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00:35.18 | Kobaz | AL13N_lappy: yes, it's called connected line updates |
00:35.29 | Kobaz | after the transfer they will send, unless they are disabled |
00:35.49 | Kobaz | it's like reverse caller-id |
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06:47.56 | Ravenheart | what phones do you guys use |
06:48.17 | Ravenheart | we bought a bunch of Grandstream GXP2140's |
06:48.31 | Ravenheart | and can't find an option to increase the bitrate |
06:48.59 | Ravenheart | and the weirdest thing of all, they don't actually use the codec you choose to use |
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06:54.38 | drmessano | What do you mean they don't use the codecs you choose to use? |
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06:56.01 | drmessano | What bitrate adjustment are you expecting to find? That's part of the codec choice, not some separate slider |
06:56.11 | Ravenheart | you can choose between several codecs, and there is a preffered codec list, N1 is codec A, N2 is codec B, N3.... |
06:56.31 | Ravenheart | it chose to use a codec from position 7 |
06:56.35 | drmessano | Right. |
06:56.36 | Ravenheart | instead of 1 or 2 |
06:56.47 | drmessano | What did you set on Asterisks end? |
06:57.06 | Ravenheart | alaw, ulaw, g726 and gsm |
06:57.17 | drmessano | Ok and which did it choose? |
06:57.23 | Ravenheart | g726 |
06:57.50 | drmessano | Ok. That seems reasonable |
06:58.09 | drmessano | If you don't want g726 to be chosen, don't allow it |
06:58.11 | Ravenheart | well no because it is way back in the preference list |
06:58.43 | Ravenheart | our SIP provider supports those four codecs |
06:58.45 | drmessano | Well yes, codecs are negotiated based on a variety of factors. |
06:59.10 | Ravenheart | the thing is, when i set all the preffered codecs to be ulaw it chooses it |
07:00.03 | Ravenheart | the problem started when external calls came in |
07:00.05 | drmessano | I would expect that. |
07:00.21 | Ravenheart | if you pick up the channels get bridged OK |
07:00.28 | Ravenheart | but there was no audio on the phone |
07:00.42 | Ravenheart | asterisk would throw this in the console: WARNING[18987][C-00000092]: translate.c:407 framein: no samples for g726tolin |
07:00.56 | drmessano | G711 isn't necessarily a better choice over g726. So if you don't provide more forceful settings, I can see it being negotiated. |
07:01.06 | drmessano | So just exclude it |
07:01.44 | Ravenheart | well it works fine on the other phones :) |
07:01.47 | Ravenheart | just this one |
07:02.28 | Ravenheart | and was just wondering if anybody else uses these Grandstream phones |
07:02.42 | drmessano | Iâm not sure what answer youâre looking for here⦠Based on everything you told me, G726 would not be unreasonable to negotiate⦠Maybe other phones have different logic⦠|
07:03.05 | drmessano | I've used other phones that would do the same |
07:03.06 | Ravenheart | no answer, was just wondering what other people use |
07:03.15 | Ravenheart | and what mine does |
07:03.19 | drmessano | There's nothing wrong with the phones |
07:04.21 | drmessano | I'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. |
07:04.49 | drmessano | It's not just based on the preference list on one leg |
07:05.28 | Ravenheart | well why would it fail to send audio then |
07:05.34 | drmessano | If you really don't want g726 showing up, stop supporting it and remove it from the allow lists |
07:06.52 | drmessano | Maybe 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 |
07:08.01 | Ravenheart | works fine on the other phones |
07:08.16 | drmessano | Ok then |
07:08.32 | drmessano | So the phone is bad. Toss it |
07:11.54 | drmessano | If 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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07:19.20 | drmessano | Hmmm |
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09:00.18 | AL13N_lappy | Kobaz: thanks, is there something needed for configuration for that? and/or maybe not all sip clients are gonna pick up on that? |
09:01.17 | AL13N_lappy | tbh: 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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15:21.41 | Qwell | seanbright: O.o |
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15:33.43 | seanbright | hmm? |
15:34.48 | seanbright | Qwell: assuming you mean astridevcon - i don't have the bandwidth to focus on it today |
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15:44.18 | Qwell | Who said anything about focusing? |
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15:48.25 | seanbright | i also don't need any new ARI features |
15:48.48 | seanbright | why is it impossible for me not to troll? |
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15:52.03 | file | seanbright: unknown |
15:52.48 | seanbright | but seriously - the focus thing |
15:52.51 | seanbright | :D |
15:53.35 | file | seanbright: squirrel! |
15:55.01 | Qwell | seanbright: 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. |
15:55.25 | Qwell | but sure, go do real work that pays bills or whatever |
15:56.07 | Qwell | (but really, go focus on real stuff) |
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16:00.56 | seanbright | lol |
16:01.01 | seanbright | i'm honored |
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20:02.34 | DannyA | hi all. how can i restart the connection to an endpoint (in this case, a voip carrier), without restarting all of asterisk? |
20:03.39 | file | define "restart the connection" |
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20:05.35 | DannyA | so 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 |
20:06.01 | DannyA | if 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? |
20:06.06 | DannyA | *trunk |
20:09.12 | file | ah, that's not a connection per se - it will continue to trying to do OPTIONS to test reachability |
20:13.32 | DannyA | oh, so i can just leave it, and if i get the settings right, it will just suddenly come up? |
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20:25.05 | file | yes |
20:25.24 | DannyA | ok cool. but if i wanted to "force" a check, is there a way to do that? |
20:26.02 | file | there may be a CLI command for it |
20:26.16 | file | I'm in dev con, so can't really look |
20:27.56 | DannyA | np, tanks! |
20:28.00 | DannyA | np, thanks! |
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