see your exact config and the full call |
| 17:11.19 | thiagoc | From: <sip:2754375@179.124.44.234>;tag=667e02ae-c114-47a8-869c-82431f524bd2 |
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| 17:28.15 | thiagoc | pjsip.conf: https://pastebin.com/82ywnPFV |
| 17:32.44 | thiagoc | pjsip log: https://pastebin.com/4QXn4g9v |
| 17:32.59 | thiagoc | sip.conf: https://pastebin.com/raw/W9uM1mTH |
| 17:33.16 | thiagoc | sip log: https://pastebin.com/raw/w8nUq1j0 |
| 17:34.42 | Samot | Server: Asterisk PBX 1.8.31.0 |
| 17:35.19 | Samot | thiagoc: What is the actual issue here? |
| 17:36.38 | thiagoc | I'm having a problem where the call drops at 2-3 minutes in pjsip and in chan_sip it doesn't drop |
| 17:37.09 | thiagoc | so I'm trying to compare the sip packets to see if there's some missing configuration on pjsip |
| 17:37.55 | thiagoc | what I see is that the From header is set with the provider IP address in pjsip |
| 17:39.11 | *** join/#asterisk JimVanM (~JimVanM@ip-45-3-5-248.user.start.ca) |
| 17:40.36 | thiagoc | in the OPTIONS packet |
| 17:41.58 | [TK]D-Fender | Here we finally see your * serveer saying BYE to the other, and not your provider saying BYE to you. |
| 17:42.04 | [TK]D-Fender | forget "options" |
| 17:44.58 | thiagoc | [TK]D-Fender, this provider uses the OPTIONS packets to make some check on their side, that's why they are dropping the call |
| 17:45.03 | thiagoc | not in this call |
| 17:45.16 | [TK]D-Fender | no, we see YOUR server saying BYE |
| 17:45.22 | thiagoc | yes yes |
| 17:45.36 | thiagoc | in this case because I just made a test to generate the log |
| 17:45.37 | [TK]D-Fender | Reliably Transmitting (NAT) to 179.124.44.234:5060: |
| 17:45.37 | [TK]D-Fender | BYE sip:179.124.44.234;did=f3c.982598f6 SIP/2.0 |
| 17:45.40 | thiagoc | https://pastebin.com/raw/2F31hAJ0 |
| 17:45.51 | thiagoc | here they send a BYE |
| 17:45.54 | [TK]D-Fender | Don't show useless garbage |
| 17:46.52 | thiagoc | right, will make another test |
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| 17:53.47 | thiagoc | [TK]D-Fender, https://pastebin.com/raw/234Ngfhr |
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| 18:01.51 | file | thiagoc: is the res_pjsip_dlg_options module loaded? |
| 18:04.03 | thiagoc | file, no... loaded it, testing now |
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| 18:13.04 | thiagoc | file, you're my hero! :) |
| 18:13.22 | thiagoc | that was the problem |
| 18:13.24 | thiagoc | thanks man! |
| 18:13.33 | thiagoc | thanks [TK]D-Fender and Samot too |
| 18:31.06 | [sID] | Can anyone use janus webrtc gateway? |
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