IRC log for #asterisk on 20201020

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00:35.18KobazAL13N_lappy: yes, it's called connected line updates
00:35.29Kobazafter the transfer they will send, unless they are disabled
00:35.49Kobazit's like reverse caller-id
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06:47.56Ravenheartwhat phones do you guys use
06:48.17Ravenheartwe bought a bunch of Grandstream GXP2140's
06:48.31Ravenheartand can't find an option to increase the bitrate
06:48.59Ravenheartand the weirdest thing of all, they don't actually use the codec you choose to use
06:50.03*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
06:54.38drmessanoWhat do you mean they don't use the codecs you choose to use?
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06:56.01drmessanoWhat bitrate adjustment are you expecting to find?   That's part of the codec choice, not some separate slider
06:56.11Ravenheartyou can choose between several codecs, and there is a preffered codec list, N1 is codec A, N2 is codec B, N3....
06:56.31Ravenheartit chose to use a codec from position 7
06:56.35drmessanoRight.
06:56.36Ravenheartinstead of 1 or 2
06:56.47drmessanoWhat did you set on Asterisks end?
06:57.06Ravenheartalaw, ulaw, g726 and gsm
06:57.17drmessanoOk and which did it choose?
06:57.23Ravenheartg726
06:57.50drmessanoOk.  That seems reasonable
06:58.09drmessanoIf you don't want g726 to be chosen, don't allow it
06:58.11Ravenheartwell no because it is way back in the preference list
06:58.43Ravenheartour SIP provider supports those four codecs
06:58.45drmessanoWell yes, codecs are negotiated based on a variety of factors.
06:59.10Ravenheartthe thing is, when i set all the preffered codecs to be ulaw it chooses it
07:00.03Ravenheartthe problem started when external calls came in
07:00.05drmessanoI would expect that.
07:00.21Ravenheartif you pick up the channels get bridged OK
07:00.28Ravenheartbut there was no audio on the phone
07:00.42Ravenheartasterisk would throw this in the console: WARNING[18987][C-00000092]: translate.c:407 framein: no samples for g726tolin
07:00.56drmessanoG711 isn't necessarily a better choice over g726.   So if you don't provide more forceful settings, I can see it being negotiated.
07:01.06drmessanoSo just exclude it
07:01.44Ravenheartwell it works fine on the other phones :)
07:01.47Ravenheartjust this one
07:02.28Ravenheartand was just wondering if anybody else uses these Grandstream phones
07:02.42drmessanoI’m not sure what answer you’re looking for here… Based on everything you told me, G726 would not be unreasonable to negotiate… Maybe other phones have different logic…
07:03.05drmessanoI've used other phones that would do the same
07:03.06Ravenheartno answer, was just wondering what other people use
07:03.15Ravenheartand what mine does
07:03.19drmessanoThere's nothing wrong with the phones
07:04.21drmessanoI've had g726 enabled on older Linksys phones and it was chosen every time over G711, unless transcoding was needed.   My provider didn't support it, so those calls always worked out to ulaw.
07:04.49drmessanoIt's not just based on the preference list on one leg
07:05.28Ravenheartwell why would it fail to send audio then
07:05.34drmessanoIf you really don't want g726 showing up, stop supporting it and remove it from the allow lists
07:06.52drmessanoMaybe your provider and the phones don't support the same version of g726.  That was a common problem.  Even Asterisk had a setting buried in Chan SIP for non-standard g726 as they called it.  IIRC it was g726 vs g726aal2
07:08.01Ravenheartworks fine on the other phones
07:08.16drmessanoOk then
07:08.32drmessanoSo the phone is bad.  Toss it
07:11.54drmessanoIf you have identical phones and one is behaving badly, and you're certain you're reproducing the call with success on another device, factory reset it and start over.
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07:19.20drmessanoHmmm
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09:00.18AL13N_lappyKobaz: thanks, is there something needed for configuration for that? and/or maybe not all sip clients are gonna pick up on that?
09:01.17AL13N_lappytbh: i was looking at a good sipclient; but like, linphone in their "newer" version seems to have dropped the guided transfer capability... and ring seems to have weird issues for me...
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11:52.23*** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.0.0, 13.37.0, 16.14.0 (2020/10/20) Standard: 17.8.0 (2020/10/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
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15:21.41Qwellseanbright: O.o
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15:33.43seanbrighthmm?
15:34.48seanbrightQwell: assuming you mean astridevcon - i don't have the bandwidth to focus on it today
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15:44.18QwellWho said anything about focusing?
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15:48.25seanbrighti also don't need any new ARI features
15:48.48seanbrightwhy is it impossible for me not to troll?
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15:52.03fileseanbright: unknown
15:52.48seanbrightbut seriously - the focus thing
15:52.51seanbright:D
15:53.35fileseanbright: squirrel!
15:55.01Qwellseanbright: Yesterday, somebody on our team asked if they should join the devcon channel.  I said that it was primarily used for you trolling.  True story.  You're letting me down.
15:55.25Qwellbut sure, go do real work that pays bills or whatever
15:56.07Qwell(but really, go focus on real stuff)
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16:00.56seanbrightlol
16:01.01seanbrighti'm honored
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20:02.34DannyAhi all.  how can i restart the connection to an endpoint (in this case, a voip carrier), without restarting all of asterisk?
20:03.39filedefine "restart the connection"
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20:05.35DannyAso when i restart asterisk, after 10-20 seconds i'll see that my connections to my carriers are established (or not). so, "Verizon is now reachable", etc
20:06.01DannyAif im playing around with the network settings and routing tables to get a connection working, and then i want to test it, how can i tell asterisk to retry connecting to that carrier connection?
20:06.06DannyA*trunk
20:09.12fileah, that's not a connection per se - it will continue to trying to do OPTIONS to test reachability
20:13.32DannyAoh, so i can just leave it, and if i get the settings right, it will just suddenly come up?
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20:25.05fileyes
20:25.24DannyAok cool.  but if i wanted to "force" a check, is there a way to do that?
20:26.02filethere may be a CLI command for it
20:26.16fileI'm in dev con, so can't really look
20:27.56DannyAnp, tanks!
20:28.00DannyAnp, thanks!
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